| 00:03.07 | *** join/#asterisk jksM (~jks@193.189.93.254) |
| 00:04.01 | *** join/#asterisk QbY (~QbY@c-24-98-101-168.hsd1.ga.comcast.net) |
| 00:04.10 | *** join/#asterisk Katty (~User@71.157.169.103) |
| 00:04.16 | Katty | Qwell: ping |
| 00:04.26 | QbY | in sip.conf would i need to create multple peers since my "peer" can come from multiple ip addresses? |
| 00:04.36 | *** join/#asterisk Rug (~rug@dyn216-8-170-99.ADSL.mnsi.net) |
| 00:07.21 | Katty | Qwell: come find me on the roulette game! |
| 00:07.40 | Qwell | Katty: fixing code! |
| 00:08.09 | jaytee | ah, the Neverending Story |
| 00:08.09 | Katty | :< |
| 00:08.13 | Katty | butbut |
| 00:08.13 | Katty | but |
| 00:08.15 | Katty | butbut |
| 00:08.15 | Katty | but |
| 00:08.25 | Katty | k |
| 00:10.54 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 00:10.55 | Rug | How much overhead can I expect on a single-line PBX (acting as a VoIP gateway)? |
| 00:11.37 | Rug | Sorry, and answering machine |
| 00:11.46 | etfonhomey | Rug, what kind of call volume are you talking about? |
| 00:12.00 | Rug | Very low call volume. |
| 00:12.24 | Rug | I am just trying to decide if I should build a dedicated box, or add it to my main server. |
| 00:12.27 | Rug | <PROTECTED> |
| 00:14.08 | Rug | 1 FXO and 2-3 FXS |
| 00:14.54 | etfonhomey | Shouldn't require many resources at all. |
| 00:15.23 | Rug | Does asterisk save sound files in .ogg? |
| 00:16.35 | etfonhomey | Not sure, if not, you could have it run something from a command prompt to covert files to that format upon hangup, for example. |
| 00:16.54 | etfonhomey | convert* |
| 00:17.10 | Rug | It doesn't really matter just wondering. |
| 00:17.27 | Rug | considering how easily & well supported .ogg is in the Linux world |
| 00:18.48 | *** join/#asterisk ppc (~ppc@96.19.60.248) |
| 00:21.37 | *** join/#asterisk Mark22 (~mark@unaffiliated/mark21) |
| 00:22.02 | Mark22 | Hello, now looking into AGI scripting. What is a good documentation source about how it works? |
| 00:22.18 | Katty | reddit roulette is boring tonight :< |
| 00:23.27 | Katty | Qwell come talk to me |
| 00:23.49 | Rug | Is the TDM410 well supported? |
| 00:23.58 | file | Katty: bribery. |
| 00:24.30 | Katty | what's bribery |
| 00:25.15 | *** join/#asterisk albertoandrade (~albertoan@189.34.121.136) |
| 00:33.02 | *** join/#asterisk voipmonk (~shido6@CPE002191f85581-CM001692568382.cpe.net.cable.rogers.com) |
| 00:35.36 | *** join/#asterisk Mango (~iMango@d154-20-110-253.bchsia.telus.net) |
| 00:37.25 | Z_God | drmessano: using it with a softphone |
| 00:38.19 | Mango | I'm at the "Asterisk Module and Build Option Selection" |
| 00:38.36 | Mango | I want to install func_curl but it states it depends on "curl(E)" |
| 00:38.54 | Mango | I did yum install curl but that didn't help. If anyone has any ideas that would be lovely :) |
| 00:39.31 | drmessano | <PROTECTED> |
| 00:39.37 | voipmonk | ding |
| 00:39.43 | Mango | thx |
| 00:40.36 | ManxPower-work | Anyone happen to have any web servers that require DIGEST authentication hanging around I can test against? |
| 00:41.22 | voipmonk | you want a webserver to eat your auth?:) |
| 00:43.08 | ManxPower-work | it doesn't matter, I'm just looking for the headers sent on the initial connection |
| 00:43.35 | ManxPower-work | every server I try uses basic auth |
| 00:49.41 | *** part/#asterisk QbY (~QbY@c-24-98-101-168.hsd1.ga.comcast.net) |
| 00:56.46 | *** join/#asterisk Tamo (~I@cpc3-sgyl27-2-0-cust52.sgyl.cable.virginmedia.com) |
| 00:57.42 | Katty | Qwell: are you done yet >.< |
| 01:00.31 | *** join/#asterisk Ta^3 (~tacvbo@189.146.190.221) |
| 01:01.26 | *** join/#asterisk norrec (~norrec@76-201-85-140.lightspeed.frokca.sbcglobal.net) |
| 01:04.15 | rue_house | wow did I ever screw up, I removed the first line of the context for the main company number, it wasn't till afternoon everyone noticed the phones weren't ringing.... |
| 01:04.33 | carrar | YOU'RE FIRED |
| 01:05.18 | *** join/#asterisk corretico (~laguilar@201.201.46.106) |
| 01:05.20 | rue_house | *oops* |
| 01:05.48 | rue_house | it was only one fo the 3 companys people answer for, so tthats part why they didn't notice |
| 01:06.08 | Katty | yeah i've done that crap before |
| 01:06.15 | rue_house | however, I think I'll make a default context |
| 01:07.00 | Katty | i think i might take a nap |
| 01:07.18 | Mango | Me too :P |
| 01:07.31 | rue_house | ring everyone |
| 01:07.47 | Mango | Does anyone know offhand if DNS SRV works in the dial plan of Linksys devices? |
| 01:07.51 | rue_house | lawnmower man |
| 01:12.03 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 01:13.34 | *** join/#asterisk Jacke (jacke@czokopinda.niebanalni.com) |
| 01:20.55 | *** join/#asterisk stallione (~stallione@c-67-173-76-139.hsd1.il.comcast.net) |
| 01:23.22 | *** join/#asterisk ZX81 (~Matt_Ridd@121-74-248-216.telstraclear.net) |
| 01:23.43 | *** part/#asterisk ZX81 (~Matt_Ridd@121-74-248-216.telstraclear.net) |
| 01:26.42 | *** join/#asterisk stallione (~stallione@c-67-173-76-139.hsd1.il.comcast.net) |
| 01:27.51 | *** join/#asterisk Kumbang (~kumbang@125.163.83.153) |
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| 01:35.27 | *** part/#asterisk ZX81 (~Matt_Ridd@121-74-248-216.telstraclear.net) |
| 01:42.31 | aces1up | whats the diff between a t1 or pri services? |
| 01:45.57 | *** join/#asterisk stallione (~stallione@c-67-173-76-139.hsd1.il.comcast.net) |
| 01:46.01 | hipitihop | I'm running an NVIDIA ION Atom 330 based mini pc running MythTV backend/frontend but is idle 90%. Considering I just want to handle 1-2 simultaneous voip calls utilizing existing ATA connected handset, is this computer ok for asterisk or does it really want a dedicated machine. |
| 01:48.30 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 01:48.56 | voipmonk | give it a whirl see what happens |
| 01:49.12 | voipmonk | i would say go for it - and listen for jitter |
| 01:49.32 | voipmonk | esp during playbck or recording |
| 01:51.34 | *** join/#asterisk Xetrov` (~xetrov@unaffiliated/xetrov/x-827361) |
| 01:55.03 | *** join/#asterisk nighty^ (~nighty@210.188.173.245) |
| 01:55.06 | hipitihop | voipmonk, thanks I'll see how I go |
| 01:55.40 | *** join/#asterisk n0cturnal (~n0c@203.161.121.158.static.amnet.net.au) |
| 01:55.59 | hipitihop | what port does the broswer gui run on ? |
| 01:56.34 | jaytee | asterisk doesn't come with a gui |
| 01:56.45 | voipmonk | well |
| 01:56.50 | voipmonk | thats not entirely true anymore |
| 01:57.01 | hipitihop | ah, must have missread the book |
| 01:57.42 | voipmonk | the new version of asterisk does have a built in gui - but it isnt the best implementation of a gui |
| 01:59.04 | jaytee | what new version? you talking about AsteriskNOW? |
| 01:59.12 | hipitihop | voipmonk, I'm mainly asking because the book gave me the impression there was a browser based config GUI available and I don't want the install to clobber the default MythWeb one |
| 01:59.40 | *** join/#asterisk f0urtyfive (~noone@75.150.130.121) |
| 01:59.55 | f0urtyfive | anyone have any hints on how to debug meetme? |
| 02:00.04 | f0urtyfive | my meetme rooms just disconnect me when I call them |
| 02:00.13 | voipmonk | oh it wont do that hipitihop |
| 02:00.14 | f0urtyfive | and it gives me nothing even with core set debug up |
| 02:00.57 | voipmonk | do u have debug enabled in logger.conf, f0urtyfive ? |
| 02:01.15 | voipmonk | enable console debugging |
| 02:01.25 | voipmonk | if thats where u want to see the debug |
| 02:01.45 | f0urtyfive | ah |
| 02:01.49 | f0urtyfive | Unable to open pseudo device |
| 02:02.38 | f0urtyfive | whats the damn module name |
| 02:03.22 | *** join/#asterisk stallione (~stallione@c-67-173-76-139.hsd1.il.comcast.net) |
| 02:05.11 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 02:06.54 | aces1up | anyone here ever worked with merlin magix systems? |
| 02:07.28 | voipmonk | heh |
| 02:07.43 | voipmonk | u have dahdi loaded, f0urtyfive ? |
| 02:07.56 | voipmonk | aces1up: what are you attempting to use asterisk for? |
| 02:08.05 | aces1up | i'm not really.. |
| 02:08.07 | aces1up | just stuck |
| 02:08.21 | aces1up | and thought someone would have some advice? is this a really crappy system? |
| 02:08.33 | aces1up | what is the cheapest polycom phone go for? |
| 02:09.29 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 02:09.31 | *** join/#asterisk xpot-mobile (~xpot@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 02:10.05 | adnc | hello, i'm looking for a variable that would deliver my the time, i used TIMESTAMP but unfortunately this doesnt give any results |
| 02:16.43 | russellb | *CLI> core show function STRFTIME |
| 02:16.45 | russellb | I think. |
| 02:17.05 | *** part/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 02:17.15 | f0urtyfive | voipmonk: I didnt, that took care of it. |
| 02:17.40 | f0urtyfive | however, now it seems that asterisk segfaults when it tries to send a voicemail via email. |
| 02:17.52 | russellb | that seems bad. |
| 02:18.18 | voipmonk | that does, doesn't it? :) |
| 02:18.26 | f0urtyfive | I'd say so. |
| 02:18.35 | f0urtyfive | it somewhat prohibits proper voicemail leaving. |
| 02:18.36 | adnc | russellb, thank you |
| 02:19.32 | russellb | f0urtyfive: latest version? |
| 02:19.48 | f0urtyfive | nopers. |
| 02:20.08 | russellb | k. |
| 02:20.27 | adnc | russellb, is there also a variable for the callerid, CALLERIDNUM doesnt work |
| 02:20.43 | russellb | adnc: ${CALLERID(num)} |
| 02:20.49 | russellb | *CLI> core show function CALLERID |
| 02:21.45 | *** join/#asterisk alexsea7 (~Sheeju_Al@124.123.176.207) |
| 02:22.24 | alexsea7 | https://issues.asterisk.org/view.php?id=16729 this issue is there for asterisk 1.4.29 version, how to fix this? |
| 02:23.07 | russellb | alexsea7: install 1.4.30-rc2 |
| 02:28.12 | alexsea7 | ok but is it only change in chan_local.c? if so I would just like to upgrade it |
| 02:29.28 | russellb | there was more to it, but yeah, you could fix it with just a change to chan_local.c. |
| 02:29.59 | alexsea7 | ok thanks.. |
| 02:33.51 | *** join/#asterisk pentanol (~Unknown@91.195.60.231) |
| 02:34.03 | *** join/#asterisk Torrieri (~Torrieri@nelug/crew/torrieri) |
| 02:42.31 | adnc | if i use monitor or mixmonitor functions, where does asterisk put these files, can this be controlled? |
| 02:42.59 | Mango | Yes. |
| 02:43.11 | Mango | exten => 200,n,MixMonitor(/home/mango/recordings/${UNIQUEID}.ulaw) |
| 02:43.35 | adnc | i see, and if there is no path to it? |
| 02:44.09 | adnc | ahh i see /var/spool/asterisk |
| 02:44.15 | Mango | I believe you can configure the default monitoring directory in asterisk.conf |
| 02:44.29 | adnc | mango thank you |
| 02:44.58 | Mango | You are very welcome :) |
| 02:51.07 | *** join/#asterisk OrNix (~ornix@l151-249-47.static.cn.ru) |
| 02:53.14 | *** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk) |
| 02:54.24 | adnc | when i do a call it quite often rings until the other parties phone starts ringing, can i do something again this? |
| 02:55.49 | *** join/#asterisk Acro (~omgponies@unaffiliated/acro) |
| 02:56.48 | Katty | hello. |
| 02:57.09 | Mango | adnc: .......huh? |
| 02:57.21 | *** join/#asterisk grey-monkey (~ericshell@75-148-103-190-Utah.hfc.comcastbusiness.net) |
| 02:57.39 | adnc | Mango, the other parties phone rings quite late |
| 02:57.51 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 02:58.13 | adnc | but my phone gives ringing signals, i wonder why this happens |
| 02:58.26 | Mango | ah. Who is your SIP provider? It's possible they're taking their time terminating the call. |
| 02:58.33 | *** part/#asterisk alexsea7 (~Sheeju_Al@124.123.176.207) |
| 02:59.09 | Mango | ringing tone is fairly easy to do - someone along the line is probably doing something like this: exten => _NXXNXXXXXX,1,Ringing |
| 02:59.10 | adnc | mango, i use sipgate.de here in germany, but i use it with an other pbx (fritzbox) where it does not behave like this |
| 03:00.21 | Mango | if you enter the CLI in verbose mode, what does it say? |
| 03:01.03 | adnc | ohh |
| 03:01.12 | adnc | i need to find out how to set it to verbose mode |
| 03:01.24 | Mango | core set verbose 3 |
| 03:01.39 | *** part/#asterisk LemensTS (~customgt@adsl-70-238-130-71.dsl.stlsmo.sbcglobal.net) |
| 03:03.22 | adnc | -- Executing [4316999@ankommend:3] Dial("SIP/sipgate.de-081b9558", "SIP/30|30|r") in new stack |
| 03:03.41 | adnc | -- SIP/30-081b4f88 is ringing |
| 03:04.00 | adnc | nothing more |
| 03:04.02 | adnc | ahh |
| 03:04.17 | adnc | i've two pbx behind the same network |
| 03:04.30 | adnc | the asterisk and the fritzbox |
| 03:04.36 | adnc | could this be causing this problem? |
| 03:05.46 | Mango | I dunno. I'm out of ideas I'm afraid. |
| 03:07.30 | *** part/#asterisk shin-shoryuken_ (~shin-shor@hephaestus.restechservices.net) |
| 03:07.37 | *** join/#asterisk xmitter (~xmitter@c-24-21-213-242.hsd1.or.comcast.net) |
| 03:07.41 | Rug | Is the TDM410 well supported? |
| 03:14.18 | Katty | Qwell: come find me on redditroulette.appspot.com |
| 03:21.24 | [TK]D-Fender | Rug: Yes |
| 03:23.58 | *** join/#asterisk came0 (~came0@167.83.189.72.cfl.res.rr.com) |
| 03:24.19 | *** join/#asterisk Tech_Travis (~Administr@cpe-76-168-191-127.socal.res.rr.com) |
| 03:25.45 | *** part/#asterisk stallione (~stallione@c-67-173-76-139.hsd1.il.comcast.net) |
| 03:31.17 | *** join/#asterisk thehar (thehar@thehar.xmission.com) |
| 03:32.40 | *** join/#asterisk nighty^ (~nighty@210.188.173.245) |
| 03:37.20 | *** join/#asterisk jmcdowell (~nooe@174-156-212-250.pools.spcsdns.net) |
| 03:37.25 | jmcdowell | hello all.. |
| 03:37.59 | jmcdowell | does anyone know anyone here have any experience with the 000000000000-directory.xml file? |
| 03:40.41 | *** join/#asterisk vrtigo1 (~fweston@86.9.205.68.cfl.res.rr.com) |
| 03:40.54 | voipmonk | what do u want to do? |
| 03:42.09 | vrtigo1 | Hi All. I'm getting an all circuits busy msg on 1.6.2.2 using Vitelity as my ITSP when trying to place any outbound calls. Inbound works fine, any suggestions? |
| 03:42.26 | voipmonk | whats your debug say , vrtigo1 ? use pastebin.ca |
| 03:42.32 | Mango | correct username/password? |
| 03:44.01 | vrtigo1 | http://pastebin.ca/1799431 |
| 03:44.13 | jmcdowell | Voipmonk : I am trying to make "presence" work. |
| 03:44.14 | *** join/#asterisk Micc (~quassel@c-98-225-57-96.hsd1.wa.comcast.net) |
| 03:44.30 | voipmonk | ugh - cant help you much there :) |
| 03:44.35 | voipmonk | sorry :) |
| 03:44.38 | [TK]D-Fender | vrtigo1: -- Executing [13862746255@internal:3] Dial("SIP/ata1-line1-00000012", "SIP/$\{EXTEN}@vitel-outbound") in new stack |
| 03:44.39 | vrtigo1 | Mango: if you were speaking to me, I am using IP auth so no user/password in sip |
| 03:44.42 | voipmonk | good luck in your search |
| 03:44.50 | jmcdowell | Caller id hacking to make the NAME and NUMBER show up? |
| 03:44.53 | voipmonk | nice \'s |
| 03:44.58 | [TK]D-Fender | vrtigo1: You have a "\" in between your ${EXTEN} |
| 03:45.02 | Micc | Is there any way for asterisk to tell me when audio breaks up? Like if the jitter buffer isn't enough and the packets fall outside the jitter buffer times? |
| 03:45.08 | [TK]D-Fender | vrtigo1: Fix the typo and it should be fine |
| 03:45.17 | Mango | [TK]D-Fender is right. |
| 03:45.21 | voipmonk | Micc: you're going to need to build a system that does that. |
| 03:45.32 | vrtigo1 | Thanks, I'll correct that and give it a try. |
| 03:46.33 | vrtigo1 | Perfect, thanks! |
| 03:46.54 | Micc | voipmonk, all the information is in asterisk to do that, don't you think? I just need to add the code to check it? |
| 03:48.05 | [TK]D-Fender | Micc: I'm sure there are places you could add hooks for where jitter gets compensated... you might want to be careful of course that your data load doesn't start becoming its own problem. |
| 03:49.24 | Micc | right, it would need to be a really fast check. |
| 03:49.46 | Micc | and maybe set a flag for reading and output later. |
| 03:53.08 | voipmonk | hrmmm |
| 03:53.15 | voipmonk | i would love to chat |
| 03:53.28 | jmcdowell | Is there a bit that one can flip to prevent asterisk from "hanging up" if the extension being transfered to is offline ? |
| 03:53.29 | voipmonk | plane to catch in the morn |
| 03:56.30 | ManxPower-work | jmcdowell: yes. |
| 03:56.47 | ManxPower-work | That would be "whatever your dialplan does after the Dial will happen. |
| 03:57.11 | jmcdowell | interesting.. |
| 03:57.24 | ManxPower-work | technically that would be "many buts" hoever. |
| 03:57.28 | ManxPower-work | ..er..however. |
| 03:57.35 | *** join/#asterisk Iamnacho (~Iamnacho@ip174-70-139-208.ks.ks.cox.net) |
| 03:57.35 | [TK]D-Fender | hoe! |
| 03:57.44 | Katty | down |
| 03:57.45 | jmcdowell | So, the dial plan... |
| 03:57.58 | ManxPower-work | Remember Asterisk isn't really a PBX. Asterisk is a TOOLKIT that lets you build a PBX, and that includes writing all the dialplan, fail over, call routing, etc |
| 03:58.00 | jmcdowell | I am tired and not thinking very clearly. |
| 03:58.07 | ManxPower-work | jmcdowell: Clearly. |
| 03:58.29 | ManxPower-work | Take a break. This stuff is not easy. It will never be easy. |
| 03:58.48 | jmcdowell | It's easier than it used to be.. |
| 03:59.08 | jmcdowell | But you are correct, it's still not that easy... |
| 04:04.31 | norrec | where can i get an default manager.conf files? |
| 04:04.33 | norrec | file* |
| 04:05.42 | [TK]D-Fender | norrec: in the /configs folder in your source tarball |
| 04:06.57 | carrar | source tarball!!! Whats that |
| 04:07.16 | ChannelZ | Where all the tar comes from |
| 04:07.37 | carrar | La Brea Tar Pits |
| 04:19.09 | *** join/#asterisk xmitter (~xmitter@c-24-21-213-242.hsd1.or.comcast.net) |
| 04:25.18 | *** join/#asterisk maxagaz (~maxagaz@soho2.i-xanadu.com) |
| 04:31.48 | *** join/#asterisk grind (~somebody@203.185.208.180) |
| 04:32.27 | grind | hey guys, im getting a " SIP/2.0 489 Bad event" from one of my external linksys phones. Anyone come accross this before? |
| 04:58.20 | carrar | What version of * are you running |
| 04:58.42 | carrar | possibly enable NAT keep alive |
| 05:02.07 | carrar | make sure you are not blocking anything |
| 05:02.57 | Mango | I get that. |
| 05:03.06 | Mango | But everything works, so I've never bothered to troubleshoot it |
| 05:04.26 | carrar | Asterisk probably doesn't know what to do with your keep alive NOTIFY |
| 05:04.48 | *** join/#asterisk chilicuil (~sistemas@unaffiliated/chilicuil) |
| 05:07.39 | *** join/#asterisk mikaeel (~dotirc@91.184.98.5) |
| 05:07.52 | mikaeel | Good day folks |
| 05:08.14 | Mango | Hey mikaeel |
| 05:08.19 | mikaeel | question, I've compiled asterisk with open h323 |
| 05:08.24 | vrtigo1 | somewhat OT - does anyone know how to tell what firmware is loaded on an ata-186 (sip, mgcp, etc)? |
| 05:08.26 | mikaeel | and now is problem |
| 05:08.29 | mikaeel | Don't know any of 0x0 formats |
| 05:09.09 | mikaeel | I can't call on sip extensions |
| 05:09.19 | carrar | vrtigo1, open a browser to it |
| 05:09.30 | carrar | all the stats are in web form |
| 05:09.47 | mikaeel | and applications does not work, what is may be? |
| 05:09.48 | *** join/#asterisk ltd_wk (~z@sixified.transact.net.au) |
| 05:10.25 | vrtigo1 | carrar, i see the web UI but it doesn't show me anything about the installed f/w. it has some references to sip but i can't seem to get it working with *, so wondering if maybe i have the wrong f/w |
| 05:11.23 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 05:14.23 | mikaeel | seems this is problem in codec |
| 05:14.27 | mikaeel | am I right? |
| 05:14.34 | carrar | just override whatever is on it |
| 05:14.42 | carrar | and the latest sip for the ATA |
| 05:14.47 | carrar | ^install |
| 05:29.31 | sbrath | Can I configure a BLF key to grab a Queue call that's doing rrmemory regardless of which extension is't ringing... |
| 05:29.35 | sbrath | ? |
| 05:30.51 | *** join/#asterisk JAMMAN2110 (~james@unaffiliated/jamman2110) |
| 05:31.46 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-wpotujfwwmfyvyii) |
| 05:50.24 | *** join/#asterisk pmhaddad (~Phil@71-13-218-72.dhcp.mrqt.mi.charter.com) |
| 05:51.05 | ManxPower-work | Queues generally ring devices, not extensions |
| 05:52.25 | b14ck | Sup all. |
| 05:52.43 | sbrath | ok, so the queue call comes in, and rings a SIP/1234 device, and then when it dosen't answer it rings SIP/2345, and so on for about 5 extensions. |
| 05:53.27 | sbrath | Can I code a BLF to flash when any call is ringing in the queue, and have a non-member of the queue ** the call ? |
| 05:53.38 | sbrath | I know it's stupid, but I thought I'd ask. |
| 05:55.05 | sbrath | It's kind of like, the Queue rings the people in cservice, but if for some reason they are all away, and forgot to pull out of queue, can someone else have a BLF flash that their is a call in the queue, ringing, and they can grab it.. |
| 05:56.59 | ChannelZ | Won't Pickup() pick up any ringing channel you tell it? |
| 06:01.28 | *** join/#asterisk smooth_penguin (~smoove@59.95.4.92) |
| 06:02.37 | sbrath | Since their are like 6 devices in the roundrobin, they'd have to Pickup(${EXTEN}) I believe... I guess I could setup the BLF to monitor when their are people in the Queue and no agents taking it, and when they press the button it logs them in as an agent, and they'd get the next call then... ? |
| 06:03.11 | *** part/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 06:05.00 | ChannelZ | wtf is the BLF |
| 06:15.25 | sbrath | BLF is Busy Lamp something... |
| 06:15.59 | sbrath | It's the term for the lights on the phone that can flash when other peoples extensions are ringing, etc. |
| 06:18.21 | *** join/#asterisk Faithful (~Faithful@ns.linuxterminal.com) |
| 06:25.25 | *** join/#asterisk ManxPower-work (~EWieling@216.186.151.147) |
| 06:25.30 | *** join/#asterisk pawz (~pawz@ppp118-208-171-146.lns20.bne4.internode.on.net) |
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| 06:35.19 | p3nguin | ~blf |
| 06:35.20 | infobot | blf is probably Busy Lamp Field, aka little lights next to speed dials that light up when the person is on the phone and blink when that line is ringing. hint extensions are static mapped to SIP or other channels. |
| 06:37.35 | *** join/#asterisk JAMMAN2110 (~james@unaffiliated/jamman2110) |
| 06:38.21 | sbrath | thanks infobot++ |
| 06:38.21 | infobot | gern geschehen, sbrath |
| 06:38.26 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 06:40.28 | *** join/#asterisk mikkel (~mikkel@130.226.36.170) |
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| 06:55.56 | mikaeel | folks, ztdummy is now unsupported? |
| 06:56.40 | mikaeel | can anybody tell me version of ztdummy that can i use for * 1.4? |
| 06:57.34 | *** join/#asterisk dangerdan (~irssi@unaffiliated/dangerdan) |
| 06:58.37 | [TK]D-Fender | mikaeel: Latest Zaptel, Latest * release. |
| 06:58.39 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 06:58.49 | *** join/#asterisk Caplain (shayne@shayne.caplain.loves.boys.fbi.gov.silverelitez.org) |
| 06:58.53 | dangerdan | Has anyone been able to get Skype for Asterisk to work on Ubuntu 9.10? |
| 07:00.25 | [TK]D-Fender | alrighty... checkout time. Later all. |
| 07:00.40 | mikaeel | [TK]D-Fender: i need to use * 1.4.8 , but where can I download ztdummy? |
| 07:01.48 | drmessano | ztdummy is part of Zaptel |
| 07:02.28 | drmessano | Zaptel is old.. It's been superceded by Dahdi |
| 07:04.12 | ChannelZ | dangerdan: I'm running it on 8.04 but I don't know why it would be any different |
| 07:08.00 | mikaeel | drmessano, thnx |
| 07:08.39 | mikaeel | and seond question, Dahdi's version depends on version of asterisk? |
| 07:08.51 | mikaeel | or i need to use latest release? |
| 07:10.34 | drmessano | Always latest release.. it works on 1.4.21 and higher (I think) and all of the 1.6.x.x series |
| 07:11.22 | kaldemar | dahdi came along with asterisk 1.4.22, so with 1.4.8 you'd go with the newest version of zaptel. |
| 07:12.07 | mikaeel | thanx folks! |
| 07:12.24 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 07:12.27 | *** join/#asterisk smooth_penguin (~smoove@59.95.4.92) |
| 07:15.17 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 07:24.13 | *** join/#asterisk Heretic (~Fallen@dsl-246-64-44.telkomadsl.co.za) |
| 07:27.58 | *** join/#asterisk e-jones (~jkastner@nat/redhat/x-oyjpmlbaheslfuuh) |
| 07:30.39 | mikaeel | :( same problem : WARNING[19587]: channel.c:718 ast_best_codec: Don't know any of 0x0 formats |
| 07:32.06 | *** join/#asterisk pmhaddad (~Phil@71-13-218-72.dhcp.mrqt.mi.charter.com) |
| 07:32.34 | kaldemar | mikaeel: when do you get that? |
| 07:33.52 | benngard | asterisk[14260]: segfault at 8 ip b7c9caf8 sp b77ff1ac error 6 in libc-2.7.so :( |
| 07:34.00 | *** join/#asterisk binbash_ (~peter@ip4da53781.direct-adsl.nl) |
| 07:34.25 | mikaeel | kaldemar: in cli |
| 07:34.33 | mikaeel | Avaya > asterisk |
| 07:35.00 | kaldemar | when, not where |
| 07:35.15 | mikaeel | aah, when i'm clling to asterisk |
| 07:35.20 | mikaeel | *calling |
| 07:35.26 | mikaeel | on sip extension |
| 07:35.31 | mikaeel | or in meetme |
| 07:35.40 | kaldemar | is that coming from the ooh323 side? |
| 07:36.03 | benngard | kaldemar: asking me? |
| 07:36.09 | mikaeel | Open H323 |
| 07:36.18 | mikaeel | yeah |
| 07:36.34 | benngard | yes, i did get the message when i received a fax |
| 07:36.43 | kaldemar | benngard: no |
| 07:37.03 | kaldemar | mikaeel: have you allowed codecs in the config file? |
| 07:37.47 | mikaeel | yes, i use alaw and ulaw, in avaya is g711 |
| 07:37.54 | mikaeel | both g711 alaw and ulaw |
| 07:38.25 | kaldemar | feel free to pastebin the config and a CLI output of a whole call. |
| 07:38.28 | benngard | mikaeel: what kind of avaya are u using cm? |
| 07:39.00 | mikaeel | g8700 |
| 07:39.10 | mikaeel | cm 3 |
| 07:39.21 | mikaeel | there is ok everything |
| 07:39.50 | mikaeel | when was the ooh323 incoming calls to asterisk works fine |
| 07:40.06 | mikaeel | but outgoing from asterisk to avaya does not |
| 07:40.31 | mikaeel | and now I changed the library of protocol on openh323 |
| 07:40.35 | benngard | mikaeel: i have a setup as: pstn - avaya cm3 - ooh323 - asterisk - sip phones and sip ata's |
| 07:42.14 | benngard | mikaeel: u want me to pastebin my ooh323.conf? |
| 07:45.09 | *** join/#asterisk MAbbas (~abbas4u@203.215.177.194) |
| 07:45.25 | *** part/#asterisk Tech_Travis (~Administr@cpe-76-168-191-127.socal.res.rr.com) |
| 07:45.31 | *** join/#asterisk mpe (~mpe@94.127.49.1) |
| 07:45.38 | mikaeel | benngard: pls if this is not problem for you |
| 07:45.52 | mikaeel | can you send me on mail? |
| 07:47.54 | MAbbas | Hi All, I am listening AMI events, is there an event for when agent's "wrapuptime" is complete? |
| 07:51.53 | *** join/#asterisk e-jones (~jkastner@nat/redhat/x-qviyimucwayiwuqm) |
| 08:06.55 | *** join/#asterisk icyValk77 (~icyValk77@gateway.ash.thebunker.net) |
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| 08:16.01 | *** join/#asterisk drcode (~c7cbb864@gateway/web/freenode/x-ylcghbuqurcivqhx) |
| 08:16.04 | drcode | hi all |
| 08:16.28 | drcode | I am looking for record software for win32 that can record gsm6.10 |
| 08:20.33 | *** join/#asterisk Tim_Toady (~moi@77.49.178.169.dsl.dyn.forthnet.gr) |
| 08:24.54 | *** join/#asterisk FloZix (~florian@vie38-3-88-170-254-128.fbx.proxad.net) |
| 08:24.57 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 08:25.03 | *** join/#asterisk c0rnoTa (~c0rnoTa@94.25.175.235) |
| 08:28.11 | *** part/#asterisk slashtom (~tom@k-rad.co.uk) |
| 08:28.31 | *** join/#asterisk scurb (~scurb@194.218.238.2) |
| 08:37.57 | Sargun | perfectly valid question: http://clientsfromhell.tumblr.com/post/385623537/we-dont-like-the-look-of-our-phone-number-can |
| 08:40.29 | carrar | YES |
| 08:40.33 | carrar | YES WE CAN |
| 08:41.17 | carrar | Just dial 1 |
| 08:41.21 | carrar | from ANYWHERE |
| 08:42.27 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 08:43.12 | *** join/#asterisk mpe (~mpe@gate.ipvision.dk) |
| 08:43.23 | *** join/#asterisk e-jones (~jkastner@nat/redhat/x-irsneauovbnpkgzi) |
| 08:45.00 | *** join/#asterisk doolittlework (~d@196.211.34.2) |
| 08:45.31 | doolittlework | can someone please inlighten me how to link crm and asterisk |
| 08:49.40 | carrar | Anyway you want too |
| 08:50.17 | carrar | not gonna stick around to find out |
| 08:56.57 | *** join/#asterisk aceking5 (~aceking5@71-94-132-102.static.mtpk.ca.charter.com) |
| 09:01.50 | *** join/#asterisk ktwilight[m] (~ktwilight@212.223-201-80.adsl-dyn.isp.belgacom.be) |
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| 09:18.15 | Dovid | does asterisk support 302 by default on outbound calls to a peer ? |
| 09:28.23 | *** join/#asterisk Akiraa (~Akira@92.81.201.94) |
| 09:35.45 | *** join/#asterisk spenguin[work] (~penguin@59.162.86.164) |
| 09:35.48 | spenguin[work] | test |
| 09:35.58 | Gido-E | 123 |
| 09:36.12 | spenguin[work] | :> |
| 09:51.36 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 09:53.33 | c0rnoTa | Hello all |
| 09:53.54 | c0rnoTa | can anybody tell me, what's mean chan_iax2.c: Raw Hangup... ip:port .... |
| 09:54.02 | c0rnoTa | what does it mean? |
| 09:54.32 | *** join/#asterisk Jak_o_Shadows (~Fake@121.209.35.170) |
| 09:55.13 | Jak_o_Shadows | I haven't really understood it much, but is it possible for a computer (asterisk server) to recieve calls made to the POTS? |
| 09:55.52 | *** join/#asterisk dezent (dezent@84.243.58.35) |
| 09:57.57 | Tim_Toady | Jak_o_Shadows yes, you just need a card to connect ur server to the telephone line |
| 09:58.03 | *** join/#asterisk xmitter (~xmitter@c-24-21-213-242.hsd1.or.comcast.net) |
| 09:58.09 | Tim_Toady | a card with an fxo port |
| 09:58.18 | Jak_o_Shadows | would a normal modem have one of these |
| 09:58.27 | Jak_o_Shadows | i know the page, but it doesn't give much info |
| 09:58.27 | Tim_Toady | yes but its not gonna work |
| 09:58.53 | Tim_Toady | thers special hardware for asterisk, modems will not work |
| 09:58.58 | Jak_o_Shadows | however, you can connect a phone to one of the ports on the internal modem, and then another from modem to wall, and use the phone when not dialuped |
| 09:59.42 | Tim_Toady | this is just a bridged port |
| 10:00.07 | Jak_o_Shadows | thought it'd be somethine like that |
| 10:00.10 | Tim_Toady | but has nothing to do with asterisk |
| 10:00.32 | Jak_o_Shadows | yeah, sadly |
| 10:00.54 | Jak_o_Shadows | whats the issue blocking that? |
| 10:00.59 | *** join/#asterisk radcliff (~radcliff@h-63-22.A259.priv.bahnhof.se) |
| 10:01.07 | Tim_Toady | drivers i guess |
| 10:01.18 | *** join/#asterisk e-jones (~jkastner@nat/redhat/x-odyhzihbxoddjioq) |
| 10:01.34 | Jak_o_Shadows | having a quick google, that question seems pretty common |
| 10:01.46 | Jak_o_Shadows | hard to believe that no attempt has been made to fix it, software wise |
| 10:01.48 | Tim_Toady | anyway get a X100P card with a single fxo, its the cheapset you can get (20-30$) an its a good start to start playing with * |
| 10:02.03 | Tim_Toady | its not top quality but its ok for home use |
| 10:02.10 | Jak_o_Shadows | nah, i think i'll just try the voip by itself |
| 10:02.44 | *** join/#asterisk Bladerunner05 (~bladerunn@81-174-56-54.static.ngi.it) |
| 10:02.50 | *** join/#asterisk slima (slima@unaffiliated/slima) |
| 10:03.34 | Bladerunner05 | hello my asterisk box have as gateway my adsl router and phone too. if adsl go down my phone didn't works... how may I resolve this ? |
| 10:04.00 | *** join/#asterisk Koma_IT (~Koma@78.6.25.209) |
| 10:04.05 | Koma_IT | Hi all |
| 10:04.18 | Jacke | <PROTECTED> |
| 10:04.31 | Mark22 | Bladerunner05: what OS do you use? |
| 10:04.43 | Bladerunner05 | Mark22: debian |
| 10:05.11 | Mark22 | as it is more a question about the OS, you could look to the route options in Debian |
| 10:05.24 | Mark22 | as you want a "fallback" gateway |
| 10:06.14 | Bladerunner05 | Mark22: yes, how to do ? |
| 10:06.24 | Dovid | does asterisk support 302 by default on outbound calls to a peer ? |
| 10:06.34 | radcliff | Hi, I am going nuts over Asterisk Realtime dynamic SIP-peers (In this case my SIP-provider). They keep disappearing from Realtime Cache and the only way to get them back is to make a call out trough them and they are in the cache for a little while more. "sip show peer SIP-PROVIDER load" does not seem to do the trick either, is it supposed to work like this? What is the meaning of realtime dynamic SIP peers if incoming calls does not work |
| 10:06.35 | radcliff | outgoing calls are not made at regular intervalls? Is it a bug or a feature? :) |
| 10:07.18 | Mark22 | Bladerunner05: ask in #debian , to be honest I didn't see it anywhere in the past working correctly |
| 10:07.35 | Bladerunner05 | Mark22: thanks |
| 10:07.40 | Bladerunner05 | join #debian |
| 10:08.27 | *** join/#asterisk nicknick (~administr@host213-123-201-13.in-addr.btopenworld.com) |
| 10:08.29 | *** part/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 10:08.42 | Koma_IT | Hi all, i've an Asterisk Appliance, i can't log in via SSH but i've access trought the port 5038. How can i check the Asterisk Version?. How can i see the log? |
| 10:10.39 | tzafrir_laptop | Koma_IT, do you have a manager interface user/pass? |
| 10:10.52 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 10:11.36 | tzafrir_laptop | Bladerunner05, maybe a local caching DNS server like dnsmasq may help |
| 10:11.43 | *** join/#asterisk Koma_IT (~Koma@78.6.25.209) |
| 10:11.48 | *** join/#asterisk garymc (~chatzilla@host81-139-136-16.in-addr.btopenworld.com) |
| 10:11.49 | Koma_IT | sorry |
| 10:11.51 | Koma_IT | Hi all, i've an Asterisk Appliance, i can't log in via SSH but i've access trought the port 5038. How can i check the Asterisk Version?. How can i see the log? |
| 10:11.58 | tzafrir_laptop | Koma_IT, do you have a manager interface user/pass? |
| 10:12.05 | Koma_IT | yes |
| 10:12.13 | Koma_IT | but is completely custom |
| 10:12.23 | Koma_IT | web interface |
| 10:12.28 | tzafrir_laptop | you can basically issue any CLI command via the event 'Command' |
| 10:12.36 | tzafrir_laptop | Event: command |
| 10:12.42 | tzafrir_laptop | Command: core show version |
| 10:13.01 | radcliff | Action: command ? |
| 10:13.17 | Koma_IT | but i cannot call any command from this interface |
| 10:13.20 | Koma_IT | is a configuration only |
| 10:13.26 | Koma_IT | customized by seller |
| 10:13.48 | c0rnoTa | Hay, guys. May i ask you annother question? What means "chan_dahdi.c Write returned -1 Resource temporarily unavailable"& |
| 10:13.58 | radcliff | Koma_IT: But if you telnet to port 5038 and do this trough asterisk manager |
| 10:13.59 | Nugget | telnet is eeeeeeevil! |
| 10:14.05 | Koma_IT | wow! |
| 10:14.16 | Koma_IT | there is a gui to make this commands? |
| 10:14.30 | Koma_IT | i mean telnet could be VERY VERY dangerous :D |
| 10:14.30 | radcliff | Koma_IT: no, use telnet |
| 10:14.40 | Koma_IT | ok |
| 10:14.42 | *** join/#asterisk KermitTheFragger (~KermitThe@118-197.bbned.dsl.internl.net) |
| 10:14.51 | tzafrir_laptop | http://www.voip-info.org/wiki/view/Asterisk+manager+API |
| 10:15.37 | tzafrir_laptop | If telnet is evil because commands are sent as plain text, then so will be any GUI sending those commands unencrypted to port 5038 |
| 10:15.43 | Koma_IT | thx a lot :) |
| 10:15.59 | tzafrir_laptop | GUIs are evil :-) |
| 10:16.03 | *** join/#asterisk soman (~somnath@stargate.starnet.fi) |
| 10:16.36 | Koma_IT | lol |
| 10:16.52 | tzafrir_laptop | c0rnoTa, is this a debug message? |
| 10:17.06 | Koma_IT | i try |
| 10:17.07 | Koma_IT | <PROTECTED> |
| 10:20.09 | c0rnoTa | tzafrir_laptop: yes, it is. But this call ended with voice drop from caller on DAHDI (silence in callee side, and callee hangup call because he couldn't hear anything.) In logs all looks clean, but there is some " Resource temporarily unavailable" messages. |
| 10:20.14 | *** join/#asterisk correcaminos (~laguilar@201.201.46.106) |
| 10:20.42 | *** join/#asterisk af_ (~getsmart@88-149-230-64.dynamic.ngi.it) |
| 10:20.46 | Koma_IT | Response: Error |
| 10:20.46 | Koma_IT | Message: Missing action in request |
| 10:20.50 | tzafrir_laptop | c0rnoTa, dahdi regularily has those in man cases |
| 10:22.38 | c0rnoTa | tzafrir_laptop: so, it shouldn't be the root of issue, right? |
| 10:23.01 | tzafrir_laptop | I guess so |
| 10:23.13 | c0rnoTa | ок, thx |
| 10:23.14 | *** join/#asterisk z53102 (~nikolay@208.89.107.74) |
| 10:23.21 | tzafrir_laptop | Koma_IT, my mistake: |
| 10:23.27 | *** join/#asterisk RoyK (~RoyKa@nilugw1.nilu.no) |
| 10:23.32 | tzafrir_laptop | Action Command |
| 10:23.33 | RoyK | looks around |
| 10:23.37 | tzafrir_laptop | grr... |
| 10:23.40 | tzafrir_laptop | Action: Command |
| 10:23.45 | Koma_IT | ok :D |
| 10:23.45 | tzafrir_laptop | Command: core show version |
| 10:24.09 | tzafrir_laptop | RoyK, hi |
| 10:24.15 | RoyK | long time |
| 10:24.39 | RoyK | I just talked to this one guy running asterisk, or trying to, on opensolaris, so I thought I'd just walk by :) |
| 10:25.25 | radcliff | Hi, I am going nuts over Asterisk Realtime dynamic SIP-peers (In this case my SIP-provider). They keep disappearing from Realtime Cache and the only way to get them back is to make a call out trough them and they are in the cache for a little while more. "sip show peer SIP-PROVIDER load" does not seem to do the trick either, is it supposed to work like this? What is the meaning of realtime dynamic SIP peers if incoming calls does not work |
| 10:25.25 | radcliff | outgoing calls are not made at regular intervalls? Is it a bug or a feature? :) |
| 10:25.47 | *** part/#asterisk Heretic (~Fallen@dsl-246-64-44.telkomadsl.co.za) |
| 10:26.19 | c0rnoTa | tzafrir_laptop: may be you can tell what means "chan_iax2.c: Raw Hangup... ip: port .... "? |
| 10:27.04 | tzafrir_laptop | c0rnoTa, not sure. Usually with a high enough debug level you should see the initial cause for the hangup |
| 10:27.29 | tzafrir_laptop | RoyK, so what're you doing these days? |
| 10:29.06 | *** join/#asterisk p3nguin_ (gpz5GvdFkf@9xf6gbZ.a2infotech.com) |
| 10:30.26 | Dovid | does asterisk support 302 by default on outbound calls to a peer ? |
| 10:30.34 | c0rnoTa | tzafrir_laptop: i have debug level 6. looks like it's not enough :) so, my issue with voice drop may be solved in higher asterisk version. I have 1.4.26.1 on IAX side, and 1.4.27 on DAHDI side. |
| 10:30.38 | RoyK | tzafrir_laptop: mostly storage stuff, and general operations. anything from fixing old DOS boxes connected to hilariously priced instruments to managing opensolaris storage pools |
| 10:30.55 | tzafrir_laptop | c0rnoTa, I'm not sure there are any messages with a higher debug value |
| 10:34.35 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 10:38.19 | *** join/#asterisk cnu (cnu@the.ultimate.lamer.la) |
| 10:42.11 | *** join/#asterisk Bladerunner05 (~bladerunn@81-174-56-54.static.ngi.it) |
| 10:43.45 | Bladerunner05 | hello my asterisk box have as gateway and nameserver my adsl router and my phones too. if adsl go down, I can't call my lan's phones and If someone from outside call asterisk it can't call my lan's phone and in cli> I notice app_dial.c:1049 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) ... how may I resolve this ? |
| 10:54.33 | *** join/#asterisk davix (~bo@corky.net) |
| 10:55.06 | davix | I have made a change in my network (changed ip addresses) and now extensions are not getting calls, they can make calls and all good |
| 10:55.16 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 10:55.16 | davix | sip show registery is emptey, but sip show peers shows them |
| 10:55.22 | *** join/#asterisk UQlev (~yuriy@nb11-125.static.cytanet.com.cy) |
| 10:55.45 | davix | any directions on where I should look |
| 11:00.02 | *** join/#asterisk krynnotaur (~krynnotau@unaffiliated/krynnotaur) |
| 11:01.08 | drcode | hi all |
| 11:01.32 | drcode | how can I record wav with gsm 6.10? |
| 11:05.32 | pentanol | hello :) |
| 11:06.38 | pentanol | I've seen there someone who use asterisk with radius, but din't remember his\her nickname, I guess your there rightnow? |
| 11:11.41 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 11:13.45 | *** join/#asterisk Koma_IT (~Koma@78.6.25.209) |
| 11:18.08 | c0rnoTa | pentanol: CDR_RADIUS? |
| 11:20.00 | pentanol | c0rnoTa that module ony for calls-statistic collecting, I'm looking for auth |
| 11:20.14 | pentanol | or |
| 11:20.31 | pentanol | actually my trouble with radius side... |
| 11:20.50 | pentanol | I can't enquire password from database.... |
| 11:21.37 | c0rnoTa | i don't know about auth over radius. Can't help you, man |
| 11:21.56 | pentanol | may I make it just via radius attributs and accounting? |
| 11:25.23 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 11:27.10 | tzafrir_laptop | pentanol, does it have to be real-time, or can you extract configuration from the Radius server at reload time with a script? |
| 11:27.25 | tzafrir_laptop | (I guess not, but worth asking) |
| 11:27.26 | *** join/#asterisk spenguin[work] (~penguin@59.162.86.164) |
| 11:28.25 | pentanol | with a script we avoid authorization or do it quite unconvinuent |
| 11:28.56 | pentanol | because if we use a script we sould maintain account in asterisk and database |
| 11:29.01 | atis_work | Corydon76-dig: you seem to have fixed memory leak in 1.6.2 app_meetme with r145649 however it's still present in 1.6.1 :p |
| 11:29.45 | atis_work | Corydon76-dig: var = ast_load_realtime("meetme", "confno", confno, NULL); |
| 11:30.04 | atis_work | Corydon76-dig: var = var->next; ast_variables_destroy(var); |
| 11:33.27 | atis_work | oops, wrong channel :p |
| 11:33.45 | *** join/#asterisk doolittlework (~d@196.211.34.2) |
| 11:37.10 | doolittlework | hi there need some help I bought a te110p card and can not seem to get it to work, i know you guys are running 1.6 already, but i am testing VICIDIAL and i need this card to work with asteriks 1.2.30.2, can anyone point me in the right direction, i can see the card cat/proc/zaptel 1, am sure my zaptel and zapate is correct, ztcfg -vv says all my channels are configured, but the command pri show span 1 shows nothing |
| 11:39.33 | *** join/#asterisk Rajmohan (~raj@122.164.184.3) |
| 11:41.35 | Rajmohan | hi, iam getting Registration error 408 |
| 11:41.40 | Rajmohan | in xlite configuration |
| 11:41.44 | Rajmohan | can any one help |
| 11:42.15 | doolittlework | see if u can ping your server from the pc where xlite is installed Rajmohan |
| 11:45.14 | doolittlework | how does one check witch version zaptel u using |
| 11:46.39 | *** join/#asterisk davidded (foobar@dslb-084-059-192-031.pools.arcor-ip.net) |
| 11:46.47 | davidded | hello |
| 11:47.11 | doolittlework | hi |
| 11:47.37 | davidded | i hate to come here and just ask a technical question, but would that be ok? |
| 11:48.05 | doolittlework | we can only try to help |
| 11:48.36 | davidded | my objective is to do something "the way it is supposed to be", when implementing a conference feature for chan-sccp-b (maybe you know the channel driver) |
| 11:49.29 | doolittlework | do you need the driver? |
| 11:49.40 | davidded | no, i am the project maintainer :) |
| 11:49.43 | davidded | but the point is, |
| 11:50.03 | davidded | if i create new ast_channels and want to masquerade existing channels to those new channels |
| 11:50.18 | davidded | am i suppose to create a monitoring thread that resembles the functionlity of the pbx thread? |
| 11:50.54 | doolittlework | are you working with cisco phones? |
| 11:50.57 | davidded | yes |
| 11:51.37 | doolittlework | sorry buddy never used cisco before, checkout sourgeforce they have some interesting docs there |
| 11:51.38 | Rajmohan | doolittlework: hi, iam able to ping the server |
| 11:51.41 | Rajmohan | from my pc |
| 11:51.45 | Rajmohan | where xlite is installed |
| 11:52.16 | davidded | actually, my question is not related to the phones |
| 11:52.30 | davidded | what we want to do is the following: |
| 11:52.33 | doolittlework | Rajmohan check your username and password also check "sip show peers"command within asteriks CLI |
| 11:52.56 | davidded | we have one or more bridged channels and want to add them to a conference bridge |
| 11:53.05 | davidded | as in asterisk's main/bridging.c |
| 11:53.36 | davidded | so what we are looking for would be how to un-bridge a channel without hanging it up |
| 11:53.47 | Rajmohan | iam new to this |
| 11:53.51 | Rajmohan | how and where do i check it |
| 11:54.02 | Rajmohan | in my freepbx i created the sip extension 101 |
| 11:54.06 | Rajmohan | and password as raj |
| 11:54.16 | Rajmohan | that is what iam using to configure xlite |
| 11:54.30 | doolittlework | this is not a freepbx channel, hold on will help u now |
| 11:55.00 | Rajmohan | ok pls |
| 11:55.10 | davidded | one way to go about was to allocate a new ast_channel channel for each leg and add these channels to the conference bridge after masquerading the original calling channels into those new channels. |
| 11:55.43 | doolittlework | in xlite you must have your username, password and autorization username match up with what u have in freepbx |
| 11:56.48 | doolittlework | davidded: do you want to move active calls to a conference? |
| 11:57.11 | davidded | yes |
| 11:57.44 | doolittlework | so u are using asterisk E1 and linking it to a cisco |
| 11:57.57 | davidded | in principle, yes |
| 11:58.05 | *** join/#asterisk catojo (~catojo@189.24.85.1) |
| 11:58.12 | doolittlework | where does the calls reside on the cisco or *? |
| 11:58.21 | davidded | doolittlework: we are the guys who write the current channel drivers for the cisco phones. |
| 11:58.27 | davidded | so the phones are served by asterisk directly |
| 11:58.31 | davidded | no call manager is involved |
| 11:58.41 | benngard | but could it be that "stupid", i have some spa2102 that use for connecting faxes to *, i had one spa that caused segfaults on *, did found out that i hadnt set any "Display Name" in that box, all other had, when i put some value there, it did stop to segfault... strange |
| 11:59.15 | davidded | benngard: sounds somebody left a null pointer there... |
| 12:00.03 | benngard | yupp |
| 12:00.04 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 12:00.09 | *** join/#asterisk Faithful (~Faithful@ns.linuxterminal.com) |
| 12:03.26 | benngard | i do hope it is the answer |
| 12:03.36 | davidded | doolittlework: any ideas how to setup a new channel without the pbx? |
| 12:03.48 | benngard | but i think that somebody should fix that issue |
| 12:05.17 | *** join/#asterisk m_c_le (~marcello@dslb-088-074-131-094.pools.arcor-ip.net) |
| 12:05.38 | Rajmohan | doolittlework: when i execute sip show peers, it gives no such command |
| 12:05.47 | *** join/#asterisk MAbbas (~abbas4u@203.215.177.194) |
| 12:06.47 | Rajmohan | doolittlework: when i execute sip show peers, it gives no such command in the asterisk cli |
| 12:07.17 | MAbbas | Hi All, I am listening for AMI events, Is there an AMI event which is fired when agents wrapuptime is completed? |
| 12:07.19 | doolittlework | Rajmohan: looks like your sip)chan is not loading, exit out of cli, and type amportal restart |
| 12:10.33 | Rajmohan | after restarting, i executed sip show peers, it executes |
| 12:11.09 | Rajmohan | but there it has Name/username host, dyn nat act port status |
| 12:11.13 | Rajmohan | and 0 peers |
| 12:11.22 | Rajmohan | 0 sip peers |
| 12:12.01 | [TK]D-Fender | Rajmohan: Does your sip.conf and included bits say it should be otherwise? |
| 12:13.03 | Rajmohan | after executing sip show peers the result i get is |
| 12:13.06 | Rajmohan | Name/username host, dyn nat act port status |
| 12:13.10 | Rajmohan | 0 sip peers |
| 12:14.36 | [TK]D-Fender | Rajmohan: Does your sip.conf and included bits say it should be otherwise? |
| 12:14.55 | MAbbas | AMI event for agent's wrapuptime complete? |
| 12:15.40 | Rajmohan | [TK]D-Fender: sorry i dont understand your question pls, iam a newbie, can you explain it for me |
| 12:16.13 | [TK]D-Fender | Rajmohan: Go look at your SIP.CONF CONFIG FILE.l Doe you even HAVE any peers that should be loaded? |
| 12:18.23 | adnc | is there a way to add a provider to my configuration which i can force to be used from the clients by pressing sort of *12# in front of my destination number? |
| 12:20.26 | [TK]D-Fender | adnc: It's your dialplan, do whatever you want. |
| 12:20.49 | adnc | i understand, but could you point me to an example on how to do this? |
| 12:20.55 | [TK]D-Fender | adnc: ... |
| 12:20.57 | [TK]D-Fender | ~book |
| 12:20.58 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 12:21.06 | [TK]D-Fender | adnc: "vi extensions.conf" |
| 12:21.29 | adnc | a book? |
| 12:21.45 | [TK]D-Fender | adnc: and a good 'ole command to go edit your dialplan |
| 12:22.11 | [TK]D-Fender | adnc: You want a coded prefix, go make a dialplan apttern for it |
| 12:22.15 | [TK]D-Fender | pattern* |
| 12:26.51 | Rajmohan | there is no peers in sip.conf |
| 12:27.10 | Rajmohan | but i added sip extension through freepbx admin |
| 12:27.29 | Rajmohan | it will automatically edit the sip.conf and set up the info? |
| 12:27.52 | [TK]D-Fender | Rajmohan: pastebin your entire sip.conf and EVERY file "INCLUDE"-d by it |
| 12:27.53 | [TK]D-Fender | ~pb |
| 12:27.54 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding 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/ , or apt-get install pastebinit |
| 12:27.55 | [TK]D-Fender | ^^^^^^^^^^^^^ |
| 12:28.14 | [TK]D-Fender | RajDon't forget you have to APPLY your changes and have it build the configs... |
| 12:28.46 | adnc | Rajmohan, is freepbx a frontend to asterisk configuration? |
| 12:29.10 | [TK]D-Fender | adnc: Yes, it is |
| 12:29.50 | adnc | do you know if freepbx will make use of my already existing configuration files? |
| 12:30.12 | Rajmohan | [TK]D-Fender: thank you, iam really a beginner, i did not apply the configuration changes |
| 12:30.22 | Rajmohan | xlite now shows my username:) |
| 12:30.24 | [TK]D-Fender | SMRT |
| 12:31.22 | adnc | [TK]D-Fender, would you recommend freepbx for beginners? |
| 12:31.34 | [TK]D-Fender | adnc: Beginner at what? |
| 12:31.43 | adnc | asterisk of course |
| 12:32.00 | [TK]D-Fender | adnc: If Asterisk is your goal, then no. |
| 12:32.22 | adnc | than i must have misunderstood freepbx |
| 12:33.45 | [TK]D-Fender | adnc: FreePBX is for people who jsut want a Magic-Clicky PBX builder. If you don't like the scope of things you can do with it because of its limited nature then you're in for a lot of work, or possible dead-end. |
| 12:34.14 | [TK]D-Fender | adnc: It teaches little execpt how NOT to build a dialplan yourself. |
| 12:34.18 | adnc | i understand, thank you, maybe i should keep continue configuration the files myself |
| 12:34.52 | [TK]D-Fender | adnc: The only reason to use a GUI for this is when you expect to have non-admins manage your PBX |
| 12:35.18 | Rajmohan | [TK]D-Fender: now i really want a anlalog phone system using asterisk |
| 12:35.33 | adnc | i did a basic configuration it works so far for one provider and an extension, i would like to add several more providers and dial plans/rules to use those providers when someone wants to call different countries |
| 12:35.35 | Rajmohan | i can setup extensions and call within office |
| 12:35.42 | [TK]D-Fender | Rajmohan: Feel free to elaborate as to what that goal means. |
| 12:35.56 | doolittlework | hi there need some help I bought a te110p card and can not seem to get it to work, i know you guys are running 1.6 already, but i am testing VICIDIAL and i need this card to work with asteriks 1.2.30.2, can anyone point me in the right direction, i can see the card cat/proc/zaptel 1, am sure my zaptel and zapate is correct, ztcfg -vv says all my channels are configured, but the command pri show span 1 shows nothing |
| 12:36.40 | Rajmohan | i mean, instead of using the xlite as sip client, i want to use analog phones, where i can call the extensions |
| 12:36.42 | [TK]D-Fender | doolittlework: pastebin your configs and "zap show channels" from CLI |
| 12:37.04 | [TK]D-Fender | Rajthat's fine. Plenty of devices to let you plug in phones. How many? |
| 12:37.30 | Rajmohan | around 5 extensions |
| 12:37.49 | [TK]D-Fender | Rajmohan: then get something like 3 x Linksys PAP2T-NA |
| 12:38.13 | [TK]D-Fender | Rajmohan: http://www.telephonydepot.com/Catalog/Dual-FXS-Analog-Adapters/Linksys-PAP2T-NA |
| 12:38.31 | Rajmohan | let me go over it thanks |
| 12:40.36 | *** join/#asterisk MAbbas (~abbas4u@203.215.177.194) |
| 12:42.25 | doolittlework | [TK]D-Fender: http://pastebin.com/m6b1b44e thx |
| 12:43.31 | [TK]D-Fender | doolittlework: channels => 1-15,17-31 <----- no "s" in "channel" |
| 12:43.53 | doolittlework | let me see |
| 12:44.04 | [TK]D-Fender | doolittlework: signalling=pri_cpe <--- 1 "l" in "signaling" |
| 12:44.37 | doolittlework | [TK]D-Fender is my hero |
| 12:44.47 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:44.47 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:44.49 | doolittlework | thanks dude you are a g |
| 12:44.56 | doolittlework | thank u |
| 12:45.22 | doolittlework | ive been looking at this the whole day |
| 12:45.47 | doolittlework | typo,typos will ruine one day |
| 12:47.03 | [TK]D-Fender | doolittlework: You're welcome. |
| 12:47.13 | [TK]D-Fender | doolittlework: And why is it you're using 1.2 in the first place? |
| 12:47.22 | *** join/#asterisk willianmazzardo (~willian@server2.sysvoip.com.br) |
| 12:47.27 | *** part/#asterisk willianmazzardo (~willian@server2.sysvoip.com.br) |
| 12:47.52 | adnc | when i do a reload in the asterisk cli i get warnings and errors like this WARNING[11566]: cdr_pgsql.c:297 my_load_module: Unable to load config for PostgreSQL CDR's: cdr_pgsql.conf |
| 12:47.59 | *** join/#asterisk caio1982 (~caio1982@CAcert-br/caio1982) |
| 12:48.04 | *** part/#asterisk caio1982 (~caio1982@CAcert-br/caio1982) |
| 12:48.07 | adnc | although i do not want these modules or use these modules |
| 12:48.22 | doolittlework | [TK]D-Fender: checking out vicidialnow, call center application |
| 12:48.27 | adnc | how can i disable them? i do not load them in my configs |
| 12:49.27 | [TK]D-Fender | adnc: "noload = > cdr_pgsql.so" <- add to modules.conf |
| 12:51.30 | adnc | [TK]D-Fender, thank you. i do have here http://pastebin.com/d6b20fc29 a list of the warnings/errors. i'm going to disable them like you showed above. but i wonder if there is anything i definately need |
| 12:52.56 | adnc | noload => pbx_dundi.so for example doesnt work |
| 12:54.51 | [TK]D-Fender | adnc: On full restart of *? |
| 12:55.05 | adnc | no, when i do reload in asterisk cli |
| 12:55.25 | [TK]D-Fender | adnc: not enough. "reload" reloads the configs for modules you HAVE loaded |
| 12:55.41 | [TK]D-Fender | adnc: So they're already loaded. Restart * |
| 12:55.48 | adnc | ok |
| 12:56.32 | adnc | [TK]D-Fender, this behaviour is because modules are autoloaded in the modules.conf at the beginning, right? |
| 12:56.43 | [TK]D-Fender | adnc: Which behavior? |
| 12:57.26 | adnc | that these modules are loaded and errors are shown when reload initiated |
| 12:58.12 | [TK]D-Fender | adnc: Perfectly normal. Module goes to load their configs and they;'re not there so it complains. |
| 13:03.36 | *** part/#asterisk m_c_le (~marcello@dslb-088-074-131-094.pools.arcor-ip.net) |
| 13:03.49 | tzafrir_laptop | [TK]D-Fender, actually "signaling" (signle 'l') is also legal in chan_dahdi.conf (though not encourged) |
| 13:05.13 | adnc | on my internal calls i do hear quite some noise in the background. what could cause this |
| 13:05.33 | *** join/#asterisk techie (~root@adsl-76-214-13-35.dsl.lsan03.sbcglobal.net) |
| 13:06.33 | [TK]D-Fender | adnc: You haven't told us what you're even using... |
| 13:06.41 | [TK]D-Fender | adnc: And what is an 'internal call"? |
| 13:07.07 | *** join/#asterisk SeriousMatters (~Sirius@87.115.170.210) |
| 13:07.23 | SeriousMatters | Hi, I think I have some NAT issue |
| 13:07.26 | adnc | internal call is via sip when calling an extension that is in the same network and not via any provider, i use twinkle on one side and siphone an iphone client on the other side |
| 13:08.53 | *** join/#asterisk oldbug (~oldbug@apokryph.org) |
| 13:09.00 | oldbug | hi there! |
| 13:09.41 | adnc | [TK]D-Fender, do you need some more information? |
| 13:10.06 | oldbug | i'm having problems to get codec_speex.so working in my installation (asterisk 1.6.2.2, and even with older versions) |
| 13:10.59 | [TK]D-Fender | adnc: Crappy mics & speakers <- |
| 13:11.10 | *** join/#asterisk coppice (~chatzilla@94.201.17.210.dyn.pacific.net.hk) |
| 13:12.09 | adnc | [TK]D-Fender, thank you for the tipp, i used a different client and it sounds great. |
| 13:12.36 | adnc | can someone recommend a wireless sip client like dect phones which can be hooked up via network? |
| 13:12.56 | oldbug | i get three errors with "speex", -> http://dpaste.com/160437/ |
| 13:13.01 | oldbug | any ides? :) |
| 13:13.10 | oldbug | *ideas* |
| 13:14.41 | dezent | adnc: i like the C450ip from siemens |
| 13:15.06 | dezent | adnc: its a dect phone and a sip client in the base station |
| 13:15.51 | radcliff | Hi, I am going nuts over Asterisk Realtime dynamic SIP-peers (In this case my SIP-provider). They keep disappearing from Realtime Cache and the only way to get them back is to make a call out trough them and they are in the cache for a little while more. "sip show peer SIP-PROVIDER load" does not seem to do the trick either, is it supposed to work like this? What is the meaning of realtime dynamic SIP peers if incoming calls does not work |
| 13:15.51 | radcliff | outgoing calls are not made at regular intervalls? Is it a bug or a feature? :) |
| 13:16.44 | *** join/#asterisk came0 (~came0@167.83.189.72.cfl.res.rr.com) |
| 13:16.51 | *** join/#asterisk Keeper82 (~Keeper@pat-mi.eni.it) |
| 13:17.07 | tzafrir_laptop | oldbug, is it asterisk from a package? self built? what distribution? |
| 13:17.30 | Keeper82 | hi, is there a way to wait for an answer when calling from *? |
| 13:17.50 | *** join/#asterisk Dibri (~gavit@pop1.isgroup.sr) |
| 13:17.52 | oldbug | it's self build, it was this AsteriskNOW, but i have downloaded the source-tarball, to see, if there are any changes... |
| 13:18.01 | tzafrir_laptop | oldbug, this symbol is from the library libspeexdsp, I believe |
| 13:18.46 | tzafrir_laptop | Maybe you have speex install but not speexdsp? (not sure if they are in the same package) |
| 13:19.01 | tzafrir_laptop | Or you have speex 1.0 installed and not speex 1.2 |
| 13:19.05 | oldbug | i have this problem with both versions, with AsteriskNOW ist was v1.4.29 |
| 13:19.22 | oldbug | hm, checking something... |
| 13:20.07 | SeriousMatters | radcliff: do you mean you cannot receive calls after a while? |
| 13:20.45 | radcliff | SeriousMatters: this is the output in the console when trying to call in to the asterisk: |
| 13:21.12 | radcliff | [Feb 17 14:07:19] WARNING[27699]: res_config_pgsql.c:479 realtime_multi_pgsql: PostgreSQL RealTime: Could not find any rows in table sipusers. |
| 13:21.13 | radcliff | [Feb 17 14:07:19] NOTICE[27699]: chan_sip.c:18096 handle_request_invite: Failed to authenticate device 070XXXXXXX <sip:070XXXXXXX@212.XXX.XXX.X;user=phone>;tag=897707995 |
| 13:21.14 | SeriousMatters | radcliff, don't paste output here, use something like pastebin.ca |
| 13:21.25 | radcliff | sorry |
| 13:21.28 | oldbug | tzafrir_laptop: do you know if i have to do *anything* more than 'make install' for speex? |
| 13:22.14 | tzafrir_laptop | oldbug, you should have it as standard packages, IIRC |
| 13:22.22 | radcliff | SeriousMatters: http://pastebin.ca/1799944 |
| 13:22.42 | tzafrir_laptop | but make; make install # should also do |
| 13:23.08 | radcliff | SeriousMatters: If I make an outgoing call trough that trunk it works again, if I don't it works maybe 30% of the time, really strange behaviour |
| 13:23.22 | oldbug | ok, yum list speex says, that there is v 1.0.5.e15_1.1 installed |
| 13:23.35 | oldbug | but i have build speex from source! |
| 13:25.45 | oldbug | 'speexdec --version' says: |
| 13:25.48 | oldbug | speexdec (Speex decoder) version 1.1.12 (compiled Feb 17 2010) |
| 13:29.00 | SeriousMatters | radcliff: sorry, I have no idea either. |
| 13:29.33 | radcliff | SeriousMatters: seems like google doesn't either... :( |
| 13:30.17 | *** join/#asterisk e4 (~e4@rrcs-76-79-59-194.west.biz.rr.com) |
| 13:31.40 | [TK]D-Fender | [08:17]<Keeper82>hi, is there a way to wait for an answer when calling from *? <- huh? |
| 13:31.51 | Keeper82 | asterisk :P |
| 13:32.13 | [TK]D-Fender | Keeper82: yes, I got the "*". Its your overall question that doesn't make sense |
| 13:32.31 | Keeper82 | ah ok I'll try to explain it better |
| 13:33.02 | Keeper82 | I'm building an emergency system which calls several numbers and playbacks a message |
| 13:33.46 | Keeper82 | the problem is that I don't know when someone answered the line |
| 13:34.04 | Keeper82 | so the message starts before someone answers |
| 13:34.50 | Keeper82 | is there a way to wait for an answer and then play the message? |
| 13:35.55 | ManxPower-work | Keeper82: you only don't know when the line is answered when using FXO ports. All other types do not have this issue. |
| 13:36.56 | ManxPower-work | so either stay away from using FXO ports or you'll have to have your dialplan loop thru, playing the message over and over. |
| 13:38.04 | oldbug | tzafrir_laptop: i will try to remove this "yum-package" and rebuild speex and asterisk... |
| 13:38.07 | Keeper82 | that's exactly what I'm doing now |
| 13:38.16 | ManxPower-work | Keeper82: then you are doing it right. |
| 13:38.48 | Keeper82 | thanks |
| 13:39.51 | Keeper82 | which other ports should I look for? |
| 13:40.05 | Keeper82 | we have 3 pstn lines atm |
| 13:40.06 | ManxPower-work | SIP or PRI |
| 13:40.53 | Keeper82 | I need to change the hw card, right? |
| 13:41.08 | ManxPower-work | Keeper82: and the lines. |
| 13:42.01 | Keeper82 | ok so I'm stuck using FX0, damn (changing lines is too much here) |
| 13:42.12 | Keeper82 | thanks again |
| 13:42.21 | *** join/#asterisk Alfio (~amunoz@75.112.88.200.m.sta.codetel.net.do) |
| 13:44.04 | ManxPower-work | Most Standard analog lines in don't provide answer indications |
| 13:44.36 | tzafrir_laptop | Keeper82, any chance you get polarity reversal on answer? |
| 13:44.55 | *** join/#asterisk rossand (~aross@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 13:45.04 | Keeper82 | I don't know |
| 13:45.11 | ManxPower-work | Keeper82: what country are you in? |
| 13:45.13 | Keeper82 | italy |
| 13:46.05 | tzafrir_laptop | oldbug, speex 1.1.2???? |
| 13:46.19 | tzafrir_laptop | oldbug, please get speex 1.2 |
| 13:46.43 | oldbug | uhm... |
| 13:46.45 | oldbug | sec |
| 13:46.55 | tzafrir_laptop | err... speex 1.1.12 != 1.1.2 . But I would still recommend using a newer version |
| 13:46.59 | *** join/#asterisk binbash_ (~peter@ip4da53781.direct-adsl.nl) |
| 13:47.32 | *** join/#asterisk p4p4 (~P4p4@248.121.113.82.net.de.o2.com) |
| 13:48.56 | plundra | Is there a dialplan function for looking up realname from voicemail-account? |
| 13:48.58 | oldbug | 1.2rc1? |
| 13:49.56 | *** join/#asterisk jaytee (~jforde@unaffiliated/jaytee) |
| 13:50.47 | ManxPower-work | plundra: I don't think so |
| 13:51.37 | plundra | ManxPower-work: Ok |
| 13:52.43 | Keeper82 | ManxPower-work, is there a way to test if my lines got polarity reversal on answer? |
| 13:53.01 | ManxPower-work | Keeper82: No. |
| 13:53.20 | ManxPower-work | We don't use that feature in the USA |
| 13:53.57 | Keeper82 | I mean some settings to change inside * and then make a test call |
| 13:54.06 | *** join/#asterisk Lord-Rahl (~quassel@173-162-45-177-michigan.hfc.comcastbusiness.net) |
| 13:55.29 | Lord-Rahl | ? in the queue_log witch column is the date? if this is a line in the log 1266413488|1266412902.362|250|SIP/107|COMPLETEAGENT|10|555|1 |
| 13:58.55 | tzafrir_laptop | Keeper82, you should get an event regarding polarity reversal from DAHDI |
| 13:59.02 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 13:59.36 | tzafrir_laptop | Asterisk may or may not do something about it, but if you enable debug (2 should be enough? not sure) you'll get a debug message about that event |
| 14:00.05 | tzafrir_laptop | oldbug, yes |
| 14:00.48 | *** join/#asterisk Pan3D (~Pan3D@streaming.sensoryresearch.net) |
| 14:00.58 | Keeper82 | I don't see anything about polarity reversal |
| 14:01.08 | Keeper82 | so it's not enabled |
| 14:01.27 | c0rnoTa | Lord-Rahl: the firs one |
| 14:01.37 | *** join/#asterisk niekie (~niek@CAcert/Assurer/niekie) |
| 14:01.56 | c0rnoTa | ~s/first/firs |
| 14:02.14 | *** join/#asterisk came0 (~came0@167.83.189.72.cfl.res.rr.com) |
| 14:02.17 | c0rnoTa | ~s/firs/first |
| 14:02.32 | c0rnoTa | didn't know how to use this function ^) |
| 14:02.41 | ManxPower-work | no ~ |
| 14:02.47 | c0rnoTa | :) thx |
| 14:03.04 | ManxPower-work | Mary had a little hedgehog. |
| 14:03.13 | ManxPower-work | s/ hedgehog/lamb |
| 14:03.25 | ManxPower-work | s/ hedgehog/lamb/ |
| 14:03.25 | [TK]D-Fender | FAIL |
| 14:03.38 | ManxPower-work | *sigh* |
| 14:03.43 | [TK]D-Fender | LAMB LAMB! |
| 14:04.33 | jaytee | WHAM-BA-LAMB OLE BLACK BETTY |
| 14:05.07 | Lord-Rahl | c0rnoTa, thanks can you fathom a guess how it broken up. I need to write a report against it |
| 14:05.42 | ManxPower-work | Lord-Rahl: it should be documented in the Asterisk tarball in the /doc directory |
| 14:06.27 | *** join/#asterisk Katty (~asteriska@mail.copi-rite.com) |
| 14:06.34 | Lord-Rahl | ManxPower-work: thnaks i will check it out |
| 14:06.52 | Katty | hi. |
| 14:06.56 | jaytee | hi |
| 14:07.30 | *** join/#asterisk dpisites (~cheng@CPE002191f85581-CM001692568382.cpe.net.cable.rogers.com) |
| 14:07.36 | *** part/#asterisk dpisites (~cheng@CPE002191f85581-CM001692568382.cpe.net.cable.rogers.com) |
| 14:08.22 | *** join/#asterisk nighty^ (~nighty@x122091.ppp.asahi-net.or.jp) |
| 14:08.41 | SeriousMatters | Can I ask a question about openSER here? nothing in details. |
| 14:09.46 | c0rnoTa | SeriousMatters: aren't you afraid that it will decrease your karma? |
| 14:10.23 | SeriousMatters | hm... |
| 14:10.30 | [TK]D-Fender | My karma ran over your dogma... |
| 14:10.53 | c0rnoTa | SeriousMatters: it should be better, if you will ask your question on #openser |
| 14:11.07 | c0rnoTa | I think so |
| 14:11.15 | adnc | dezent, thank you, it seems to be quite expensive, but looks like a good phone C450ip, do you know if there are alternatives? |
| 14:11.58 | dezent | adnc: not that i know of, maybe you can by a used from ebay? |
| 14:12.11 | adnc | good idea |
| 14:12.22 | ManxPower-work | C450IP?? |
| 14:12.36 | dezent | works like a champ |
| 14:12.51 | ManxPower-work | You won't get much help for that phone here. |
| 14:12.57 | dezent | adnc: http://www.voip-info.org/wiki/view/DECT |
| 14:13.06 | ManxPower-work | ~phones |
| 14:13.07 | infobot | While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Linksys SPA-9XX, Snom, Cisco 79XX, everything else. Do not consider Grandstream phones. Ever. |
| 14:13.10 | dezent | its pretty straight forward |
| 14:13.10 | adnc | dezent, thanks |
| 14:13.43 | adnc | ManxPower-work, i'm looking for dect or even wireless phones |
| 14:13.44 | [TK]D-Fender | Seimens makes pretty decent DECT stuff |
| 14:13.48 | dezent | adnc: and go for dect phones, voip phones tend to suck |
| 14:14.13 | adnc | well voip over dect would be what i want, or voip over wireless |
| 14:14.17 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 14:14.21 | [TK]D-Fender | dezent: Not "VoIP".... WiFi <- |
| 14:14.28 | ManxPower-work | dect isn't wireless? |
| 14:14.38 | dezent | [TK]D-Fender: yes, Wifi |
| 14:14.41 | adnc | ManxPower-work, yes it is, but i was about wifi |
| 14:14.56 | ManxPower-work | then say it. Being inaccurate in telecom can cost you money. |
| 14:15.06 | adnc | ManxPower-work, how much ;) |
| 14:16.32 | adnc | i've a fritzbox with a mt-d mobile dect device, can this device be used without the fritzbox itself? |
| 14:16.47 | adnc | is there a dect basestation that could connect with the mt-d? |
| 14:17.02 | Poincare | is there an 'official looking' certificate to explain our local RIAA-thing that the MOH provided with asterisk is license free when used with Asterisk |
| 14:17.16 | *** join/#asterisk Ryushin (proxy@windwalker.openinnovations.com) |
| 14:17.17 | dezent | adnc: np |
| 14:17.22 | Katty | hummmm |
| 14:18.42 | *** part/#asterisk icyValk77 (~icyValk77@gateway.ash.thebunker.net) |
| 14:18.46 | ManxPower-work | AAARRRGGGHHH!!!!!!! |
| 14:19.01 | adnc | ManxPower-work, whats wrong? |
| 14:19.07 | ManxPower-work | Why does my boss have to dispatch the most usless of our techs to work with me? |
| 14:19.23 | Faustov | did anyone manage to compile wanpipe (I guess with the right patches) after the net_device_ops kernel ABI changes? |
| 14:19.43 | *** join/#asterisk visz (vis@irkki.fi) |
| 14:19.46 | ManxPower-work | Me: "OK, shutdown the PBX". Tech: yanks out the power cord to the PBX. |
| 14:19.47 | visz | ehlo |
| 14:20.08 | dezent | ManxPower-work: kill him, make it look like an accident |
| 14:20.12 | adnc | visz, is this a typo or are you talking smtp? |
| 14:20.30 | ManxPower-work | dezent: Don't tempt me. He's 1,000 miles away |
| 14:20.37 | visz | it was syntactically incorrect extended hello ,) |
| 14:20.40 | dezent | ManxPower-work: hehe |
| 14:20.48 | adnc | ManxPower-work, remote kill? |
| 14:20.49 | dezent | ManxPower-work: airplane tckets are cheap |
| 14:21.18 | dezent | ManxPower-work: http://www.dailyhaha.com/_pics/standard_TCPIP_punch.jpg |
| 14:22.18 | adnc | dezent, ;) |
| 14:22.22 | oldbug | compiling... compiling... com... *sigh* |
| 14:24.58 | adnc | dezent, the c470ip costs about 80 euros in germany |
| 14:25.19 | adnc | dezent, do you have one? |
| 14:25.40 | adnc | can you tell me if the display behaviour if this device is typically siemens very slow? |
| 14:26.25 | dezent | adnc: no i have the older one, C450ip |
| 14:26.36 | Mark22 | Hello, did someone find an option to use Polycom Soundpoint phones to work with asterisk using the web interface? |
| 14:26.38 | dezent | adnc: it is pretty slow on the C450ip |
| 14:26.49 | adnc | dezent, typically siemens |
| 14:27.07 | dezent | adnc: but its a very good phone |
| 14:27.18 | visz | gnaah.. if i get goto destination from asterisk db, it drops pipe-characters |
| 14:27.22 | adnc | but the fritzbox.fon mt-d is also slow, but here i need to say that it gets his addressbook from the fritz.box router-addressbook |
| 14:27.24 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 14:27.32 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 14:27.39 | visz | as in 'ext-queues|1234|1' -> 'ext-queues' |
| 14:28.04 | *** join/#asterisk minotaur01 (~minotaur0@24.215.3.50) |
| 14:29.11 | *** join/#asterisk AJLove (~adde@c193-150-246-164.bredband.comhem.se) |
| 14:29.13 | *** join/#asterisk Godfather_ (~Godfather@79.109.251.13.dyn.user.ono.com) |
| 14:29.13 | adnc | if i also want to see which encoding for rtp is used what verbosity level would i need? |
| 14:29.33 | Godfather_ | o/ |
| 14:29.55 | Katty | hi Godfather_ |
| 14:29.59 | ManxPower-work | Mark22: Um, polycoms work just fine with Asterisk. |
| 14:30.21 | Katty | is on the phone currently, and using a polycom. |
| 14:30.29 | AJLove | At an incoming call nothing happens more than i see this in the COCOL: "Using SIP RTP CoS mark 5" - Could anyone send me in the right direction? |
| 14:30.35 | visz | polycom web interface is rather clumsy |
| 14:30.40 | AJLove | Conencted to a SIP gateway |
| 14:30.48 | Katty | visz: perhaps you'd like to write a better one? |
| 14:30.48 | visz | takes patience to configure a single phone |
| 14:30.57 | Katty | why are you configuring one phone at a time? |
| 14:31.00 | Katty | that doesn't make much sense. |
| 14:31.02 | visz | indeed |
| 14:31.13 | Katty | if i were you, i'd configure them using ftp |
| 14:31.17 | visz | because i had to |
| 14:31.21 | ManxPower-work | Nobody configures a Polycom via the web interface anymore. |
| 14:31.23 | Katty | aww poor baby. |
| 14:31.26 | Katty | cause you had to |
| 14:31.36 | Katty | i don't believe that for a minute. |
| 14:31.37 | visz | and did not have time to construct a provisioning framework |
| 14:31.41 | ManxPower-work | Me neither. |
| 14:31.41 | Katty | ftp servers take 2 minutes to setup |
| 14:31.48 | Katty | maybe less. |
| 14:32.01 | ManxPower-work | No, he doesn't have the time to NOT set up a provisioning server. |
| 14:32.02 | spenguin[work] | :> |
| 14:32.13 | Katty | hi there smoooth operator |
| 14:32.27 | Katty | listens to the song, again |
| 14:32.32 | spenguin[work] | hey Katty, hows riddick and the critters |
| 14:32.34 | Katty | Qwell: you missed fun times on redditroulette last night |
| 14:32.39 | Katty | spenguin[work]: well riddick is fine |
| 14:32.42 | Katty | checks crittercam |
| 14:33.24 | Katty | looks like the squirrels have knocked the tray crooked |
| 14:33.30 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:33.30 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:33.35 | Katty | hi blitzrage |
| 14:33.44 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 14:33.44 | leifmadsen | zup yo? |
| 14:33.48 | Katty | yup zo |
| 14:34.04 | leifmadsen | thinking about getting a snack |
| 14:34.05 | *** join/#asterisk basty (~basty@2001:4cd8:1:0:21c:b3ff:fec2:ec18) |
| 14:34.06 | leifmadsen | and some water |
| 14:34.07 | basty | Hi |
| 14:34.13 | Katty | mmm, waters. |
| 14:34.16 | Katty | hi basty |
| 14:34.35 | AJLove | Is there a way to get more detailed info on whats happening in the consol..more than the : "Using SIP RTP CoS mark 5" Getting very frustrating not getting incoming calls :S |
| 14:34.50 | ManxPower-work | AJLove: What specific thing are you looking for? |
| 14:34.56 | leifmadsen | AJLove: sip set debug on |
| 14:35.15 | Godfather_ | hi Katty |
| 14:35.58 | Mark22 | ManxPower-work: it should work, but currently it doesn't work for me (hopefully re-configuring everything solves it) |
| 14:36.00 | AJLove | Thanks leif... |
| 14:36.33 | Mark22 | I also use the webinterface currently (as I would like to provide the configs using http in the future, but for now this is fine) |
| 14:36.36 | adnc | dezent, http://www.sipgate.de/voipshop/voip/dect-telefone |
| 14:36.47 | Mark22 | the http server is located on the internet |
| 14:37.17 | *** part/#asterisk RoyK (~RoyKa@nilugw1.nilu.no) |
| 14:37.38 | *** join/#asterisk came0 (~came0@rrcs-71-42-53-182.se.biz.rr.com) |
| 14:37.53 | AJLove | hmmm... the call is coming in to the asterisk, but it wont handle the call in any way.... |
| 14:38.19 | AJLove | In the phone calling the asterisk its just quiet... |
| 14:38.37 | spenguin[work] | AJLove: well check the console? |
| 14:38.40 | spenguin[work] | what do you see |
| 14:39.17 | AJLove | With debug on it shows the phonenumber dialing and so on... but nothing regarding the dialplan |
| 14:39.29 | basty | I have a strange problem. I updated from asterisk 1.2 to 1.4.29. If somebody (from external) calls an extension like 069123456 and the phone (snom 320) is using a call forward to an internal extension "300". The caller doesnt hear anything (no ringing / no musiconhold) untill somebody of "300" answeres the call. It seems to be a problem with the local channel (http://pastebin.com/d72b07d5a). Anyone could give me a hand ? |
| 14:39.49 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 14:40.46 | AJLove | not really an advanced dialplan i got either: http://pastebin.com/m69cb8fbc |
| 14:42.39 | *** join/#asterisk Geminizer (~me@cpe-76-180-27-4.buffalo.res.rr.com) |
| 14:43.09 | *** join/#asterisk Tobarja_ (~chatzilla@user-0c8h5rb.cable.mindspring.com) |
| 14:43.47 | Geminizer | hello all. is there a way to disable logging to /var/log/asterisk/full altogether ? |
| 14:43.47 | rue_house | AJLove, have you turned on debuggin yet? |
| 14:43.50 | tzafrir_laptop | Geminizer, edit /etc/asterisk/logger.conf and run 'logger reload' |
| 14:43.59 | AJLove | yes debugging on |
| 14:44.05 | rue_house | AJLove, core set verbose 99 |
| 14:44.09 | rue_house | AJLove, core set debug 99 |
| 14:45.29 | AJLove | with thos two commands it showed nothing more... but i got lot of info when sip set debug was turned on |
| 14:45.33 | Geminizer | Thanks, tzafrir_laptop |
| 14:46.17 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 14:46.47 | *** join/#asterisk Madkiss (~madkiss@freenode/staff-emeritus/madkiss) |
| 14:46.48 | Madkiss | hi folks |
| 14:47.08 | Madkiss | Can somebody please point me to a howto on how to set up asterisk as a sip *provider*? |
| 14:47.28 | rue_house | AJLove, you just trying to dial from one to the other? |
| 14:47.40 | AJLove | that works |
| 14:47.53 | AJLove | from one sip client to another using the extensions |
| 14:48.16 | rue_house | so what isn't working? |
| 14:48.46 | tzafrir_laptop | Madkiss, what do you mean by a "sip provider"? |
| 14:49.02 | tzafrir_laptop | (technically) |
| 14:49.11 | Madkiss | tzafrir_laptop: I want my asterisk to act like sipgate does. i.e. others can set up their asterisk to use mine as sip trunk. |
| 14:49.43 | AJLove | rue: Ive posted the extension and the consol output: http://pastebin.com/d65a00ac9 |
| 14:50.09 | rue_house | ok I got lost on whats not working |
| 14:50.14 | AJLove | Basically nothing happens butit should send the incoming call to extension 101 |
| 14:50.39 | rue_house | [cellip-in] ? |
| 14:51.25 | rue_house | or are you trying to dial 1000 from a phone? |
| 14:52.03 | AJLove | 1000 is the extension that is assocuiated with the registry string |
| 14:52.46 | AJLove | so when i dial my Virtual Phonenumber my sip gatway provider sends it to my Asterisk which in its turn should send the call to a phone |
| 14:52.56 | AJLove | in this testing case a SIP client |
| 14:52.56 | rue_house | you need to tell me whats not happening when you expect it to |
| 14:53.00 | AJLove | extension 101 |
| 14:53.04 | rue_house | ah |
| 14:53.08 | rue_house | ok, so |
| 14:53.21 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 14:53.42 | rue_house | your provider is a sip account correct? |
| 14:53.49 | AJLove | yes |
| 14:54.04 | rue_house | in sip.conf, what is the "context" line of the config for them? |
| 14:55.18 | [TK]D-Fender | AJLove: SIP/2.0 401 Unauthorized <----------------- |
| 14:55.19 | AJLove | register => PHONENUMBER:pwd@sip.mygateway.com/1000 |
| 14:55.23 | *** join/#asterisk TimeRider (~steve@78.32.26.1) |
| 14:55.32 | [TK]D-Fender | AJLove: Your call is rejected outright. Dialplan is irrelevant |
| 14:57.30 | AJLove | Fender: What scenarios would cause this? |
| 14:58.07 | [TK]D-Fender | AJLove: Wrong user/pass. Carrier sending un-authed calls to you and you're not telling * to allow it. |
| 14:58.09 | rue_house | I didn't catch that :) |
| 14:58.28 | [TK]D-Fender | rue_house: WARNING! chan_myopia.so is already loaded! |
| 14:58.54 | Naikrovek | hehe |
| 14:59.15 | rue_house | I was gonna work on the other issues |
| 15:00.02 | AJLove | sip.conf / disallow=all <--- Problem? |
| 15:00.18 | rue_house | (make sure context was right and have him use s instead of 1000) |
| 15:00.34 | rue_house | depends on the allows after it |
| 15:00.35 | *** join/#asterisk nighty^ (~nighty@x122091.ppp.asahi-net.or.jp) |
| 15:00.46 | AJLove | ulaw and alaw |
| 15:00.51 | [TK]D-Fender | AJLove: That is for CODEC negotiation. Not involved. |
| 15:00.58 | *** part/#asterisk benngard (~benngard@213.88.138.230) |
| 15:00.59 | rue_house | right, thats just audio codecs |
| 15:01.28 | rue_house | you have a account connection problem, between them and you |
| 15:01.46 | [TK]D-Fender | AJLove: To your inbound peer (should be the [mysecretary], RIGHT?). To this entry add "insecure=port,invite" |
| 15:02.50 | AJLove | it was insecure=very |
| 15:03.23 | AJLove | that was the problem.... thanks... |
| 15:03.45 | AJLove | What exactly did we do here? :) Must now else ill never learn ;) |
| 15:03.46 | rue_house | it gets thru to 101? |
| 15:03.49 | AJLove | *know |
| 15:03.55 | AJLove | yepp |
| 15:04.21 | rue_house | hmm |
| 15:05.24 | *** join/#asterisk Ryushin (proxy@windwalker.openinnovations.com) |
| 15:05.35 | AJLove | insecure=very (was actually the recomendations from my provider) Could it been for older version of asterisk? |
| 15:06.44 | [TK]D-Fender | AJLove: Yes |
| 15:07.27 | AJLove | Whiel im here... to exit from Consol ive read STOP NOW - But that command doesnt work for me... ive been using !killall asterisk, what is the correct way? |
| 15:07.42 | AJLove | and fender and rue: thanks alot |
| 15:08.02 | ManxPower-work | You need to read the Asterisk book. You are making total n00b mistakes. |
| 15:08.47 | ManxPower-work | STOP NOW is only valid in the CONSOLE, not the REMOTE Asterisk CLI |
| 15:09.13 | mmlj4 | hrm? |
| 15:09.23 | AJLove | ok, i dont have access to the console |
| 15:09.29 | ManxPower-work | AJLove: of course you do. |
| 15:09.36 | ManxPower-work | "asterisk -rvvv" |
| 15:09.47 | [TK]D-Fender | 10:08]<ManxPower-work>STOP NOW is only valid in the CONSOLE, not the REMOTE Asterisk CLI <- ummm.. pardon? |
| 15:10.11 | ManxPower-work | Sorry, I reversed them. |
| 15:10.11 | [TK]D-Fender | ManxPower-work: Or was this concerning an OS console? |
| 15:10.45 | [TK]D-Fender | AJLove: And "stop now" is an * CLI command, not an OS application |
| 15:10.46 | *** join/#asterisk hluesea (~hulusikah@88.247.127.66) |
| 15:12.07 | AJLove | Asterisk Ready. |
| 15:12.07 | AJLove | <PROTECTED> |
| 15:12.07 | AJLove | *CLI> stop now |
| 15:12.07 | AJLove | No such command 'stop now' (type 'core show help stop now' for other possible commands) |
| 15:12.07 | AJLove | *CLI> |
| 15:13.00 | [TK]D-Fender | .... WTF |
| 15:13.39 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:13.39 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:13.49 | [TK]D-Fender | AJLove: PASTEBIN the output of 'help" from * cLI |
| 15:13.51 | [TK]D-Fender | ~pb |
| 15:13.52 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding 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/ , or apt-get install pastebinit |
| 15:13.53 | [TK]D-Fender | ^^^^^^^ |
| 15:14.10 | hluesea | hello, i am trying to use dahdi and i take like that pri debug result. How can i fix the problem ? or Where i should look around for solution http://pastebin.com/d58ff1a72 |
| 15:14.34 | AJLove | http://pastebin.com/m7484e337 |
| 15:16.10 | [TK]D-Fender | AJLove: core stop gracefully Gracefully shut down Asterisk |
| 15:16.12 | [TK]D-Fender | core stop now Shut down Asterisk immediately |
| 15:16.13 | [TK]D-Fender | core stop when convenient Shut down Asterisk at empty call volume |
| 15:16.26 | [TK]D-Fender | AJLove: so now prefixed with "core" |
| 15:16.53 | AJLove | okay... thanks... is this a recent change? |
| 15:17.35 | [TK]D-Fender | AJLove: Apparently |
| 15:18.01 | [TK]D-Fender | hluesea: Ext: 1 Cause: Incompatible destination (88), class = Invalid message (e.g. parameter out of range) (5) ] |
| 15:18.16 | AJLove | Anyway guys... thanks for all your help... I came past one speedbump...Now its time to reading and testing :) |
| 15:18.19 | AJLove | Take care |
| 15:18.26 | [TK]D-Fender | hluesea: Presentation: Presentation permitted, user number not screened (0) '250' ] <--- pretty sure they don't like your CALlErID |
| 15:24.26 | oldbug | tzafrir_laptop: done rebuilding with speex v1.2rc1 |
| 15:24.30 | oldbug | but now... |
| 15:24.35 | oldbug | libspeex.so is missing :\ |
| 15:25.29 | tzafrir_laptop | libspeex.so ? where do you get that error? |
| 15:25.32 | hluesea | <[TK]D-Fender> thanks i am trying different types |
| 15:25.46 | oldbug | on asterisk -vvvvvvvvr |
| 15:25.59 | oldbug | while loading codec_speex.so |
| 15:30.41 | [TK]D-Fender | hluesea: And you should probably have "pridialplan=unknown" and "prilocaldialplan=unknown" |
| 15:32.11 | ManxPower-work | [TK]D-Fender: *gasp* like the sample config files says!?!?! |
| 15:32.26 | mmattice | is it just general practice that if you're having issues with a PRI to upgrade dahdi to the latest? |
| 15:32.37 | *** part/#asterisk z53102 (~nikolay@208.89.107.74) |
| 15:32.47 | ManxPower-work | mmattice: only if you ask Digium. |
| 15:33.04 | oldbug | you might have a |
| 15:33.05 | oldbug | look at this: http://dpaste.com/160507 |
| 15:33.08 | ManxPower-work | Digium things ALL PROBLEMS EVER have been fixed in DAHDI. |
| 15:33.32 | [TK]D-Fender | mmattice: it's general practice for people to come in here asking vague questions about even vaguer scenarios and expecting a specific response |
| 15:33.42 | ManxPower-work | Echo? Upgrade to DAHDI and get even more echo! Lost interruprs? Upgrade to DAHDI to get even more missed interrupt goodness! |
| 15:34.14 | beek | ManxPower-work: Hmmmm.... do I detect a note of sarcasm? |
| 15:34.27 | [TK]D-Fender | oldbug: You don't seem to have libspeex |
| 15:35.14 | *** join/#asterisk Deeewayne (~dwayne@75.76.254.162) |
| 15:35.15 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:35.22 | *** join/#asterisk Abdou`` (~abdou@41.227.228.124) |
| 15:35.37 | mmattice | [TK]D-Fender: my PRI will take one call after a reset, then the second call won't enter state 6 and instead goes to 11 |
| 15:36.48 | [TK]D-Fender | mmGot debug? How about versions for * & DAHDI? how about what CARD? You know... DETAILS? |
| 15:36.53 | Abdou`` | hello, plz anyone talk french ? salut quelq'un parle français ? |
| 15:37.26 | *** join/#asterisk Polysics (~luca@host113-41-static.25-87-b.business.telecomitalia.it) |
| 15:37.28 | Polysics | hello |
| 15:37.32 | mmlj4 | non |
| 15:37.34 | [TK]D-Fender | Abdou``: Non, deuxieme porte a la gauche s-t-p |
| 15:37.41 | Polysics | what is the general consensus on Adhearsion? |
| 15:37.51 | Abdou`` | :) |
| 15:38.05 | Polysics | i need to do a variety of actions through AGI/AMI, Adhearsion looks cool but I can't decide |
| 15:38.12 | Polysics | also, the front end app is in PHP, not ruby |
| 15:38.21 | oldbug | [TK]D-Fender: indeed ;) but why? |
| 15:38.45 | [TK]D-Fender | oldbug: Maybe because you didn't install it? |
| 15:39.02 | oldbug | libspeex should be part of speed, right? |
| 15:39.09 | oldbug | *speex* :) |
| 15:39.10 | [TK]D-Fender | Polysics: could your needs be a little more vague please... |
| 15:39.15 | [TK]D-Fender | oldbug: No. |
| 15:39.22 | oldbug | oh... |
| 15:39.25 | Abdou`` | boff puisque ya pas frenchies, can somone help me installing instant messages with asterisk ? |
| 15:39.30 | [TK]D-Fender | oldbug: libspeex = libspeex |
| 15:39.36 | oldbug | ok |
| 15:39.38 | mmattice | dahdi v2.2.0.1 * ver 1.6.1.0, card is a digium with pci id d161:0220 and running the wct4xxp module. I don't remember the model number but it's the two port PCIE card with the echo cancelation daughterboard |
| 15:39.50 | Geminizer | Is there a way asterisk can say digits in a formal manner (e.g. 180 as "one hundred eighty" as opposed to "one eight zero") ? |
| 15:39.51 | [TK]D-Fender | Abdou``: 8 doesn't do "messaging" |
| 15:39.54 | [TK]D-Fender | * |
| 15:40.08 | [TK]D-Fender | Geminizer: core show application saynumber |
| 15:40.10 | beek | Geminizer: as in SayNumber() |
| 15:40.15 | Polysics | [TK]D-Fender, for example, setting a status in a DB when a SIP account status changes or registers |
| 15:40.26 | *** join/#asterisk highvoltz (rogers@bling.bling.org) |
| 15:40.28 | Abdou`` | ok [TK]D-Fender ok thx any solution or hack ? |
| 15:40.32 | beek | Morning [TK]D-Fender |
| 15:40.33 | Polysics | creating a queue on the fly and adding some SIP users to it |
| 15:40.49 | Polysics | logging queue results in a specific format |
| 15:40.52 | *** join/#asterisk rubberneck (~chatzilla@ext-52.sagetelecom.net) |
| 15:40.56 | oldbug | [TK]D-Fender: but... where can i get this? google points to speex.org :\ |
| 15:40.59 | [TK]D-Fender | Abdou``: No. * isn't a MESSAGING platform, it is a TELEPHONY platform. |
| 15:41.21 | [TK]D-Fender | oldbug: there are docs in your TARBALL. Go read them. |
| 15:41.22 | rubberneck | Is there a way to get info from SDP messages through the dialplan? |
| 15:41.45 | ManxPower-work | Polysics: It depends on who you ask. Most scientists think it is a basic principle of physics, but others disagree. |
| 15:41.48 | [TK]D-Fender | rubberneck: "core show function SIP_HEADER" |
| 15:42.25 | Polysics | ManxPower-work, what is a simple principle of physics? :-) |
| 15:42.40 | ManxPower-work | Polysics: Adhearsion |
| 15:42.48 | rubberneck | [TK]D-Fender: That might do the trick. thanks. |
| 15:43.02 | Polysics | ManxPower-work, i appreciate the humor, but you know what i mean :-) |
| 15:43.30 | ManxPower-work | The humor was you thinking (most) everyone here knew what you were talking about. |
| 15:43.45 | Polysics | application flow in as few words as possible: users connect and get registered through a webpage that hosts a Zoiper webphone controlled though JS |
| 15:43.56 | Polysics | users can see who is online and what their status is |
| 15:44.10 | Abdou`` | [TK]D-Fender elle estou la deuxieme porte a gauche ? et merci en tout cas |
| 15:44.16 | *** part/#asterisk Abdou`` (~abdou@41.227.228.124) |
| 15:44.27 | ManxPower-work | Polysics: *nod* We do something similar. |
| 15:44.38 | Polysics | initiate calls, eventually being queued if an user is busy. at the end of the call, the call gets logged for billing purposes |
| 15:45.12 | Polysics | they can also join a queue for a specific language (it's a mutual translation service), and wait till someone that speaks that is free |
| 15:45.18 | Polysics | that is more or less all |
| 15:45.40 | Polysics | ManxPower-work, may I ask with what technology/ies you did that? |
| 15:45.55 | ManxPower-work | Polysics: Polycom Sidecars mostly. |
| 15:46.33 | Polysics | oh, the same queue can be joined by calling a PSTN number with an IVR |
| 15:46.46 | Polysics | will asking what a Sidecar is make me look bad? :-) |
| 15:47.07 | ManxPower-work | no, we don't do any of that stuff. We monitor extension status using the Sidecars. That was all you had said when I said "we do something similar" |
| 15:47.17 | Polysics | ok, i'd say it is a sort of supplementary display |
| 15:47.22 | ManxPower-work | yes |
| 15:47.28 | ManxPower-work | polycom's web site has it listed |
| 15:47.33 | Polysics | do you monitor them through AMI? |
| 15:47.55 | ManxPower-work | no. We use BLF. |
| 15:48.24 | Polysics | reads up on BLF |
| 15:48.34 | ManxPower-work | Polysics: just read the Asterisk Book |
| 15:48.49 | Polysics | i probably skipped a few pages :-) |
| 15:49.28 | Polysics | no, anyway, BLF looks like it is not for us as we are not using 1 ext for each sip user |
| 15:50.23 | Polysics | my first question was more general, anyway - anyone has an opinion on adhearsion? am i better off just making a bunch of PHP scripts? |
| 15:50.42 | Polysics | then we can look at solutions, i would love that :-) |
| 15:50.53 | *** join/#asterisk albertoandrade (~albertoan@201.47.12.223.dynamic.adsl.gvt.net.br) |
| 15:51.01 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
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| 15:57.19 | *** part/#asterisk c0rnoTa (~c0rnoTa@94.25.175.235) |
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| 15:59.05 | *** part/#asterisk Tobarja (~chatzilla@user-0c8h5rb.cable.mindspring.com) |
| 15:59.15 | ManxPower-work | What. Is. Adhersion? |
| 15:59.52 | beek | http://adhearsion.com |
| 16:00.04 | Polysics | ManxPower-work, read the Asterisk book :-P |
| 16:00.16 | Polysics | j/k, but it is mentioned in there |
| 16:00.41 | *** join/#asterisk UQlev (~yuriy@nb11-125.static.cytanet.com.cy) |
| 16:00.52 | Polysics | it is a Ruby framework that sits on top of AGI/AMI, ideally making work on that kind of app saner and more "standard", while also using a decent programming language |
| 16:01.53 | leifmadsen | depends on your definition of decent :) |
| 16:02.33 | [TK]D-Fender | and "standard" |
| 16:02.45 | Polysics | i am not madly in love with Ruby, but it beats extensions.conf syntax :-) |
| 16:03.05 | leifmadsen | depends on who you are I suppose |
| 16:03.11 | leifmadsen | I love the dialplan |
| 16:03.20 | leifmadsen | I can make it bend to my will |
| 16:03.31 | ManxPower-work | I'll use AEL, thankyouverymuch |
| 16:04.23 | Polysics | AEL looks suspiciously like an Adhearsion app, or vice-versa :-) |
| 16:05.02 | Polysics | ok, i will go with one question at a time: how can I react to an user registering/calling/answering/hanging up, and logging his current status somewhere so it can be displayed on a web page? |
| 16:05.13 | Polysics | there are many ways, what is one? :-9 |
| 16:06.06 | Polysics | talking about SIP accounts in the above |
| 16:06.14 | *** join/#asterisk benngard (~benngard@90-230-92-67-no148.tbcn.telia.com) |
| 16:06.52 | [TK]D-Fender | Polysics: Make an AMI daemon app |
| 16:07.28 | Polysics | [TK]D-Fender, which bring me to having to use one of the languages I know best, which is either PHP or Ruby |
| 16:07.42 | [TK]D-Fender | Polysics: Whichever you feel like |
| 16:07.46 | Polysics | and PHP looks very ill-suited, though I haven't really tested |
| 16:08.37 | Polysics | part 2 of the same reasoning is: since i will probably be using Ruby, i stumbled on Adhearsion and was wondering if it was a good thing |
| 16:08.38 | ManxPower-work | I write most of my Asterisk AGIs i PHP -- we already have a php library of our internal custom functions for our GUI, might as well reuse as much of the code as we can for AGIs |
| 16:09.00 | Polysics | ManxPower-work, AGI is one thing, an AMI daemon is another |
| 16:09.16 | Polysics | eg. i was thinking of using threads/fibers to respond to events in non-blocking manner |
| 16:09.24 | ManxPower-work | Polysics: We use AMI extensivly in our library. |
| 16:09.42 | Polysics | ManxPower-work, by issuing commands or by responding to events? |
| 16:09.52 | ManxPower-work | Just like CGI, AMI and AGI are language agnostic |
| 16:10.05 | ManxPower-work | Polysics: mainly issuing commands and getting the response. |
| 16:10.06 | *** join/#asterisk CunningPike (~CunningPi@S01060014bf81366b.vc.shawcable.net) |
| 16:10.18 | Yudaisrael1984 | does ai know this isnt the ruby channel alhtough im guessing someone here knows the answer, are ruby files encrypted or can i just use a regular text editor to view them?? also where would cgi scripts be on asterisk |
| 16:10.23 | Polysics | commands are easier than events, which is what is giving me issues :-) |
| 16:11.04 | Polysics | Yudaisrael1984, they should be plain text, but it is not a question for here |
| 16:11.17 | *** join/#asterisk theHub (~theHub@69.177.93.21) |
| 16:11.36 | Yudaisrael1984 | i know and where are the cgi files for asterisk that would contain the ruby files |
| 16:11.53 | Yudaisrael1984 | which folder would i expect to find it? |
| 16:12.06 | ManxPower-work | Yudaisrael1984: Asterisk does not come with Ruby files. |
| 16:12.10 | Polysics | was just warning you because a few mods here are quick with the axe :-P |
| 16:12.47 | Yudaisrael1984 | i understnad that but if someone would write a script in ruby or in PHP where would it be stored that asterisk can access it easily |
| 16:13.28 | *** join/#asterisk xmitter (~xmitter@c-24-21-213-242.hsd1.or.comcast.net) |
| 16:13.31 | ManxPower-work | Yudaisrael1984: /var/lib/asterisk/agi-bin |
| 16:13.36 | Yudaisrael1984 | thanks |
| 16:14.22 | [TK]D-Fender | Yudaisrael1984: Anywhere * can read from |
| 16:14.45 | Katty | blahhh |
| 16:14.48 | Katty | why do taxes take forever to file |
| 16:15.04 | Naikrovek | they don't. i had my rebate like 3 weeks ago |
| 16:15.13 | Naikrovek | maybe 2 weeks ago |
| 16:15.39 | p3nguin_ | Filling out the return can sometimes take a long time, depending on how much crap you have to calculate. |
| 16:15.53 | Naikrovek | true |
| 16:15.55 | Katty | Naikrovek: they do when you have a cpa |
| 16:15.59 | p3nguin_ | But any refund you get should pay for your time of filing. |
| 16:16.00 | Naikrovek | ah yeah |
| 16:16.08 | Katty | beats head on desk |
| 16:16.11 | Naikrovek | they have a lot of other people they're helping at the same time as you |
| 16:16.17 | Katty | well yeah |
| 16:16.22 | Katty | but the office visit takes like 2 hours |
| 16:16.23 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 16:16.28 | Katty | because she wants to Talk to me |
| 16:16.31 | Katty | and ChitChat |
| 16:16.32 | p3nguin_ | Should have used TurboTax! |
| 16:16.37 | Katty | JUST PUT IN THE BLOODY NUMBERS |
| 16:18.02 | Katty | yeah i think i will next time |
| 16:18.07 | Katty | or at least use H&R block's website |
| 16:19.18 | p3nguin_ | They probably charge for that, though. |
| 16:19.26 | Katty | i don't care |
| 16:19.31 | Katty | they can't charge more than my cpa |
| 16:19.36 | oldbug | uhm, so, back from reading *again*... :) |
| 16:19.50 | p3nguin_ | If you don't have any fancy stuff to deal with, the online TurboTax is free. |
| 16:20.08 | Katty | yeah i have some ....extra stuff |
| 16:20.45 | oldbug | as i understand this document, anything i have to do is ./configure, make and make install in /usr/src/speex, and this should install libspeex and libspeexdsp, too |
| 16:20.58 | oldbug | what am i missing? |
| 16:21.14 | Naikrovek | oldbug: traditionally, i just ./configure;make;make install |
| 16:21.18 | Katty | you're never fully dressed |
| 16:21.19 | Katty | without a smile |
| 16:21.54 | oldbug | Naikrovek: been there, done that! but libspeex.so.1 is mising |
| 16:22.10 | Katty | your clothes may be all grummelly!! |
| 16:22.12 | Naikrovek | oldbug: i've never build speex so i dunno |
| 16:22.19 | Katty | but you're never fully dressed without a smile!!! |
| 16:24.10 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 16:24.31 | Katty | ohaider. |
| 16:24.49 | *** join/#asterisk Znuff (~ibm86@89.165.131.103) |
| 16:26.08 | *** join/#asterisk moy (~moy@74.12.129.100) |
| 16:26.14 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 16:26.35 | Katty | hi bmoraca |
| 16:29.00 | *** join/#asterisk mikkel (~mikkel@84-238-113-66.u.parknet.dk) |
| 16:29.45 | beek | hugs Katty |
| 16:30.10 | Polysics | anyway, no definite response on whether to use Adhearsion or go with my own daemon |
| 16:30.23 | Polysics | i guess I will start with the daemon then see what happens |
| 16:31.31 | Katty | hugs benngard |
| 16:31.32 | Katty | oh |
| 16:31.34 | Katty | hugs beek |
| 16:32.01 | Katty | beek: let's go out for lunch |
| 16:33.32 | Pimmetje | Does someone know if it is possible to rout sip over some kind of proxy |
| 16:34.07 | Pimmetje | Your provider block some ports and if i could route the data over a external server that would be great :D |
| 16:34.32 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 16:35.09 | *** join/#asterisk m0t3jl (~m0t3jl@ip-41.galance.net) |
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| 16:36.42 | MAbbas | I am using Dial() to connect a call to an agent. Agent is free but I get following warning says Agent is busy. |
| 16:36.43 | MAbbas | [Feb 17 13:18:29] WARNING[25075]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'Agent' (cause 17 - User busy) |
| 16:37.21 | Katty | hmm |
| 16:37.51 | beek | Katty -- where would you like to go? |
| 16:38.16 | Katty | idk :< |
| 16:38.22 | Katty | some place not greasy |
| 16:38.56 | MAbbas | [TK]D-Fender, Any idea? |
| 16:39.55 | oldbug | jesus... got it! |
| 16:40.06 | oldbug | ...i think :) |
| 16:41.03 | Katty | considers buying a macaw |
| 16:41.08 | *** join/#asterisk Victor_Yure (~victor@unaffiliated/victoryure/x-837844) |
| 16:41.11 | bmoraca_work | Katty: don't do it |
| 16:41.21 | bmoraca_work | birds are expensive and very high maintenance |
| 16:41.33 | bmoraca_work | especially larger birds |
| 16:41.41 | Katty | that's the point of having a pet |
| 16:41.52 | Katty | pets are like kids. |
| 16:42.03 | bmoraca_work | maybe that's why i'm a cat person...leave some food out and they'll take care of themselves |
| 16:42.05 | ManxPower-work | only if you treat them like kids. |
| 16:42.06 | Katty | but, i already have a very large puppy, so i really don't need any more pets (= |
| 16:42.27 | Katty | ManxPower-work: i love animals. |
| 16:42.47 | Katty | i should go work at an animal sanctuary |
| 16:42.56 | ManxPower-work | You can love animals without treating them like kids. |
| 16:43.59 | Katty | i treat riddick like another person. |
| 16:44.14 | Katty | just as if he were my kid. |
| 16:44.34 | ManxPower-work | I treat my cat like another person, like a companion, not a kid. |
| 16:44.59 | [TK]D-Fender | MAbbas: I don't see youshowing me anything |
| 16:45.25 | Katty | ManxPower-work: don't twist my meaning please (= |
| 16:46.00 | MAbbas | <[TK]D-Fender>: wait, letme paste output on pastbin .. |
| 16:46.39 | ManxPower-work | Children are loud, annoying, evil, disease carrying creatures I don't have any interest in being around. Nothing like a pet. |
| 16:47.12 | Katty | heh |
| 16:47.33 | bmoraca_work | damnit...HP STILL doesn't have any Core i5-based ML300 series servers |
| 16:47.40 | Naikrovek | kids are freaking awesome |
| 16:47.43 | Naikrovek | awe-some |
| 16:47.51 | Naikrovek | they're beyond awesome actually |
| 16:47.55 | Naikrovek | there isn't a word |
| 16:47.56 | Katty | some kids are. |
| 16:48.01 | Katty | other kids...not so much |
| 16:48.07 | Polysics | i have both, but i actually prefer cats |
| 16:48.08 | Katty | but some kids are pretty awesome |
| 16:48.16 | Naikrovek | well some kids have shitty parents, and shitty parents = shitty kid |
| 16:48.18 | Naikrovek | kid(s) |
| 16:48.20 | Katty | Polysics: i enjoy dogs more than cats (= |
| 16:48.23 | Polysics | we have a great kid, we are very proud |
| 16:48.24 | Katty | Naikrovek: aye :< |
| 16:48.36 | Polysics | but i still like my norwegian forest cat more :-) |
| 16:48.41 | Katty | Polysics: lol |
| 16:48.47 | Polysics | she never got along with my wife's 2 cats |
| 16:49.00 | MAbbas | man pastebin is slow like tortize .. |
| 16:49.00 | Polysics | but she is fierce in defending the kid, even a little too much |
| 16:49.10 | Katty | i doubt i'll ever have kids. |
| 16:49.47 | Naikrovek | fierce defenders are smart. overprotecting is not |
| 16:49.49 | m0t3jl | Hi, how does Asterisk determine a fax call from a normal voice call? |
| 16:49.53 | [TK]D-Fender | bmoraca_work: i5? .... geez... i7 or bust |
| 16:50.09 | Polysics | Katty, well, that depends on a few factors, mostly 1: the right person to raise them with |
| 16:50.14 | [TK]D-Fender | m0t3jl: Fax tones |
| 16:50.20 | Polysics | rest are moot points as far as i am concerned |
| 16:50.21 | bmoraca_work | [TK]D-Fender: my customers don't need i7 most of the time, and i5s should be roughly 40% cheaper |
| 16:50.53 | m0t3jl | [TK]D-Fender, so it is actually possible to configure all incoming numbers to be able to receive both fax and voice calls? |
| 16:51.35 | Katty | Polysics: err |
| 16:51.35 | [TK]D-Fender | m0t3jl: depends. You haven't said WHAT you are receiving the calls over |
| 16:51.39 | Katty | Polysics: that's not the issue at all |
| 16:51.43 | Katty | Polysics: i just don't really want to have any kids |
| 16:51.52 | oldbug | ok, solved my problem |
| 16:51.54 | Polysics | Katty, that's even more of a point :-) |
| 16:52.08 | oldbug | for those who are just interested in this: |
| 16:52.24 | oldbug | ./configure --prefix=/usr/ solved this ;) |
| 16:52.25 | m0t3jl | [TK]D-Fender, it is not actually working yet, I am just creating a schema ;) But there would be some PSTN lines together with some VoIP lines. |
| 16:52.42 | [TK]D-Fender | m0t3jl: Still too vague |
| 16:52.47 | oldbug | thank you guys for your help! |
| 16:53.57 | *** join/#asterisk dpisites (~cheng@CPE002191f85581-CM001692568382.cpe.net.cable.rogers.com) |
| 16:54.06 | m0t3jl | [TK]D-Fender, What else would you like to hear? ;) |
| 16:54.33 | *** join/#asterisk ccesario (~ccesario@189-19-6-236.dsl.telesp.net.br) |
| 16:55.27 | Katty | dudddeeeeee |
| 16:55.40 | *** part/#asterisk dpisites (~cheng@CPE002191f85581-CM001692568382.cpe.net.cable.rogers.com) |
| 16:56.22 | *** part/#asterisk oldbug (~oldbug@apokryph.org) |
| 16:56.36 | aces1up | whats typically easier to work with, a pri or t1? |
| 16:56.38 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 16:56.46 | Katty | we use a pri here |
| 16:56.52 | Katty | tho i commonly see partial t1s |
| 16:57.06 | Katty | where they share a few phone lines and get their internets on the same connection |
| 16:58.30 | Katty | but its turned into analog lines at that point |
| 16:59.10 | spenguin[work] | i r bored |
| 16:59.26 | spenguin[work] | well tired working as of now |
| 16:59.28 | spenguin[work] | :s |
| 16:59.43 | MAbbas | <[TK]D-Fender>: as we discussed few days back. I am using AGI to calculae suitable agent for call .. take a look at my dial plan to get an idea http://pastebin.com/m72417885 |
| 17:00.23 | *** join/#asterisk cesar_CR (~cesar@201.192.86.30) |
| 17:00.32 | Katty | spenguin[work]: let's go have lunch! |
| 17:00.47 | Katty | spenguin[work]: before i procrastinate another hour |
| 17:00.58 | spenguin[work] | heh Katty, I am waiting for dinner to arrive :p |
| 17:01.01 | MAbbas | Now I am load testing this thing and when I try to Dial() agent I get following warnings and calls go to Queue() ..http://pastebin.com/m47f901e3 |
| 17:01.05 | spenguin[work] | but whats for lunch! |
| 17:01.14 | Katty | spenguin[work]: i hear food might be a good choice |
| 17:01.31 | spenguin[work] | :p |
| 17:01.45 | spenguin[work] | has live fish |
| 17:01.54 | Katty | live? ^_- |
| 17:01.57 | Katty | creepy. |
| 17:01.57 | spenguin[work] | like any other penguin. |
| 17:02.02 | Katty | ha |
| 17:02.24 | m0t3jl | [TK]D-Fender, The PSTN lines would most probably be handled by some PSTN2VoIP gateway. |
| 17:10.08 | *** join/#asterisk Heretic (~Fallen@dsl-246-79-211.telkomadsl.co.za) |
| 17:10.56 | eppigy | TRABAJO |
| 17:11.03 | spenguin[work] | hi eppigy |
| 17:11.03 | *** join/#asterisk rossand (~aross@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 17:11.07 | eppigy | hello |
| 17:11.40 | spenguin[work] | sup |
| 17:12.02 | *** join/#asterisk fgs (~fgsch@87-194-38-117.bethere.co.uk) |
| 17:12.05 | fgs | hi |
| 17:12.15 | fgs | i'm using ael to define my dialplan |
| 17:12.27 | fgs | and i'm finding something weird wrt includes |
| 17:12.39 | fgs | i have a context foo, where i have extension 1000 |
| 17:12.56 | fgs | in context bar i include bar |
| 17:13.01 | MAbbas | <[TK]D-Fender>: I am leaving for today. Lets discuss this thing tommorow .. |
| 17:13.06 | fgs | and define _X. as a catchup all, almost |
| 17:13.20 | fgs | but _X. matches 1000 |
| 17:13.26 | *** join/#asterisk oej_ (~olle@95.209.10.223.bredband.tre.se) |
| 17:13.31 | fgs | if i define it locally (not via includes) it works |
| 17:13.40 | fgs | sounds like a bug, but not sure. am i missing something? |
| 17:13.51 | fgs | this is 1.6.0.22 |
| 17:15.07 | ManxPower-work | Yes, _X. will match "1000" |
| 17:15.48 | *** join/#asterisk catojo (~catojo@189.24.85.1) |
| 17:15.51 | ManxPower-work | Remember, include'd exten =>s will be the lowest priority, so if anything could match the number in the current context, then that will match. |
| 17:16.02 | ManxPower-work | order of includes does not (usually) matter for this. |
| 17:17.13 | fgs | ok, but ael does not use priorities |
| 17:17.21 | fgs | so how to make it match 1000? |
| 17:17.49 | *** join/#asterisk terracon (~greisky@CPE001cf097a750-CM0012254076d6.cpe.net.cable.rogers.com) |
| 17:20.23 | ManxPower-work | put 1000 in the context or out both 1000 and _X, in included contexts. |
| 17:20.50 | fgs | yeah, that's what i did |
| 17:21.02 | fgs | so no way to do it having one defined in a different context and one in the current context? |
| 17:21.29 | m0t3jl | Should it be possible to receive faxes using the ReceiveFAX function from a SIP connection? |
| 17:24.42 | *** join/#asterisk garymc (~chatzilla@host86-164-36-128.range86-164.btcentralplus.com) |
| 17:28.50 | ManxPower-work | fgs: yes, but if it CAN match in the current context then it WILL. |
| 17:28.55 | ManxPower-work | m0t3jl: no |
| 17:30.37 | rubberneck | m0t3jl: I have had a lot of problems trying to get fax to work. you'll need a lot of luck and a lot of patients. |
| 17:30.50 | Naikrovek | patience, too |
| 17:30.51 | Qwell | rubberneck: we've got plenty of patients here |
| 17:31.01 | rubberneck | heh. oops |
| 17:34.29 | m0t3jl | ManxPower-work, so it's better to have special PSTN lines completely separated from Asterisk PBX? |
| 17:34.59 | ManxPower-work | m0t3jl: Have a standard analog line for fax. It is by far the most reliable way. |
| 17:37.38 | m0t3jl | ManxPower-work, that's what I was trying to suggest in my previous message ;) |
| 17:38.35 | ManxPower-work | m0t3jl: Is English your first language? |
| 17:39.18 | ManxPower-work | special could mean "different" (your usage) or "non-standard". |
| 17:39.28 | ManxPower-work | I was thinking you meant non-standard. |
| 17:40.04 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 17:40.07 | fgs | manxpower: ok, ta |
| 17:41.07 | m0t3jl | ManxPower-work, got me ;) No, my first language is Czech... You're right, I should've used ... to have PSTN lines completely separated ... |
| 17:41.42 | raden_work | why when i hit the keys on my phone real fast on my phone does the IVR not recognize the numbers ? |
| 17:42.41 | [TK]D-Fender | RadenWaht phone? call arriving on * how>? |
| 17:43.04 | ManxPower-work | m0t3jl: Your English is better than many people with only English as their language. |
| 17:43.18 | ManxPower-work | [TK]D-Fender: I think he's rolling. |
| 17:43.45 | ManxPower-work | and trolling too. |
| 17:43.56 | *** join/#asterisk dpisites (~cheng@CPE002191f85581-CM001692568382.cpe.net.cable.rogers.com) |
| 17:44.01 | *** part/#asterisk dpisites (~cheng@CPE002191f85581-CM001692568382.cpe.net.cable.rogers.com) |
| 17:45.06 | raden_work | ok looking at debug its not catching all the digits |
| 17:45.33 | raden_work | but i hear them on my phone any ideas ? |
| 17:45.41 | m0t3jl | ManxPower-work, nicely said ;) Thanks, though. |
| 17:46.50 | *** join/#asterisk bzing2 (~dr105@dhcp-194-66-208-235.canterbury.ac.uk) |
| 17:48.01 | p3nguin_ | Is there any way to write a newline or carriage return in Verbose()? |
| 17:48.38 | bmoraca_work | p3nguin_: another Verbose() statement? |
| 17:48.39 | plundra | Verbose()\nVerbose()? :-) |
| 17:48.58 | [TK]D-Fender | raden_work: Hello.... DETAILS.... |
| 17:49.08 | p3nguin_ | bmoraca_work: Another Verbose would do it, but completely defeats the purpose of my question. |
| 17:49.20 | plundra | There probably is some statement/function that can evaluate a \n and return it. |
| 17:49.40 | p3nguin_ | Verbose itself cannot, though? |
| 17:49.45 | bmoraca_work | p3nguin_: Occum's Razor. that said, I don't believe that \n works in Verbose() so there probably isn't a way to do it |
| 17:50.43 | p3nguin_ | That's too bad. I think that would be a good option to have. |
| 17:52.34 | raden_work | <PROTECTED> |
| 17:52.34 | raden_work | <PROTECTED> |
| 17:52.46 | raden_work | for example thats password 5050 for user 500 |
| 17:52.50 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 17:52.50 | raden_work | did that 2 times in a row |
| 17:53.02 | p3nguin_ | DTMF not registering correctly? |
| 17:53.04 | raden_work | does it for all our extensions if they keys are dialed to fast |
| 17:53.13 | raden_work | p3nguin_, how would I correct that ? |
| 17:53.32 | p3nguin_ | Heck, I don't know. |
| 17:53.49 | raden_work | p3nguin_, if i dial slow like a 4 digits in 2 seconds its fine but if i got 5050 in like a second it misses a digit |
| 17:54.00 | raden_work | or am i just that fast ? |
| 17:54.12 | [TK]D-Fender | RadenNo... not too swift ;) |
| 17:54.22 | raden_work | lol :) |
| 17:54.25 | raden_work | [TK]D-Fender, any ideas ? |
| 17:54.29 | [TK]D-Fender | hates it when even 5 cars isn't enough to get an autocomplete |
| 17:54.39 | raden_work | [TK]D-Fender, and i check the call log on phone the correct numbers are getting hit |
| 17:54.48 | [TK]D-Fender | raden_work: I asked you for DETAILS twice now. |
| 17:54.59 | raden_work | [TK]D-Fender, what details would you like ? |
| 17:55.11 | [TK]D-Fender | raden_work: how about the pertinent ones. |
| 17:56.12 | raden_work | [TK]D-Fender, If I dial in our voicemail and enter numbers to fast it does not catch all them put 5050 in really fast itll catch 500 or 500 for user might come up as 00 |
| 17:56.17 | bmoraca_work | besides HP and dell, who would be the other Tier 1 server manufacturers? |
| 17:56.27 | raden_work | and the screen on my phone is showing that im dialing them. |
| 17:57.11 | raden_work | a few people have also complained when they call the IVR that they press a number and it restarts, only way i can make this happen is if i tap a key very fast and i mean just tap it |
| 17:57.28 | raden_work | this happens from VOIP phones on LAN our wan phones and land lines alike |
| 17:57.39 | raden_work | bmoraca_work, IBM ? |
| 17:57.46 | plundra | Any ideas on how to check if a Queue have a working member with a certain penalty? Irregarding (uhm?) if he is busy or not. |
| 17:58.33 | [TK]D-Fender | raden_work: Lets try this again... you don't seem to understand "pertinent" |
| 17:59.08 | plundra | Because I want to set a MAX_PENALTY, so if a certain member actually is available for calls, no one over it gets called. |
| 17:59.08 | raden_work | [TK]D-Fender, what would pertinent details be ? |
| 17:59.23 | Geminizer | when I access a dialplan from my cell phone (calling into the pbx via a DID), it works when my context has the following "exten => _NXXNXXXXXX,n, ...." but not "exten => s,n...". I thought "s" was meant to represent "any extension" (wildcard). |
| 17:59.25 | raden_work | <raden_work> -- Incorrect password '500' for user '500' (context = default) |
| 17:59.25 | raden_work | <raden_work> -- <SIP/103-b6804ab0> Playing 'vm-incorrect-mailbox.gsm' (language 'en') |
| 17:59.37 | raden_work | I dialed 5050 and 500 according to the screen on my phone |
| 17:59.46 | raden_work | thats SIP debug ^^^^^^^^ |
| 17:59.47 | plundra | It's better the caller gets to wait, rathern then having to speak with someone in a different language. |
| 18:00.04 | [TK]D-Fender | raden_work: no that isn't SIP DEBUG |
| 18:00.09 | p3nguin_ | plundra: That's how penalties already work. |
| 18:00.18 | [TK]D-Fender | raden_work: what PHONE MODEL, and what DTMF MODE? |
| 18:00.21 | ManxPower-work | Geminizer: no, "s" means "no extension" |
| 18:00.37 | [TK]D-Fender | ~stdextens |
| 18:00.38 | infobot | [~stdextens] The "s" Standard Extension : Where a call goes to when * does not know the destination of the call. Ex : Calls coming in on FXO ports (no DID), or from an ITSP that doesn't specify or where it was not set in the REGISTER line, or FXS port goes off-hook and "immediate=yes" in zapata.conf. "s" is also used to make IVRs & macros. |
| 18:00.39 | p3nguin_ | manxpower-work: unless the extension is 's' |
| 18:00.40 | [TK]D-Fender | Geminizer: ^^^ |
| 18:01.02 | Chainsaw | raden_work: The main problem so far in your interaction with Fender is that you're redacting your CLI output. |
| 18:01.07 | plundra | p3nguin_: If I've understood it correct, if all penalty 1 members are busy, it will try higher penalty members, isn't tha ttrue? |
| 18:01.08 | p3nguin_ | geminizer: s is certainly not a wildcard. |
| 18:01.19 | Chainsaw | raden_work: Please use a pastebin site so there is ample room for the whole thing. Redacted output is not acceptable. |
| 18:01.20 | raden_work | Astra 9133i DTMF METHOD RTP |
| 18:01.21 | Geminizer | ahh, ok... |
| 18:01.55 | plundra | p3nguin_: As long as there is a member, busy or not, just as long as it's reachable and unpaused, I want to keep the caller on that penalty level. |
| 18:02.30 | plundra | (The voip-info-wiki might be wrong though :-P I'll try finding it in the real docs) |
| 18:02.48 | p3nguin_ | plundra: If you have some members that are all penalty 1, and some that are penalty 2, and more that are penalty 3... if a single penalty 1 member is available, the call will not go to penalty 2 nor 3. |
| 18:03.38 | plundra | p3nguin_: But what defines available? If he's busy with a call, is he still available? |
| 18:04.30 | [TK]D-Fender | raden_work: try another mode like SIP INFO |
| 18:04.40 | Chainsaw | plundra: It depends on your maxcalls setting. But yes, with call waiting enabled you can end up in the awkward situation where calls ring in as a second conversation with the busy agent. |
| 18:04.41 | raden_work | OK |
| 18:04.42 | Geminizer | also, how would a context recognize when someone presses the pound (#) key ? |
| 18:05.08 | Chainsaw | Geminizer: It's a terminator, so you wouldn't normally receive that DTMF character. |
| 18:05.31 | raden_work | [TK]D-Fender, SIP INFO or BOTH are my other options |
| 18:05.36 | Chainsaw | Geminizer: So you can't have it as an extension to jump to, you'd have to take input. |
| 18:06.00 | [TK]D-Fender | Geminizer: Context's don't detect keys... IVR'sdo. |
| 18:06.00 | p3nguin_ | plundra: If you have the call-limit set to a high enough number and the phone can accept more than one call, the member should remain available. |
| 18:06.12 | [TK]D-Fender | Geminizer: And # is just another char to match |
| 18:06.45 | Chainsaw | p3nguin_: I tried a call limit of 1 which neatly broke attended transfer for my 7960s :( |
| 18:06.58 | p3nguin_ | ick |
| 18:07.24 | Chainsaw | p3nguin_: Because I do recognise the problem of calls ending up with busy agents. |
| 18:07.25 | p3nguin_ | I think there is commenting in the sample config regarding that. |
| 18:07.27 | raden_work | [TK]D-Fender, switch to SIP INFO flawless now |
| 18:07.51 | Chainsaw | raden_work: I take it you weren't on ulaw or alaw then. |
| 18:08.08 | p3nguin_ | Something about call-limit should be at least 2 for some condition. I can't remember exactly what it said or where it was documented. |
| 18:08.12 | raden_work | Chainsaw, Ulaw |
| 18:08.33 | ManxPower-work | raden_work: did you have "relaxdtmf=yes"? |
| 18:08.52 | Chainsaw | raden_work: Hm, okay. Can't explain the failure mode then. |
| 18:08.52 | Chainsaw | p3nguin_: My config file now has a neat comment about how it breaks attended transfers and shouldn't be <2. |
| 18:09.35 | plundra | Chainsaw, p3nguin_: Hmm, call-limit is said to disappear, and I don't know if the phones would be great with multiple calls on the same time. |
| 18:09.52 | ManxPower-work | call-limit is being REPLACED |
| 18:10.20 | Chainsaw | p3nguin_: I have Polycom 670 handsets planned anyway. I expect they'll deal with it gracefully. |
| 18:10.24 | plundra | I am using a local/-channel in the queues, maybe I could just dial again and again? Isn't very pretty I guess though. |
| 18:10.39 | p3nguin_ | plundra: If you don't want multiple calls on a phone which is a queue member, don't try to circumvent the penalties. |
| 18:10.46 | bmoraca_work | ManxPower-work: really? what's replacing it? |
| 18:11.11 | ManxPower-work | bmoraca_work: look in UPGRADE*.txt |
| 18:11.14 | *** join/#asterisk catojo (~catojo@189.24.85.1) |
| 18:11.22 | angryuser | can someone tell me about the biggest voip install ever ? I head some university did 20k users with asterisk and opensips kamalio, they even had a site, someone know it's url ? |
| 18:11.26 | bmoraca_work | ManxPower-work: latest svn or branch? |
| 18:11.31 | plundra | p3nguin_: I want to keep the caller in the queue, I don't see that as circumventing anything. |
| 18:11.36 | ManxPower-work | bmoraca_work: 1.6.mumble.mumble |
| 18:11.58 | ManxPower-work | bmoraca_work: I never run SVN |
| 18:12.06 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 18:12.12 | plundra | But to sum one thing up, no functions/applications for looking up queue-members in an dialplan-friendly way? :) |
| 18:12.17 | p3nguin_ | angryuser: "its url" |
| 18:12.36 | Cain | Hi, i want to route the phone calls from pstn to ip phones, i told i have to take a pap2t and install asterisk to handle it, is pap2t able to do it or i need something else ? |
| 18:12.38 | bmoraca_work | well, i had to ax |
| 18:13.02 | angryuser | p3nguin_, yes, you heard about it ? |
| 18:13.05 | ManxPower-work | Cain: the PAP2 has FXS ports, you need FXO ports for PSTN lines. |
| 18:13.07 | p3nguin_ | angryuser: Nope. |
| 18:13.12 | Katty | hi |
| 18:14.04 | Katty | spenguin[work]: ended up having Arby's |
| 18:14.08 | Katty | spenguin[work]: do you have that chain out there? |
| 18:14.13 | p3nguin_ | cain: Look for the SPA-3102 (or the deprecated SPA-3000) for an ATA which has the right port type. |
| 18:14.28 | ManxPower-work | it only has ONE FXO, however. |
| 18:14.39 | p3nguin_ | One is certainly better than zero. |
| 18:14.49 | Cain | sure ! |
| 18:14.57 | Katty | can think of instances where zero is better than one. |
| 18:15.22 | Cain | dropes Pap2t from the window ! |
| 18:15.26 | p3nguin_ | Context is important here, my dear katty. |
| 18:15.37 | Katty | psh |
| 18:15.42 | Katty | since when are my comments ever in context. |
| 18:15.49 | Katty | let's not be riddicurus. |
| 18:16.13 | Katty | also |
| 18:16.19 | Katty | i'm making Flounder for dinner. |
| 18:16.24 | Katty | anyone have a fish recipe that i could use? |
| 18:16.43 | Qwell | pour salt on fish, put fish in oven |
| 18:16.47 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 18:16.52 | p3nguin_ | cain: The SPA-3102 will run you about $70 USD, but you can find the obsoleted SPA-3000 for around $40 USD if you look in the right places. |
| 18:16.53 | *** join/#asterisk fskrotzki_ (~fskrotzki@cpe-74-74-245-250.rochester.res.rr.com) |
| 18:17.31 | Cain | p3nguin_ so for my use i don't need a pap2t at all ? :'( (i'v bought 2 !) |
| 18:17.44 | *** join/#asterisk fofware (~chatzilla@190.7.25.160) |
| 18:17.58 | bmoraca_work | i did mahi mahi in a ginger glaze...it was really good...but i don't know how mahi mahi compares to tuna in terms of texture or taste, so i couldn't say whether or not it would be as good |
| 18:18.01 | Katty | Qwell: that doesn't sound very tastey |
| 18:18.07 | p3nguin_ | cain: That's correct. The PAP2T can only hook up a phone or fax machine. |
| 18:18.17 | Qwell | Katty: you didn't ask for a good one.. |
| 18:18.25 | p3nguin_ | katty: Something lemon pepper, maybe? |
| 18:18.31 | Katty | Qwell: :P |
| 18:18.34 | Katty | Qwell: okay smart tail |
| 18:18.49 | bmoraca_work | er, compares to flounder |
| 18:18.52 | Katty | p3nguin_: i'm browsing recipezaar and found a parmesan tilapia recipe |
| 18:19.11 | Katty | bmoraca_work: hmm. any chance you've got the recipe laying around somewhere? |
| 18:19.41 | p3nguin_ | cain: The PAP2T provides two FXS ports (like manxpower-work mentioned before), which are for connecting phone devices to. |
| 18:20.40 | p3nguin_ | cain: Since the FXS ports provide voltage to the phones, if you connect it to your wall jack (which also provides voltage), you could damage the ATA. |
| 18:21.01 | *** join/#asterisk oej (~olle@109.58.50.127.bredband.tre.se) |
| 18:21.02 | bmoraca_work | Katty: not at work, no...but it was fairly easy...honey, soy sauce, sugar, balsamic vinegar, fresh garlic and ginger, marinade for like 30 minutes, cook fish in skillet with oil, add remaining marinade in last few minutes, and then keep marinade going to reduce down to glaze...season to taste... |
| 18:21.21 | Katty | nods |
| 18:21.31 | Katty | when do you get off work? |
| 18:21.36 | eppigy | NEVER |
| 18:21.39 | Katty | IT"S DAVE |
| 18:21.40 | Katty | OMG |
| 18:21.42 | Katty | faints |
| 18:21.42 | eppigy | HI |
| 18:21.45 | Katty | hugs eppigy |
| 18:21.46 | eppigy | catches Katty |
| 18:21.48 | bmoraca_work | Katty: http://recipes.sparkpeople.com/recipe-detail.asp?recipe=335578 |
| 18:21.52 | bmoraca_work | that's the recipe i used |
| 18:21.52 | eppigy | hugs Katty |
| 18:21.52 | Katty | where haveee you been?!?!?!?!?!!! :> |
| 18:21.58 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 18:22.00 | eppigy | workin and stuff |
| 18:22.02 | eppigy | stressin |
| 18:22.04 | eppigy | RAGING |
| 18:22.09 | Katty | bmoraca_work: thanks |
| 18:22.13 | Katty | eppigy: eww :< |
| 18:22.15 | Katty | eppigy: that's no good. |
| 18:22.17 | eppigy | YES |
| 18:22.26 | raden_work | hey eppigy :) |
| 18:22.28 | Katty | eppigy: are you destressing yet? |
| 18:22.32 | raden_work | how goes it Katty |
| 18:22.38 | eppigy | Katty: kinda |
| 18:22.45 | *** join/#asterisk benngard (~benngard@213.88.138.230) |
| 18:22.49 | Katty | bmoraca_work: do you think i could get away with ground ginger instead of ginger root? |
| 18:23.07 | Katty | bmoraca_work: or is fresh ginger infinately better? |
| 18:23.16 | Katty | raden_work: ehhhh |
| 18:23.18 | bmoraca_work | Katty: it wouldn't taste as good, but probably. just make sure you halve the amount. |
| 18:23.20 | Katty | raden_work: pretty good. |
| 18:23.26 | Katty | bmoraca_work: yeah. |
| 18:23.32 | Katty | bmoraca_work: well i'll give this recipe a try (= |
| 18:23.36 | Katty | bmoraca_work: i've got all the ingredients at home |
| 18:23.45 | bmoraca_work | it's a pretty simple one, yeah |
| 18:24.11 | bmoraca_work | just remember that this was meant to be made with mahi mahi, a very light fish. i'm not sure how flounder compares, so the flavors might not mingle right. |
| 18:24.12 | Katty | bmoraca_work: do you think it would be better to cook the fish IN the ginger glaze? |
| 18:24.24 | Katty | bmoraca_work: or to boil it in a seperate pan and then pour it over the fish |
| 18:24.35 | raden_work | bmoraca_work, I have master working beautiful now is there a way to add a 3rd master and have a script or something if one fails to switch what master the master is listening to ? |
| 18:24.40 | bmoraca_work | Katty: no...texture wouldn't come out right. boil it in the same pan after the fish is done (deglaze) |
| 18:24.55 | Katty | bmoraca_work: but with fish still in there? |
| 18:24.59 | raden_work | bmoraca_work, master master in mysql for asterisk realtime |
| 18:25.01 | Katty | bmoraca_work: or will the fish start to burn? |
| 18:25.04 | bmoraca_work | Katty: up to you. |
| 18:25.11 | Katty | bmoraca_work: k |
| 18:25.14 | bmoraca_work | Katty: probably take the fish out first |
| 18:25.17 | Katty | nods |
| 18:25.21 | Katty | what did you serve with this? |
| 18:25.25 | Katty | eppigy: have you chatrouletted? |
| 18:25.33 | bmoraca_work | Katty: 30 minutes of marinade with fish is a lot of flavor. |
| 18:25.47 | eppigy | Katty: I have not |
| 18:25.52 | bmoraca_work | Katty: i served it with au gratin potatos, lol |
| 18:25.55 | Katty | bmoraca_work: yeah it looks like it'll be great for what i'm wanting. if the glaze is a keeper, i'll try it with chicken and what not |
| 18:25.59 | Katty | ^_- |
| 18:26.06 | Katty | oooookay |
| 18:26.11 | bmoraca_work | Katty: but plain jasmine rice would work well too |
| 18:26.12 | Katty | weird. |
| 18:26.18 | Katty | no veggies? |
| 18:26.20 | bmoraca_work | i didn't have any at the time and i really wanted some potatos...so... |
| 18:26.24 | Katty | ah |
| 18:26.30 | bmoraca_work | steamed veggies are always good |
| 18:26.38 | Katty | i was thinking peas |
| 18:26.39 | bmoraca_work | broccoli would go well, i think |
| 18:26.46 | Katty | and perhaps mac n cheese |
| 18:26.47 | bmoraca_work | i'm not a big fan of peas |
| 18:27.00 | Katty | you're just weird. |
| 18:27.17 | Katty | eppigy: you must chatroulette with me sometime |
| 18:27.30 | Katty | eppigy: i will make funny faces at you |
| 18:27.36 | eppigy | i thought chatroulette was random? |
| 18:27.39 | Katty | it is |
| 18:27.43 | Katty | you will have to FIND ME |
| 18:27.45 | eppigy | oh |
| 18:27.47 | eppigy | OH |
| 18:27.50 | Qwell | just click next 19,000 times |
| 18:27.52 | Katty | or we could redditroulette |
| 18:27.57 | bmoraca_work | raden_work: probably could, sure. i'd have to look at your implementation in detail to begin to speculate on how to do this. although, i was wrong when I said DUNDI wouldn't be useful...it's expanded since the last time i looked at it and regexten would make it very possible to use (though more difficult to maintain) |
| 18:27.58 | Katty | that would make it infinately easier |
| 18:28.11 | Katty | yeah you can pretend you're at an aiport and hold a Katty sign |
| 18:28.27 | raden_work | bmoraca_work, well i have the DUNDI stuff installed |
| 18:28.33 | raden_work | I have 2 servers right now |
| 18:28.43 | Katty | Qwell: no one wants to talk to me on chatroulette |
| 18:28.48 | Katty | Qwell: except old creepy guys |
| 18:28.49 | Qwell | why? |
| 18:28.55 | Katty | Qwell: i'm probably not hotttt enough |
| 18:29.04 | [TK]D-Fender | Katty: steamed = fail. broccoli & peppers on a skillet on high with just a dab of oil. Get them where thy still hve their nutritional value & crunch |
| 18:29.22 | raden_work | mysql master master b listens to a and a listens to be ran a offset of 10 and a increment of 1 on a and 2 on b so no index overwrites and running everything Innodb so its transaction safe |
| 18:29.26 | bmoraca_work | raden_work: i haven't looked at how dundi has changed in 1.6 yet, but some of the changes i've looked in to make it look as though it'll work well with what you're trying to do |
| 18:29.28 | Katty | [TK]D-Fender: i've saving my green peppers to make stuffed peppers later in the week (= |
| 18:29.40 | Katty | [TK]D-Fender: with turkey and brown rice |
| 18:29.44 | Katty | [TK]D-Fender: it'll be goooooood |
| 18:29.45 | [TK]D-Fender | kayBy more. cause thats what MORE means! |
| 18:29.47 | Katty | GOOD |
| 18:29.49 | [TK]D-Fender | Buy* |
| 18:29.58 | Katty | yeah i could always buy more |
| 18:30.05 | Katty | it's not like the grocery store is running out of peppers |
| 18:30.38 | Katty | [TK]D-Fender: do you use a cast iron skillet for that? |
| 18:30.44 | Katty | [TK]D-Fender: with oil and water |
| 18:30.54 | Katty | [TK]D-Fender: possibly vinegar |
| 18:31.03 | Katty | [TK]D-Fender: or a nonstick skillet? |
| 18:31.11 | [TK]D-Fender | Katty: I'd recommend sesame oil. Use lightly... |
| 18:31.25 | [TK]D-Fender | Katty: It shouldn't sit around long enough to stick. |
| 18:31.29 | Katty | [TK]D-Fender: yeah about 1 t will saute a couple big onions |
| 18:31.34 | Katty | k |
| 18:31.39 | Katty | just charing it then? |
| 18:31.47 | Katty | BLACKENED PEPPER |
| 18:31.53 | Katty | MmMMmmmm |
| 18:31.57 | Katty | that sounds good |
| 18:32.09 | Katty | verra nice |
| 18:32.18 | bmoraca_work | Katty: i'm going to make zucchini with green peppers and mushrooms, sauteed in marinara sauce |
| 18:32.19 | [TK]D-Fender | Katty: If even... it'll cook crisp, and you should get some very light browning here and ther. the goal isn't to deep-fry them :) |
| 18:32.31 | Katty | bmoraca_work: i have an awesome, very simple, roasted garlic mushroom recipe |
| 18:32.37 | [TK]D-Fender | Katty: Also a great way to do asperagus. |
| 18:32.37 | bmoraca_work | usually do it with italian sausage, but i have zucchini right now, and not italian sausage |
| 18:32.43 | Katty | [TK]D-Fender: ugah, asperagus |
| 18:32.47 | Katty | [TK]D-Fender: ugggahhhhhh /gag |
| 18:33.02 | bmoraca_work | [TK]D-Fender: Guy Fieri's beer battered asparagus is really good |
| 18:33.03 | [TK]D-Fender | Katty: No, pan-friend in sesame is great. Boiled/steamed = shit |
| 18:33.07 | Katty | bmoraca_work: not a big fan of italian sausage :< |
| 18:33.17 | bmoraca_work | Katty: BLASPHEMER! |
| 18:33.23 | Katty | bmoraca_work: i sorry :< |
| 18:33.26 | Katty | bmoraca_work: but it's true |
| 18:33.36 | Katty | bmoraca_work: it's greasy |
| 18:33.42 | spenguin[work] | Katty: nope we have no arbys |
| 18:33.53 | Katty | bmoraca_work: it's like you pick it up, and the grease is dripping off of it still |
| 18:33.55 | bmoraca_work | i'm italian, so i have to like it. but, really, a good spicy italian sausage sauteed with onions and bell peppers is fantastic and not greesy at all |
| 18:34.00 | beek | is very disheartened to learn of Katty's aversion to asparagus and italian saugage. |
| 18:34.01 | Katty | spenguin[work]: do you have any place that sells roast beef? |
| 18:34.16 | spenguin[work] | nope, well some places but rate |
| 18:34.19 | spenguin[work] | err rare |
| 18:34.51 | Katty | beek: why? it's not a bad thing. i also don't like beets. |
| 18:34.56 | [TK]D-Fender | beek: Today's catch-phrase is "tragically white" :p |
| 18:34.57 | bmoraca_work | Katty: i did some cornish game hens a couple weeks ago basted and slowcooked with oranges...they were pretty good |
| 18:35.09 | Katty | spenguin[work]: ah, well that's what Arby's is basically. it's a roast beef sammich chain resturant. |
| 18:35.27 | Katty | bmoraca_work: aren't those still whole? |
| 18:35.29 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
| 18:35.29 | Katty | bmoraca_work: with bones |
| 18:35.43 | Katty | bmoraca_work: seems like a lot of work |
| 18:35.45 | *** join/#asterisk ttl_ (~patrick@d5153A420.access.telenet.be) |
| 18:35.45 | bmoraca_work | Katty: yes. no guts though. they were cleaned when i bought them. |
| 18:36.01 | Katty | bmoraca_work: ah, well then that's easier |
| 18:36.08 | bmoraca_work | Katty: it was...but not NEARLY as much work as the cheese-stuffed pretzels i made this weekend |
| 18:36.15 | Katty | bmoraca_work: i'm lazy. i'd rather just stick chicken tenderloins in the crockpot |
| 18:36.37 | spenguin[work] | anyone here eat bull frogs :p |
| 18:36.43 | Katty | bmoraca_work: ohhh that sounds good. string cheese wrapped in dough? and then salted? |
| 18:36.47 | Katty | spenguin[work]: WHAT?! |
| 18:36.51 | Katty | spenguin[work]: frog legs? |
| 18:36.54 | spenguin[work] | loves bull frog soup |
| 18:36.56 | spenguin[work] | well yeah :s |
| 18:36.59 | *** join/#asterisk ttl_ (~patrick@d5153A420.access.telenet.be) |
| 18:37.00 | Katty | no. |
| 18:37.03 | spenguin[work] | why |
| 18:37.04 | Katty | but they serve them. |
| 18:37.10 | spenguin[work] | l |
| 18:37.11 | Katty | idk, it just sounds creepy |
| 18:37.18 | spenguin[work] | k, its tastier than chicken |
| 18:37.28 | bmoraca_work | Katty: no...shredded cheddar cheese sandwiched in the dough and salted. they were pretty good, but it was a LOT of work, and pretzel dough is really hard to work with |
| 18:37.34 | Katty | yeah but chicken tenderloins no longer resemble a real chicken :P |
| 18:37.53 | Katty | bmoraca_work: yeah it is. |
| 18:37.58 | Katty | bmoraca_work: you get any photos? |
| 18:38.14 | *** join/#asterisk simplydrew_ (~simplydre@pool-74-97-190-109.prvdri.fios.verizon.net) |
| 18:38.33 | Katty | spenguin[work]: what other kinda weird stuff do you eat? |
| 18:39.01 | spenguin[work] | rabbit meat, tortoise meat |
| 18:39.08 | Katty | tortoise? |
| 18:39.10 | spenguin[work] | but none of these often |
| 18:39.11 | spenguin[work] | yeah |
| 18:39.20 | Katty | i don't think i can get that anywhere around here. |
| 18:39.34 | spenguin[work] | Theres ostrich meat available |
| 18:39.35 | Katty | my dad raises meat rabbits |
| 18:39.49 | Katty | there's bison around here. and you can commonly get Elk |
| 18:39.53 | spenguin[work] | heh |
| 18:40.03 | Katty | not sure about ostrich. might be some up in STL tho |
| 18:40.16 | Katty | Bison is awesome |
| 18:40.21 | Katty | very very lean. |
| 18:40.25 | Katty | no wild taste at all |
| 18:40.26 | spenguin[work] | nice |
| 18:40.34 | Katty | little more expensive than beef. |
| 18:40.41 | spenguin[work] | are they bred |
| 18:40.42 | spenguin[work] | ? |
| 18:40.44 | Katty | but they're not fed corn, and shot up with antibiotics either |
| 18:40.52 | spenguin[work] | thats cool |
| 18:40.54 | Katty | yeah. there are a few Bison ranches around here |
| 18:41.06 | Katty | we have a place called the Crown Valley Winery |
| 18:41.15 | Katty | It's a wine place, plus a resturant |
| 18:41.25 | Katty | they raise Bison, and serve bison on their menu |
| 18:41.33 | Katty | fresh daily. they told me they go through about a bison a day |
| 18:41.54 | bmoraca_work | Katty: i don't ahve any pictures of the pretzels, no. the kitchen was a MESS, though, hehe |
| 18:41.54 | spenguin[work] | cool |
| 18:41.57 | Katty | you can also buy Next Day ground bison, which is the ground up leftovers from the previoius day at the resturant |
| 18:42.04 | spenguin[work] | meat goes well with wine |
| 18:42.07 | Katty | bmoraca_work: my kitchen just stays a mess >.< |
| 18:42.09 | bmoraca_work | Fuddrucker's has bison burgers...i've been wanting to try them |
| 18:42.20 | Katty | bmoraca_work: do they? hmm. i'll have to check that out |
| 18:44.28 | [TK]D-Fender | bison = great. So is horse... |
| 18:44.32 | *** join/#asterisk superbeef (~lanej@74.84.194.4) |
| 18:44.33 | Katty | never had horse. |
| 18:44.39 | Katty | SUPERBEEF |
| 18:44.41 | Katty | dundundunDUN |
| 18:45.06 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 18:45.24 | bmoraca_work | i got some funnel cake mix and the little ring for funnel cakes...can't wait to try them |
| 18:46.02 | ecrane | I cook dinner by setting up the rice cooker and throwing a can of beans on top of the rice before hitting the start button. |
| 18:46.03 | mmattice | what would cause a call into a PRI to show up, * appear to answer it, and then suddenly have the D Channel go down and come right back up? |
| 18:48.45 | ManxPower-work | mmattice: a variety of things |
| 18:48.59 | mmattice | what should I look at to narrow it down? |
| 18:49.16 | ManxPower-work | mmattice: put a copy of the CLI output of a failed call on pastebin.ca |
| 18:49.55 | superbeef | Katty! |
| 18:53.50 | mmattice | ManxPower-work: http://pastebin.ca/1800349 |
| 18:54.33 | b14ck | sup everyone |
| 18:55.27 | [TK]D-Fender | mmattice: [Feb 17 09:36:41] WARNING[4067]: chan_dahdi.c:3360 pri_find_dchan: No D-channels available! Using Primary channel 24 as D-channel anyway! <-- you shouldn't be getting this.. you have an **E1*** |
| 18:55.38 | [TK]D-Fender | mmattice: That says your D should be on 16, not 24 |
| 18:56.15 | [TK]D-Fender | mmattice: I'm thinking you may not have set the jumper right for your card |
| 18:56.35 | tzafrir_laptop | or the parameter t1e1override |
| 18:57.32 | mmattice | it's worked in the past reasonably reliably, we just had to shelve this project for a while |
| 18:58.21 | *** join/#asterisk oej (~olle@109.58.19.225.bredband.tre.se) |
| 18:58.23 | mmattice | and it's passing dchannel traffic fine for the first call after we cycle the circuit |
| 18:59.31 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 19:00.24 | ManxPower-work | mmattice: What type of PRI? T-1 or E-1? |
| 19:00.28 | mmattice | T1 |
| 19:00.38 | mmattice | I'm in the US. E1's are hard to come by |
| 19:02.05 | mmattice | dchan is set to 24 in my dahdi/system.conf. span=1,1,1,esf,b8zs bchan=1-6 |
| 19:03.18 | [TK]D-Fender | mmNix that.. read off other host info and go your case mixed up |
| 19:03.56 | mmattice | huh? |
| 19:04.59 | [TK]D-Fender | mmattice: Nevermidn... bad aim on my part... |
| 19:05.06 | *** part/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 19:10.01 | *** join/#asterisk basty (~basty@212.218.65.136) |
| 19:10.03 | basty | Hi |
| 19:10.14 | mmattice | dahdi_scan will show what it's jumpered as, right? |
| 19:10.49 | basty | I have something weirdo going on. I updated from 1.2. to 1.4.29..so far so good..but now I just called somebody - he put me on "hold" but of some reasons I did not hear HIS musiconhold music....I have heard my own...what could be the matter ? |
| 19:12.27 | bmoraca_work | mmattice: are you getting your PRI from annother piece of hardware, such as an Adtran or a Cisco 2800? |
| 19:12.52 | mmattice | it's coming from a fiber mux box provided by the telco. |
| 19:12.56 | *** join/#asterisk ruben23 (~AGENT@122.55.48.243) |
| 19:13.04 | bmoraca_work | hrm |
| 19:13.21 | bmoraca_work | could be encapsulated all the way back to them or terminated right there, then |
| 19:14.13 | bmoraca_work | does the d channel flap like that with no load and with outbound calls, or just on inbound calls? |
| 19:14.22 | bmoraca_work | and is it a new symptom? |
| 19:14.59 | mmattice | we don't make outbound calls on it so I don't know about that. |
| 19:15.58 | mmattice | the issue is new, yes. this box has been sitting a while as we've had to prioritize other projects ahead of finishing it, but we had a contractor out tinkering with dialplan stuff trying to get our installation of RT integrated in and we didn't notice the problem back then |
| 19:16.16 | fifer | I'm having an issue getting an Aastra 480i to re register to * |
| 19:16.58 | fifer | It was working fine yesterday then I had to redo the extensions on this test * server and after changing the sip extension on the 480i it will not register to the new extension |
| 19:17.26 | mmattice | a call into it I made a minute or so ago seems to be looping through the dialplan even though I got a fast busy instead of an answer |
| 19:17.33 | fifer | I have tested the extension with a couple of soft phones but no amont of re-checking the 480i config and repooting it is making a diference |
| 19:17.36 | ruben23 | hi how do i setup my did number to worked on my asterisk server, i have the did number and username, password of it.. |
| 19:17.51 | bmoraca_work | mmattice: pastebin that call |
| 19:17.58 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 19:18.16 | mmattice | only thing of note is 'Channel 0/6, span 1 got hangup ACK' |
| 19:18.26 | fifer | ruben23: Is this just a strait * system or AsteriskNow, or other with FreePBX? |
| 19:19.36 | ruben23 | fifer: its a scratch asterisk install |
| 19:20.22 | fifer | ruben23: This is a new sip trunk with a DID? |
| 19:20.57 | ruben23 | <PROTECTED> |
| 19:21.11 | fifer | Have you set up the trunking yet? |
| 19:21.19 | fifer | What provider? |
| 19:24.26 | mmattice | after a dahdi restart, I made two calls into a more simple number that just plays a disconnected file http://pastebin.ca/1800386 |
| 19:24.35 | mmattice | first call works, second doesn't. |
| 19:25.23 | mmattice | next call I get the 'got hangup ACK' instead of the dchannel going down |
| 19:29.08 | ruben23 | <PROTECTED> |
| 19:29.24 | *** join/#asterisk Akiraa (~Akiraaaa@79.112.38.195) |
| 19:29.24 | ruben23 | probider is delridgeis |
| 19:31.29 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 19:33.46 | *** join/#asterisk Dandan (dandan@ip72-221-92-215.ri.ri.cox.net) |
| 19:33.54 | Dandan | hello :) |
| 19:33.57 | Dandan | Kobaz: ayt? |
| 19:34.09 | bmoraca_work | mmattice: i'd call your provider and make sure there's not something screwy at their end |
| 19:34.31 | benngard | guys, can this be right? if u not enter "Display Name:" in a spa2102, just "User ID:" and "Password:" let it register and call the spa, * coredump... |
| 19:34.49 | Dandan | hm, I am signing up for voicepulse, first msg with referral code gets the referral from me... |
| 19:36.37 | Dandan | gdb that coredump, see what causes it to core dump |
| 19:37.31 | *** join/#asterisk Alagar (~Administr@122.164.34.239) |
| 19:42.00 | p3nguin_ | dandan: You're going to use them for business phone service? |
| 19:42.19 | Kobaz | mmm |
| 19:42.23 | Dandan | nah, for personal for now, if it works OK, then I will consider it |
| 19:42.39 | p3nguin_ | dandan: Flowroute and VoIP.ms are probably cheaper. |
| 19:43.05 | Katty | I"M IN THE MOOD FOR LOVE |
| 19:43.09 | Katty | SIMPLY BECAUSE YOU"RE NEAR ME |
| 19:43.10 | Kobaz | it appears that Hangup() from within a 'h' exten behaves differently when the hangup handler is called during processing agi, or during processing regular dialplan |
| 19:43.24 | *** join/#asterisk gelpg (~chatzilla@92-249-159-220.pool.digikabel.hu) |
| 19:43.31 | Kobaz | Dandan: kobazer |
| 19:43.39 | Katty | sways |
| 19:43.40 | p3nguin_ | Why would you need to Hangup() from within the 'h' exten? |
| 19:43.48 | Dandan | p3nguin_: will try them as well |
| 19:43.54 | Kobaz | p3nguin_: to exit from the hangup handler |
| 19:44.07 | p3nguin_ | Doesn't it do that by default? |
| 19:44.08 | *** join/#asterisk Dibri (~gavit@pop1.isgroup.sr) |
| 19:44.17 | Kobaz | what if you want to exit early |
| 19:44.18 | *** join/#asterisk t_j (~tj@tomjudge.vm.bytemark.co.uk) |
| 19:44.31 | p3nguin_ | You run a bunch of stuff when 'h' is called? |
| 19:44.34 | Kobaz | yeah |
| 19:44.36 | t_j | any polycom users in here? I have a Q about provisioning and config files |
| 19:44.47 | Katty | there are many polycom users in here. |
| 19:44.48 | Naikrovek | t_j: lots of us. ask away |
| 19:44.49 | p3nguin_ | I see. That sorta makes more sense. |
| 19:45.02 | Kobaz | if you run Hangup() from within 'h', and you're processing dialplan... it's fine |
| 19:45.02 | Dandan | Katty is not one of them, her * broke yesterday... |
| 19:45.03 | Katty | FUNNY BUT WHEN YOU"RE NEAR ME |
| 19:45.06 | Kobaz | the call ends, and you're done |
| 19:45.07 | Katty | I"M IN THE MOOD FOR LOVE |
| 19:45.12 | Katty | Dandan: it did? |
| 19:45.17 | Katty | Dandan: oh noes |
| 19:45.27 | p3nguin_ | kobaz: I was thinking you did exten => h,1,Hangup() ... which seemed silly. |
| 19:45.27 | Kobaz | if you run AGI().... and someone hangs up during the AGI... and now it hits 'h'... and you call Hangup().... the 'h' runs a second time |
| 19:45.29 | *** join/#asterisk geneticx (~chatzilla@host-208-88-126-198.biznesshosting.net) |
| 19:45.32 | Dandan | want me to grep logs? :) |
| 19:45.32 | t_j | Does the MAC-phone1.cfg file have to be completed or can I just included the snipits I need? |
| 19:45.43 | Katty | Dandan: baroo? |
| 19:45.48 | Katty | Dandan: what are you talking about |
| 19:46.02 | Kobaz | either agi or fastagi |
| 19:46.13 | Naikrovek | t_j: just include the parts you want to change from the file that comes in the firmware, then include the file that comes in the firmware in the list of config files |
| 19:46.13 | p3nguin_ | katty: How dare you pretend like your * didn't break! |
| 19:46.23 | Katty | I KNOW |
| 19:46.25 | Katty | how dare i |
| 19:46.26 | Dandan | heh :) |
| 19:46.30 | Katty | how did it break again? |
| 19:46.36 | Dandan | it did, didn't it? |
| 19:46.48 | Katty | things don't magically break |
| 19:46.49 | Dandan | I do not know, pebkac? |
| 19:46.53 | Katty | usually. |
| 19:47.06 | Katty | unless it is hardware failure |
| 19:47.13 | Katty | or LIGHTENING |
| 19:47.17 | Katty | dundundun |
| 19:47.20 | p3nguin_ | Even then, it's magical hardware failure. |
| 19:47.20 | t_j | Naikrovek: so CONFIG_FILES="SIP-3.2.2/sip.cfg, SIP-3.2.2/phone1.cfg, MAC-phone1.cfg" ? |
| 19:47.26 | Katty | yesh. |
| 19:47.30 | Naikrovek | t_j: i use something like: CONFIG_FILES="systemPrefs.cfg, server.cfg, 000412345678-phone1.cfg, phone1.cfg, sip.cfg" |
| 19:47.35 | Naikrovek | t_j: backwards |
| 19:47.50 | geneticx | Hello you all. I need some assistance, I have two PBXs connected through a VPN tunnel that was working just fine, yesterday I did some small changes to the VPN router and now my asterisk peers show as unreachable even though I can ping the host.. anyone has clue? |
| 19:48.05 | Kobaz | p3nguin_: wiggity, eh? |
| 19:48.11 | Katty | geneticx: err well, that is obvious. |
| 19:48.13 | Naikrovek | t_j: include customized BEFORE standard. first mentioned gets priority, so to override a setting in sip.cfg with a file sip2.cfg do it like this "sip2.cfg, sip.cfg" |
| 19:48.19 | *** join/#asterisk catojo (~catojo@189.24.85.1) |
| 19:48.20 | Katty | geneticx: it was working /before/ you made router changes. |
| 19:48.29 | Katty | geneticx: it wasn't workign /after/ you made router changes. |
| 19:48.32 | bmoraca_work | geneticx: what changes did you make to the routers? what kind of routers are they? |
| 19:48.34 | Katty | geneticx: that should tell you something |
| 19:48.39 | t_j | Naikrovek: interesting, it sys processes in order listed in the comments... |
| 19:48.48 | Dandan | what "small" changes are you talking about? |
| 19:49.02 | Katty | probably disconnected the vpn tunnel ;P |
| 19:49.09 | Katty | that's a small change, right? ;P |
| 19:49.15 | ManxPower-work | t_j: Polycom has extensive documentation about provisioning. Read them. |
| 19:49.20 | Naikrovek | yes |
| 19:49.24 | *** join/#asterisk Victor_Yure (~victor@unaffiliated/victoryure/x-837844) |
| 19:49.34 | t_j | Naikrovek: so I can just have a <phone1><reg...></phone1> block and nothing else ... |
| 19:49.48 | Katty | t_j: take a look. |
| 19:49.51 | Katty | t_j: it's in a book. |
| 19:49.51 | t_j | ManxPower-work: i have, they are confusing to new users... |
| 19:49.56 | Katty | t_j: reeeeeeeadddinggg rainbowwww |
| 19:50.04 | Katty | t_j: when in doubt, open a ticket with polycom |
| 19:50.05 | *** part/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 19:50.10 | Katty | t_j: DO EET |
| 19:50.16 | t_j | ManxPower-work: i have the white paper next to me |
| 19:50.19 | geneticx | Katty: I know it's obvious, but the changes I made were not meant to break the connection, all I did was add a new crypto map for a new VPN connection |
| 19:50.20 | Naikrovek | ticket? |
| 19:50.31 | Katty | Naikrovek: ticket. |
| 19:50.34 | Katty | Naikrovek: you know. |
| 19:50.40 | Katty | Naikrovek: click it or ticket |
| 19:50.44 | Naikrovek | seems like overkill |
| 19:50.48 | Katty | it works. |
| 19:50.51 | geneticx | bmoraca_work: it's a cisco 2811, just added a new VPN connection |
| 19:50.51 | Naikrovek | don't mention click it or ticket to me |
| 19:50.57 | ManxPower-work | t_j: also have the admin guide in front of you |
| 19:50.58 | Katty | CLICK EET OR TICK EET |
| 19:51.12 | Katty | i sure am chipper this afternoon |
| 19:51.13 | geneticx | bmoraca_work: to a remote location. |
| 19:51.13 | Naikrovek | why in the HELL do we waste money on "click it or ticket" campaigns when EVERY car has an extremely annoying sound designed in BY LAW to remind you |
| 19:51.19 | *** part/#asterisk garymc (~chatzilla@host86-164-36-128.range86-164.btcentralplus.com) |
| 19:51.30 | Katty | Naikrovek: because people don't care. |
| 19:51.39 | Katty | Naikrovek: which results in death |
| 19:51.40 | t_j | ManxPower-work: in the same folder :) |
| 19:51.41 | bmoraca_work | geneticx: which location is having trouble, then? did you make sure to add the new subnet to your localnet settings in sip.conf? |
| 19:51.47 | Katty | Naikrovek: i'm okay with them taking my tax money to save a few lives. |
| 19:51.50 | ManxPower-work | Naikrovek: because a significant number of people still don't use them. |
| 19:52.06 | Kobaz | Dandan: same with me... i ported into vonage from verizon... so i could port it back out |
| 19:52.07 | Naikrovek | well if their annoying beeping in the car doesn't remind them, a roadsign or commercial isnt' going to either |
| 19:52.10 | Naikrovek | it's wasted money |
| 19:52.16 | Katty | how do you know |
| 19:52.27 | Katty | you're not the average cookie. |
| 19:52.42 | Katty | you don't know how their brains work. |
| 19:52.59 | Qwell | Katty: poorly |
| 19:54.39 | p3nguin_ | I believe in natural selection. |
| 19:54.50 | geneticx | bmoraca_work: I have a total of 2 VPNs right now, and there are 2 asterisk peers that connect through it. Yesterday I added a third VPN, without a problem. But I lost my peers in asterisk, although I can ping the PBXs just fine it shows as unreachable for some reason |
| 19:55.12 | ManxPower-work | sounds to me like something is NATTING |
| 19:56.18 | Katty | p3nguin_: and if your appendix went out, would you not go to the hospital? |
| 19:56.24 | Katty | p3nguin_: because you believe in natural selection |
| 19:56.25 | bmoraca_work | geneticx: you're throwing too many pronouns around for me to be able to get any kind of idea about your topology. specifics would be nice. also, debugs and configs would be super, too |
| 19:56.29 | Katty | p3nguin_: you'd just let yourself die? |
| 19:56.40 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 19:56.49 | *** join/#asterisk basty (~basty@212.218.65.136) |
| 19:56.49 | Dandan | die p3nguin_ die! |
| 19:56.51 | basty | hi again |
| 19:58.04 | basty | I have another strange problem. (Asterisk 1.4.29 and dahdi 2.2.1) As soon as I try to call (from external) a phone that has a call forwarding to a local number, the call is open - but I dont hear anything (http://pastebin.com/d5887f920). Could somebody please help me ? Thanks ! |
| 19:58.58 | came0 | hey where can I get a list of calls in my parking lot? |
| 19:59.04 | came0 | programaticly |
| 19:59.09 | came0 | ie the rawman or cli |
| 19:59.12 | Naikrovek | Katty: if some dude STILL doesn't buckle their safety belt when they drive, they DESERVE to be flung through a windshield and die bleeding on a road |
| 19:59.15 | came0 | i'd prefer a rawman command |
| 19:59.35 | ManxPower-work | Naikrovek: or be in a coma for 10 years, costing the state millions of dollars. |
| 19:59.39 | Katty | basty: that doesn't look like a full phone call to me |
| 20:00.28 | basty | Katty: mhh...I just called the number - phone 100 (snom phone) has a call forwarding to "5". And the "5" is actually a Playback |
| 20:00.30 | Katty | basty: setup a test extension to play that audio file so you can listen to it directly |
| 20:00.51 | Naikrovek | ManxPower-work: that would cost less than putting up a single sign on the free way, a giant dot matrix sign, that can display anything, but only ever displays "click it or ticket" |
| 20:01.20 | ManxPower-work | Sometimes they say "Zombies head!" |
| 20:01.23 | ManxPower-work | and ahead too |
| 20:01.35 | Naikrovek | not those |
| 20:01.45 | Naikrovek | the giant ones that arch over the roads |
| 20:01.47 | Naikrovek | HUGE ones |
| 20:01.53 | basty | Katty: Well - if I call the Playback directly (local or with an external number) it works...but as soon somebody puts a call forward - it doesnt work....its not just this extension...as soon as sombody puts a normal call forward to an other extension...the caller doesnt even hear a "ringing" |
| 20:01.54 | Naikrovek | lots of those around here, always blank |
| 20:01.59 | Naikrovek | or "click it or ticket" |
| 20:02.02 | Naikrovek | useless wastes of money |
| 20:02.13 | Naikrovek | put some sports scores on there or something chripes |
| 20:02.32 | Naikrovek | or weather forecasts |
| 20:02.38 | Naikrovek | or news headlines |
| 20:02.41 | bmoraca_work | Naikrovek: the ones i get the most peeved about are the ones for the Census. they advertise in primetime with ridiculous commercials that don't make any sense. |
| 20:02.49 | Naikrovek | bmoraca_work: yes |
| 20:03.11 | Katty | basty: please post a COMPLETE phone call along with sip debug |
| 20:03.25 | p3nguin_ | katty: I think you misinterpreted what I said. If the people are so friggin' stupid or lazy that they can't put on their seatbelts, and then happen to crash and die as a direct result of not wearing said seatbelt(s), that's okay by me. |
| 20:03.26 | Katty | basty: 10 verbosity will also be good |
| 20:03.33 | basty | okay sec |
| 20:03.47 | Katty | p3nguin_: i just want to save lives. |
| 20:03.47 | *** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no) |
| 20:03.51 | Katty | p3nguin_: is that so bad? |
| 20:03.51 | p3nguin_ | katty: If they did wear the seatbelts and died anyway, that's a tragedy. |
| 20:03.59 | Katty | p3nguin_: any death is a tragedy |
| 20:04.13 | p3nguin_ | katty: Ah, you don't know about Darwin awards? |
| 20:04.25 | Katty | p3nguin_: i don't care about darwin awards |
| 20:04.30 | Katty | p3nguin_: i care about people living (= |
| 20:04.34 | Dandan | p3nguin_: as long as they do not take any innocent motorists with them, it is also OK with me |
| 20:04.36 | Katty | p3nguin_: living to see their children another day |
| 20:04.45 | p3nguin_ | katty: Those cases are not tragic. |
| 20:04.47 | Katty | p3nguin_: and if a billboard just happens to make it happen, then so be it |
| 20:04.56 | bmoraca_work | p3nguin_: social darwinsim is very frowned upon now-adays, and personal responsibility is completely taboo. |
| 20:05.09 | Dandan | nods |
| 20:05.55 | basty | Katty: http://pastebin.com/d4575a664 |
| 20:06.00 | p3nguin_ | bmoraca_work: Responsibility and accountability are virtually nonexistent nowadays. |
| 20:06.40 | Katty | basty: that's only part of what i asked for |
| 20:06.54 | Katty | basty: if you need help, please list everything we need to help you |
| 20:07.04 | bmoraca_work | there is no personal responsibility anymore...you're upside down on your house and can't meet your balloon interest payments, it's not your fault, it's the banks! you smoked for 20 years and have emphasema? cigarette companies' fault. |
| 20:07.06 | basty | Katty: thats the whole output with debug / verbose 10 |
| 20:07.12 | p3nguin_ | I still believe a person should held accountable for his or her actions, even if no one else does. |
| 20:07.16 | bmoraca_work | it goes on and on and on |
| 20:07.27 | p3nguin_ | And if death is the direct result of those actions, so be it. |
| 20:07.55 | p3nguin_ | Accidents happen. Those are exceptions. |
| 20:08.01 | Katty | i guess i just value life more than you do :P |
| 20:08.32 | p3nguin_ | Accidents due to the lack of sense... that's a different story, because it's technically not an accident if you were negligent. |
| 20:08.57 | bmoraca_work | Katty: you can't protect people from themselves |
| 20:09.02 | Katty | i can try :P |
| 20:09.14 | p3nguin_ | I do not value the lives of morons. Morons reproduce and then we have more morons. |
| 20:09.15 | Katty | confinscates all spoons. |
| 20:09.27 | Dandan | no spooning! |
| 20:09.31 | Katty | :<<< |
| 20:09.33 | Katty | Dandan: GET OUT |
| 20:09.42 | Dandan | make me! :P |
| 20:09.47 | Katty | meh, too much effort |
| 20:09.52 | Dandan | pfffft |
| 20:09.54 | Dandan | :> |
| 20:09.59 | Katty | don't you pffft me |
| 20:10.06 | Katty | i have a spoon and i know how to use it! |
| 20:10.08 | beek | FIGHT! FIGHT! |
| 20:10.23 | Katty | why fight when we can hurl insults? |
| 20:10.57 | Dandan | heh, show me! |
| 20:10.58 | Dandan | :) |
| 20:12.48 | Dandan | do not chicken out now! |
| 20:12.52 | Dandan | show me how you use it! |
| 20:14.58 | Dandan | http://www.to.com.pl/apps/pbcs.dll/article?AID=/20100217/KRAJ_SWIAT/810571916 |
| 20:15.00 | Dandan | taaaak :) |
| 20:16.45 | bmoraca_work | interesting website: http://pleaserobme.com/ |
| 20:16.54 | Dandan | argh :/ |
| 20:17.01 | Dandan | wrong window, sry :) |
| 20:17.10 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 20:18.29 | p3nguin_ | bmoraca_work: People don't get it. |
| 20:19.21 | p3nguin_ | They make things so easy already, but now someone was nice enough to combine all that info into one nice convenient list! |
| 20:20.28 | [TK]D-Fender | [15:10]<Katty>why fight when we can hurl insults? <- CORRECT. civilization was born the first time someone threw an insult instead of a rock. |
| 20:21.32 | beek | Excellent statement. |
| 20:24.16 | *** join/#asterisk Dibri (~gavit@190.98.32.116) |
| 20:29.00 | Kobaz | all i do is report bugs |
| 20:29.21 | Kobaz | i need to hire someone, whose sole purpose is to fill out asterisk bug reports |
| 20:30.00 | ruben23 | hi |
| 20:30.23 | ruben23 | for DID registration i need to set it on the sip.conf right..? |
| 20:31.06 | ruben23 | how would it be look lije |
| 20:31.09 | ruben23 | like |
| 20:32.49 | Kobaz | ~book |
| 20:32.50 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 20:33.14 | Kobaz | ruben23: it's free online... for the pdf... it should answer many questions you have about sip |
| 20:34.53 | p3nguin_ | ruben23: The DID's route is handled by your ITSP. They have to know where you are so they can send the calls to you. That can be handled statically on their web site or via register statements in sip.conf or iax.conf. |
| 20:36.31 | p3nguin_ | ruben23: http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/asterisk-CHP-8-SECT-6.html#asterisk-CHP-8-SECT-6.2 |
| 20:38.31 | *** join/#asterisk roe (~roe___@216-164-215-125.c3-0.eas-ubr15.atw-eas.pa.cable.rcn.com) |
| 20:39.03 | roe | anyone else experiencing a voicepulse outage? I can't confirm anything as their website is down and I don't have a contact number |
| 20:40.33 | p3nguin_ | roe: (732) 339-5100 |
| 20:40.41 | roe | nice thanks! |
| 20:41.06 | ManxPower-work | Kobaz: Filing them is not the problem. Defending them is the problem. |
| 20:41.21 | p3nguin_ | If your phones are down and you don't have a secondary phone, I could probably give them a call for you. |
| 20:41.43 | roe | ha! their phone system seems down as well, not a good day to be an employee of voicepulse |
| 20:42.30 | p3nguin_ | The system answered when I called. |
| 20:43.37 | *** join/#asterisk d-k-t (~D@112.202.232.46) |
| 20:43.37 | roe | now it's ringing, before it was just 'all circuits busy', which I guess means lots of people are calling and complaining |
| 20:43.45 | p3nguin_ | I am caller number 6 and there is 8 minutes wait time. |
| 20:43.58 | ManxPower-work | roe: where are you calling from? |
| 20:44.03 | roe | my cell |
| 20:44.17 | p3nguin_ | I disconnected, so I'm not going to take up a queue slot. |
| 20:44.35 | roe | ouch, I'm 14 with estimaged wait time of 20mins |
| 20:44.41 | p3nguin_ | oops |
| 20:44.44 | roe | estimated* |
| 20:44.47 | p3nguin_ | I should have stayed on. |
| 20:44.56 | bmoraca_work | estimagicked |
| 20:45.00 | p3nguin_ | and transfered to you |
| 20:45.03 | roe | that what they make speaker phone for |
| 20:45.28 | p3nguin_ | I usually either do speaker and mute, or headset. |
| 20:45.33 | roe | yup |
| 20:46.11 | p3nguin_ | I'm calling back to see where it puts me. |
| 20:46.37 | *** join/#asterisk wvds_nl (~wvds-nl@230-237-045-062.dynamic.caiway.nl) |
| 20:47.06 | p3nguin_ | number 7, 8 minutes |
| 20:47.12 | *** part/#asterisk wvds_nl (~wvds-nl@230-237-045-062.dynamic.caiway.nl) |
| 20:47.30 | roe | I'm all about the fail today I guess |
| 20:47.39 | p3nguin_ | haha |
| 20:47.58 | p3nguin_ | Did you press option 3, then 4? |
| 20:48.09 | roe | I'm just now number 8, yup. |
| 20:48.58 | roe | thanks for your help. |
| 20:49.15 | p3nguin_ | No problem. I hope they can answer your question and get you back online. |
| 20:50.05 | *** join/#asterisk radcliff (~radcliff@h-63-22.A259.priv.bahnhof.se) |
| 20:56.17 | Katty | ruhroe |
| 20:56.47 | Katty | infobot: ruh roh? |
| 20:59.11 | Katty | imagebin is neat. |
| 20:59.16 | Katty | i like to see what people have posted recent |
| 20:59.17 | Katty | ly |
| 21:02.05 | Katty | http://imagebin.org/85209 <- someone's been doing algebra |
| 21:02.12 | Katty | very basic algebra |
| 21:03.30 | Gido-E | katty you know sage (math tool)? |
| 21:04.20 | Katty | can't say i do |
| 21:04.48 | Katty | why do you need a math tool for algebra? |
| 21:05.10 | Gido-E | :-) ok |
| 21:05.19 | rubberneck | Is the timeout arg of the Dial() application in seconds or milliseconds? |
| 21:05.19 | Kobaz | when executing a hangup handler ('h' exten) during an AGI script... how would I tell which side of the call hung up.... when you're not using AGI... ${CHANNEL} contains the channel hung up... but when using AGI... it's always the caller channel |
| 21:05.33 | rubberneck | core show app dial doesnt specify. |
| 21:05.56 | Kobaz | seconds |
| 21:05.57 | Gido-E | katty, first do some "basic" training at www.projecteuler.net |
| 21:06.08 | Katty | ^_- |
| 21:06.11 | Katty | why? |
| 21:06.14 | Kobaz | everything tends to be in seconds, unless otherwise specified |
| 21:06.18 | Gido-E | do you know that site? |
| 21:06.34 | Katty | what does that site have to do with me? |
| 21:06.57 | Gido-E | you where talking about math thingies |
| 21:07.00 | Katty | i passed my physics final |
| 21:07.04 | Katty | i think i'm good on math |
| 21:07.10 | Katty | oh |
| 21:07.17 | Katty | no i was just linking an image from imagebin |
| 21:07.33 | Katty | it was interesting to see someone posting pages from an alegbra book |
| 21:08.23 | [TK]D-Fender | Kobaz: Dial with "g" and you'll know. |
| 21:08.26 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 21:08.55 | Kobaz | [TK]D-Fender: hmm, good point |
| 21:09.26 | Katty | mmm |
| 21:09.29 | Katty | quantum mechanics |
| 21:09.31 | Katty | <3 |
| 21:10.01 | Katty | teh kitteh is quantumly entangled |
| 21:10.04 | Katty | and not in a ball of string. |
| 21:10.24 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 21:10.36 | *** join/#asterisk tech_travis (~tech_trav@adsl-66-126-9-83.dsl.lsan03.pacbell.net) |
| 21:10.39 | *** join/#asterisk Blackgibson (~shawn@S01060015e912f559.vs.shawcable.net) |
| 21:10.40 | Kobaz | [TK]D-Fender: what's weird... is if you don't use AGI... the hangup handler will get called with a zombie channel if the callee hangs up... and I've seen this consistantly |
| 21:10.41 | rubberneck | anyone know if the timeout arg of the Dial() application in seconds or milliseconds? |
| 21:11.08 | Katty | seconds. |
| 21:11.15 | Katty | and 1 ring is about 4 seconds. |
| 21:11.19 | *** join/#asterisk tech_travis (~tech_trav@adsl-66-126-9-83.dsl.lsan03.pacbell.net) |
| 21:11.27 | rubberneck | thanks |
| 21:11.30 | Katty | mhmm |
| 21:12.04 | Katty | i wish they had a quantum physics course out here :< |
| 21:14.00 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 21:15.29 | carrar | They don't teach that in Elementry school out there? |
| 21:15.33 | carrar | WTF |
| 21:15.42 | ManxPower-work | Is that the class that is always yestermorrow? |
| 21:16.00 | ManxPower-work | always meets yestermorrow |
| 21:18.22 | ManxPower-work | carrar: she's in the USA, not the rest of the world. |
| 21:18.41 | ManxPower-work | Katty: standard USA ring + silence is 6 seconds. |
| 21:19.02 | ManxPower-work | I think, as you said 4 seconds of ring, plus 2 seconds of silence. |
| 21:20.05 | carrar | haha |
| 21:20.11 | p3nguin_ | That's backward. |
| 21:20.21 | p3nguin_ | 2 seconds ring, 4 seconds silence. |
| 21:20.25 | Kobaz | heh, yeah |
| 21:20.47 | ManxPower-work | I gues my lisdexia kicked in |
| 21:21.59 | carrar | it kicked in tomorrow in the past |
| 21:22.10 | p3nguin_ | That's right now! |
| 21:22.17 | p3nguin_ | Trickster! |
| 21:22.30 | *** join/#asterisk Time (~m-b@74.94.48.177) |
| 21:23.32 | [TK]D-Fender | checkout time, later all |
| 21:24.07 | *** join/#asterisk ipkaf (~chatzilla@bgl93-7-88-189-218-150.fbx.proxad.net) |
| 21:24.23 | qsrfh | hello |
| 21:24.25 | qsrfh | there |
| 21:24.29 | qsrfh | is there anyone here ? |
| 21:24.51 | Qwell | ~ask |
| 21:24.51 | 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. |
| 21:25.16 | qsrfh | i need asterisk for openwrt where can i find it ? |
| 21:25.57 | Kobaz | questions must be in the form of an answer |
| 21:27.29 | *** join/#asterisk raden_work (~jon@69-179-99-17.stat.centurytel.net) |
| 21:30.57 | Blackgibson | <PROTECTED> |
| 21:31.27 | *** part/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 21:31.30 | qsrfh | well i just install asterisk |
| 21:31.33 | Kobaz | Blackgibson: #freepbx, #trixbox |
| 21:31.38 | qsrfh | and i m freepbx user |
| 21:31.53 | Kobaz | ~freepbx |
| 21:31.54 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 21:31.58 | qsrfh | now i want to know how to creat sip extension ? |
| 21:32.03 | qsrfh | on asterisk |
| 21:32.06 | Blackgibson | Kobaz: Tried already.. asked here last |
| 21:32.06 | qsrfh | now i want to know how to creat sip extension ? |
| 21:32.17 | Blackgibson | ah |
| 21:32.20 | qsrfh | now i want to know how to creat sip extension with asterisk now ? |
| 21:32.28 | Kobaz | qsrfh: settle doen |
| 21:32.28 | Blackgibson | thanks anyways Kobaz |
| 21:32.34 | Kobaz | down |
| 21:32.46 | Kobaz | Blackgibson: yeah, i dunno... we dont really know freepbx/etc here |
| 21:33.05 | qsrfh | no i m not in seattlle |
| 21:33.11 | qsrfh | Kobaz: |
| 21:33.35 | Kobaz | Blackgibson: it's like going to a briggs and straton engine dealer, asking them why your car wont start... they just make the engine |
| 21:33.57 | *** join/#asterisk aceking5 (~aceking5@71-94-132-102.static.mtpk.ca.charter.com) |
| 21:34.46 | qsrfh | well how to create sip extension with asterisk ? |
| 21:34.51 | *** join/#asterisk grey-monkey (~ericshell@75-148-103-190-Utah.hfc.comcastbusiness.net) |
| 21:34.52 | Kobaz | you just asked that |
| 21:34.55 | Kobaz | read the book |
| 21:35.10 | Kobaz | look at the examples that come with asterisk... sip.conf |
| 21:35.15 | Kobaz | ~book |
| 21:35.16 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 21:35.34 | raden_work | Naikrovek, hola |
| 21:35.50 | qsrfh | i understand nothing on this book |
| 21:35.54 | Naikrovek | heya |
| 21:35.59 | qsrfh | very complicate |
| 21:36.09 | raden_work | qsrfh, did u read it |
| 21:36.21 | qsrfh | i buy it for 53⬠|
| 21:36.38 | qsrfh | if you are the author of this book |
| 21:36.57 | qsrfh | can you return back my money ? |
| 21:37.01 | Kobaz | it's free online |
| 21:37.21 | Kobaz | qsrfh: if you don't understand the book, how do you expect to configure asterisk |
| 21:37.22 | raden_work | qsrfh, thousands of people have setup asterisk off that book including mysql. |
| 21:37.33 | raden_work | myself wow been working with mysql was to much LOL |
| 21:37.34 | Kobaz | mysql? |
| 21:37.37 | Kobaz | heh |
| 21:37.53 | kaldemar | raden_work: that happens when you deal with databases too much |
| 21:38.00 | raden_work | Kobaz, been up 34 hours time for a break LOL :) |
| 21:38.06 | Kobaz | it happens when you deal with crappy databases too much |
| 21:38.16 | qsrfh | me i buy that book for 53⬠in france |
| 21:38.21 | raden_work | im going to try postgres one of these days |
| 21:38.32 | raden_work | qsrfh, whats the problem with the book ? |
| 21:38.33 | qsrfh | in year 2008 |
| 21:38.50 | qsrfh | and since nothing understand on it |
| 21:38.56 | Kobaz | qsrfh: i'm sorry it hasn't helped you... but asterisk is a very advanced platform, and you need a certain minimum level of computer and systems knowledge to be able to work with it |
| 21:39.00 | raden_work | NOTICE: This is not #Customer-Service |
| 21:39.21 | Kobaz | qsrfh: if the book hasn't helped you... we can't help you either |
| 21:39.28 | qsrfh | i don't know if it is or not |
| 21:39.42 | qsrfh | ppl recommand that book |
| 21:39.47 | Kobaz | it's a good book |
| 21:39.50 | qsrfh | ppl here recommand that book |
| 21:39.52 | Naikrovek | because it's a good book |
| 21:40.09 | Kobaz | qsrfh: you need to learn to walk before you learn to run |
| 21:40.11 | qsrfh | so i have to share my point of view |
| 21:40.13 | raden_work | I so need to get my pdf done for setting up simple asterisk |
| 21:40.14 | qsrfh | that's allll |
| 21:40.17 | Kobaz | qsrfh: get youself a linux book |
| 21:40.24 | ruben23 | hi how to show registration on asterisk console |
| 21:40.30 | raden_work | qsrfh, Running Linux |
| 21:40.37 | Kobaz | ruben23: sip show registrations |
| 21:40.42 | raden_work | ruben23, sip registration ? |
| 21:41.05 | kaldemar | ruben23: sip show registry |
| 21:41.13 | qsrfh | what are talking about walking b4 runing ? |
| 21:41.35 | Kobaz | qsrfh: compare asterisk to working on a car.... you buy the manual for your car, and if you don't understand what's in the manual... you sure as hell aren't going to be able to work on the car |
| 21:41.38 | qsrfh | and do you know that i know or not about linux ? |
| 21:41.52 | Naikrovek | qsrfh: they're assuming that you don't know linux |
| 21:42.01 | raden_work | anyone know what hotkey in kde turns my freaking numpad back into a numpad :( |
| 21:42.08 | raden_work | numlock on still moves mouse :( |
| 21:42.13 | Kobaz | kaldemar: ah yeah it's one of those |
| 21:42.13 | Naikrovek | qsrfh: correctly or not, that's what they're assuming |
| 21:42.19 | Naikrovek | raden_work: yay kde |
| 21:42.32 | kaldemar | Kobaz: it's the one i said. :) |
| 21:42.47 | raden_work | qsrfh, what are you trying todo ? |
| 21:43.02 | Kobaz | he's trying to set up a sip peer, but can't figure it out from the book |
| 21:43.14 | Kobaz | qsrfh: here's the thing |
| 21:43.36 | qsrfh | a basic setup for gsm voip gateway modem raden_work |
| 21:43.58 | Kobaz | qsrfh: we can walk you through every single step you need, and hold you hand every inch of the way... but then you still wont be able to do anything else with asterisk, unless you have a certain minimum level of system knowledge |
| 21:44.24 | *** join/#asterisk smooth_penguin (~smoove@59.95.26.96) |
| 21:44.37 | Kobaz | qsrfh: and the fact that you didn't understand the book, is showing a lack of that very required knowledge |
| 21:44.48 | Kobaz | qsrfh: i strongly recommend something other than asterisk for you |
| 21:44.57 | qsrfh | ahaha |
| 21:45.02 | Kobaz | qsrfh: download 3cx for windows |
| 21:45.22 | raden_work | qsrfh, if you can wait 2 weeks ill have a howto done i can mail you |
| 21:45.27 | qsrfh | tell me why i understand better with trixbox without tears |
| 21:45.37 | qsrfh | about all function on freepbx |
| 21:45.49 | raden_work | cause they have GUI |
| 21:45.51 | qsrfh | and with you ur book i understand nothing |
| 21:45.51 | kaldemar | qsrfh: try to do it yourself first with the examples. when you run into a problem, come here to show what you've done so far and show the problem. that's the best way to get help. |
| 21:45.51 | ManxPower-work | Asterisk isn't a phone system/PBX. It is really a toolkit that helps you build a PBX from scratch. Good knowledge of Linux, Asterisk, telecom, networking are all needed to be successful with Asterisk. |
| 21:46.12 | ManxPower-work | ~freepbx |
| 21:46.13 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 21:46.13 | raden_work | ManxPower-work, don't forget patience :) |
| 21:46.14 | qsrfh | trixbox without tears it was for free all the time |
| 21:46.20 | Kobaz | qsrfh: asterisk is an ENGINE... you need to be very good with working on ENGINES |
| 21:46.24 | raden_work | slams numpad with stapler |
| 21:46.30 | Naikrovek | i like his lack of humility here. |
| 21:46.34 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 21:46.35 | Naikrovek | for some reason |
| 21:47.08 | *** join/#asterisk fnordus (~dnall@70.70.0.215) |
| 21:47.26 | raden_work | LMAO |
| 21:47.46 | Naikrovek | he doesn't like it, dammit, and he's gonna tell us |
| 21:47.52 | Naikrovek | i respect that |
| 21:47.52 | Kobaz | qsrfh: freepbx is easy since it's point and click... and happens to be built on top of asterisk... but freepbx have put in hundreds of thousands of hours of development into making it easy to use |
| 21:48.11 | Naikrovek | au revoir |
| 21:48.16 | Kobaz | qsrfh: asterisk isn't meant to be easy to use... it's meant to be flexible and powerful... and you need certain skills to even attempt to work with it |
| 21:48.21 | Kobaz | qsrfh: which you obviously lack |
| 21:49.02 | qsrfh | no i m talking about the way that it was developed |
| 21:49.12 | qsrfh | i just only talking about the manual |
| 21:49.22 | Kobaz | it's not "the manual" |
| 21:49.22 | qsrfh | i just compare the book |
| 21:49.31 | Kobaz | it's a single book, that happens to cover asterisk |
| 21:49.40 | Kobaz | there are many resources online you can use |
| 21:49.51 | qsrfh | trixboxwithouttears and asterisk oreilly |
| 21:50.09 | raden_work | qsrfh, they f'ing use trixbox |
| 21:50.19 | Kobaz | anyways |
| 21:50.24 | raden_work | then |
| 21:50.31 | raden_work | jesus I cannot type nor think today |
| 21:50.56 | Kobaz | qsrfh: I highly recommend you stray away from asterisk, and use something like trixbox instead.. it'll meet your skillset much better |
| 21:51.48 | qsrfh | haha |
| 21:51.55 | qsrfh | what a joke |
| 21:52.01 | qsrfh | thank for your advice |
| 21:52.12 | Kobaz | qsrfh: i've been working with asterisk for three years now... and I still don't know it all... are you willing to put in that much time... to become an advanced user? |
| 21:52.21 | Kobaz | qsrfh: or do you want to install something that 'just works' ? |
| 21:52.52 | qsrfh | well |
| 21:52.53 | Kobaz | if you don't want to play, and tinker, and go through the pain and suffering... use a gui |
| 21:53.04 | qsrfh | well |
| 21:53.15 | qsrfh | i got openwrt firmware |
| 21:53.18 | qsrfh | on my router |
| 21:53.24 | qsrfh | i just install asterisk on it |
| 21:53.40 | raden_work | is there a way on astra phone to just transfer a call without having to talk to the person you are transfering to ? |
| 21:54.05 | raden_work | I have to pickup a call hit Xfer ( tell the person this is who is calling ) and then hang up and it transferes |
| 21:54.07 | Kobaz | raden_work: aastra should have a blind transfer button you can enable, if i remember right |
| 21:54.18 | raden_work | if i just hit transfer and hangup call is lost |
| 21:54.26 | Kobaz | raden_work: you need to hit transfer twice |
| 21:54.34 | Kobaz | or enable the blind transfer button |
| 21:54.37 | qsrfh | as i can't install freepbx on it, i want to create a sip extension to work with my voip gsm gateway modem |
| 21:54.40 | qsrfh | that's all |
| 21:54.58 | Kobaz | qsrfh: i already pointed you to the sip.conf examples. that come with asterisk |
| 21:55.11 | Kobaz | qsrfh: but you'll need more than that... you'll have to create your dialplan to handle your call routing |
| 21:55.35 | Kobaz | qsrfh: and all of these topics are covered in the book. and on the wiki |
| 21:55.50 | qsrfh | ok |
| 21:56.15 | qsrfh | i try to read it again that book |
| 21:56.26 | qsrfh | thank you so much Kobaz |
| 21:56.33 | qsrfh | u r great |
| 21:56.35 | qsrfh | thanks |
| 21:56.35 | Kobaz | heh |
| 21:56.36 | qsrfh | bye |
| 21:56.49 | Kobaz | there's only so much I can do.. you need to put in the effort to learn the system |
| 21:57.12 | qsrfh | ok i try |
| 21:57.13 | qsrfh | bye |
| 21:57.16 | Kobaz | it's a pain... i agree... it took me about three months to figure out the basic stuff |
| 21:57.19 | Kobaz | good luck |
| 21:57.25 | qsrfh | thx |
| 21:57.27 | *** part/#asterisk qsrfh (~chatzilla@bgl93-7-88-189-218-150.fbx.proxad.net) |
| 21:58.03 | Kobaz | do de do |
| 21:58.04 | Kobaz | back to coding |
| 22:02.43 | *** join/#asterisk tiav (~tiav@ram94-3-82-225-11-215.fbx.proxad.net) |
| 22:03.18 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 22:05.10 | fifer | I have a Aastra 480i that would register to one sip extenion but I had to make some changes and now, after the new sip ext is in place and all the settings are changed on the 480i it will not register |
| 22:05.27 | fifer | No amount of reloading, rebooting phone, or anything else is making a dif |
| 22:05.36 | fifer | The sip ext works, I can connect to it with other phones |
| 22:05.53 | fifer | just not the 480i |
| 22:06.27 | Kobaz | double check your phone config versus sip.conf... turn on sip debugging... paste the logs and configs |
| 22:06.38 | fifer | nothing else has changed other than using a dif ext |
| 22:06.38 | Kobaz | ~details |
| 22:06.39 | infobot | If you want help on a topic, you HAVE to say more than "it doesn't work, help!" or else you'll get no help whatsoever. Give as many details as you can or else no one can give any suggestions. |
| 22:06.39 | *** join/#asterisk pyite (~dschreibe@unaffiliated/pyite) |
| 22:06.50 | Kobaz | fifer: obviously something else changed, otherwise it would work |
| 22:07.16 | raden_work | Kobaz, THANK YOU !!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!! |
| 22:07.29 | raden_work | omfg thats so nice :) |
| 22:07.40 | Kobaz | raden_work: heh |
| 22:07.59 | Kobaz | raden_work: which one... blind xfer button... or hitting xfer twice |
| 22:08.08 | raden_work | the twice |
| 22:08.12 | raden_work | hit it dial hit it |
| 22:08.14 | Kobaz | yeah... it's the same on polycom phones |
| 22:08.19 | raden_work | nice :) |
| 22:08.22 | Kobaz | but you should be able to turn on the blind xfer in the web interface |
| 22:08.29 | raden_work | ill have to look |
| 22:08.34 | Kobaz | you can also provision it with aastra.cfg tftp |
| 22:08.42 | raden_work | that works good though cause sometimes i do need to fill them in or something :) |
| 22:09.02 | Kobaz | you can do some cool stuff with aastra phones that i wish you could do with polycoms |
| 22:09.13 | raden_work | really ? |
| 22:09.16 | Kobaz | like, they have all those extra buttons... so you can program them to like, park and page, login/logout, etc |
| 22:09.31 | raden_work | i tried like 5 difference phones at first and aastra hands down won in the sub 150 market |
| 22:09.41 | raden_work | Kobaz, yeah i love that :) |
| 22:09.46 | raden_work | i have the 9133i's |
| 22:09.49 | Kobaz | i love polycoms... they are by far the most configurable phones, and the best audio quality |
| 22:09.55 | Kobaz | we tried 57i's |
| 22:10.07 | Kobaz | the conference feature was only partially supported |
| 22:10.25 | Kobaz | you should be able to do a three way, hang up... and the other two are on the phone |
| 22:10.35 | Kobaz | aastra, it hangs up everyone when you hang up |
| 22:12.13 | raden_work | u can do that with aastra :) |
| 22:12.41 | norrec | are you able to add in ilbc support on an asterisk install from the repos or only when compiled from source? |
| 22:12.49 | Kobaz | raden_work: oh... hmm... how? |
| 22:12.51 | raden_work | takes a lil work but can be done I cant rember what i had todo |
| 22:13.43 | raden_work | you talking like a actual 3 way call one person calls you then you hit conf and then call #3 ? |
| 22:13.53 | raden_work | and when u hang up you want to leave them to talking right ? |
| 22:13.55 | Kobaz | yeah |
| 22:14.12 | raden_work | lemee look at notes |
| 22:16.15 | raden_work | 7. To exit the conference leaving |
| 22:16.15 | raden_work | the other two parties connected, |
| 22:16.15 | raden_work | disconnect by placing |
| 22:16.15 | raden_work | the handset back on-hook or by |
| 22:16.15 | raden_work | pressing N. |
| 22:16.21 | raden_work | Just tried it on mine it works |
| 22:16.48 | raden_work | on some u need to hit the hold botton on the older ones |
| 22:16.49 | *** join/#asterisk fnordus (~dnall@70.70.0.215) |
| 22:17.03 | raden_work | and some u must press the hang up button instead of putting on hook |
| 22:18.01 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 22:18.47 | Kobaz | hmm |
| 22:18.52 | Kobaz | i think it might depend on the phone model |
| 22:19.03 | Kobaz | 57i definitly didn't support that as far as i could tell |
| 22:19.20 | Kobaz | i think all the 9x's support it though |
| 22:19.56 | Kobaz | the 480's too |
| 22:27.47 | raden_work | checking |
| 22:27.55 | raden_work | pulling 57i manual up |
| 22:28.13 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 22:28.16 | raden_work | thats what i have at home, and honestly i like the 9133i better |
| 22:28.17 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 22:28.32 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 22:28.45 | Kobaz | yeah it kinda looks boxy and old |
| 22:29.33 | raden_work | the 9133 ? |
| 22:29.42 | Kobaz | the 57i |
| 22:29.55 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 22:29.55 | *** join/#asterisk voipmonk (~shido6@12.50.179.194) |
| 22:29.55 | raden_work | aastra never made a cosmetically attractive phone |
| 22:30.00 | Kobaz | hehe |
| 22:30.07 | raden_work | the 9133i very basic but they work very good |
| 22:31.34 | jaskew | so the 480i isn't good looking after all? |
| 22:32.15 | jaskew | dang - an now it's too late to return it |
| 22:33.03 | raden_work | jaskew, they can do it |
| 22:33.17 | raden_work | when you want to leave the conference and leave them connected hit the xfer key :) |
| 22:33.26 | raden_work | all is good ;) |
| 22:33.35 | Kobaz | raden_work: wow, that's really strange |
| 22:33.44 | raden_work | If you are the Host of a conference and want to leave that conference: |
| 22:33.44 | raden_work | While in a 3-way conference press the Transfer key on the phone. |
| 22:33.44 | raden_work | Your phone leaves the 3-way conference but the remaining parties are still connected to the |
| 22:33.44 | raden_work | conference. |
| 22:33.51 | *** join/#asterisk El_Ninja (~r@adsl-76-208-10-235.dsl.sbndin.sbcglobal.net) |
| 22:34.04 | Kobaz | that's like |
| 22:34.04 | raden_work | Page 169 |
| 22:34.04 | Kobaz | really stupid |
| 22:34.04 | Kobaz | heh |
| 22:34.04 | raden_work | I know |
| 22:34.05 | raden_work | but it does it :) |
| 22:34.11 | raden_work | so now you know hope it helps |
| 22:34.17 | Kobaz | you have no idea how long i spent trying to find that |
| 22:34.17 | raden_work | there is nothing there phones cannot do |
| 22:34.18 | Kobaz | if it was there |
| 22:34.24 | raden_work | just some are really odd |
| 22:34.39 | raden_work | Kobaz, if you only knew how many times i got frusterated over blind transfer :P |
| 22:34.45 | Kobaz | hehe |
| 22:34.51 | raden_work | we both learned something today |
| 22:35.08 | jaskew | that's handy to know - glad I was lurking today :) |
| 22:35.12 | raden_work | now i can talk to the boss so much less :) |
| 22:36.20 | *** join/#asterisk hluesea (~hulusikah@88.247.127.66) |
| 22:37.11 | Kobaz | heh |
| 22:37.53 | jaskew | ~blindtransfer |
| 22:38.05 | jaskew | doesn't work... |
| 22:38.21 | raden_work | jaskew, what kinda phone ? |
| 22:38.36 | jaskew | no - I mean the infobot |
| 22:38.48 | raden_work | didnt see that |
| 22:38.55 | raden_work | needs sleep :( 4 more hours to go |
| 22:44.55 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-146.cablep.bezeqint.net) |
| 22:48.46 | *** join/#asterisk gelpg (~chatzilla@92-249-159-220.pool.digikabel.hu) |
| 22:51.10 | *** join/#asterisk cweagans (~cweagans@67.42.166.69) |
| 22:52.13 | cweagans | is there a channel that I can go to for questions regarding Cisco 7940s? It's kinda off topic here....even though I know a lot of people in here could probably answer :) |
| 22:52.24 | voipmonk | still at it? |
| 22:52.38 | cweagans | voipmonk: I am! |
| 22:52.49 | voipmonk | why? |
| 22:52.50 | raden_work | #cisco ? |
| 22:52.55 | voipmonk | you shoulda been done with that |
| 22:52.55 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 22:52.56 | voipmonk | :) |
| 22:54.02 | norrec | are you able to add in ilbc support on an asterisk install from the repos or only when compiled from source? |
| 22:55.00 | *** join/#asterisk fnordus (~dnall@70.70.0.215) |
| 22:57.12 | *** join/#asterisk SeriousMatters (~Sirius@87.115.170.210) |
| 22:58.02 | SeriousMatters | hi, I am having an NAT issue. |
| 22:58.12 | Kobaz | hmmmm |
| 22:58.20 | SeriousMatters | Asterisk is behind a router with 5060 forwarded |
| 22:58.26 | Kobaz | is DEVICE_STATE purely for userdefined state? |
| 22:58.32 | voipmonk | good |
| 22:58.44 | Kobaz | it doesn't seem to report if a phone is actually in use or ringing or not |
| 22:58.52 | voipmonk | SeriousMatters: UDP and 10,000 through 20,000 as well, right? |
| 22:58.58 | p3nguin_ | cweagans: If you have a question related to using your phone with Asterisk, ask it here. |
| 22:59.36 | *** join/#asterisk visz (~vis@irkki.fi) |
| 22:59.44 | Kobaz | how would i tell if a phone is ringing or otherwise in use... because DEVICE_STATE always returns NOT_INUSE no matter what the phone is doing |
| 22:59.44 | SeriousMatters | its not RTP, its password |
| 23:00.00 | SeriousMatters | SIP client inside the NAT is able to REGSITER, but SIP client outside get '403 forbidden, wrong password' |
| 23:00.31 | p3nguin_ | seriousmatters: Did you configure the appropriate NAT settings for the peer? |
| 23:01.05 | SeriousMatters | I checked sip debug manually, the md5-ed secrets are indeed different if client is outside. |
| 23:01.16 | *** join/#asterisk t_j (~tj@tomjudge.vm.bytemark.co.uk) |
| 23:01.21 | visz | Kobaz, if DEV_STATE cannot be used, you could just try dial and check DIAL_STATUS |
| 23:01.24 | SeriousMatters | NAT=yes, qualify=yes |
| 23:01.43 | Kobaz | visz: i don't want to dial.. i want to see if the phone is in use first |
| 23:01.47 | visz | allthough you'd want to fix devstate |
| 23:02.02 | Kobaz | i would |
| 23:02.12 | Kobaz | i think it's broken |
| 23:02.12 | norrec | SeriousMatters: did u add nat=yes in the global |
| 23:02.28 | p3nguin_ | I don't know if NAT=yes is valid -- I presume it is case-sensitive. |
| 23:02.49 | Kobaz | oh wait |
| 23:02.49 | Kobaz | hmm |
| 23:02.51 | *** join/#asterisk jelly-bean (~mikesmull@75-148-103-190-Utah.hfc.comcastbusiness.net) |
| 23:02.57 | p3nguin_ | If Asterisk is behind NAT, you need to configure externip or externhost as well as localnet. |
| 23:02.59 | Kobaz | lemme check something, it might not be broken |
| 23:02.59 | jelly-bean | what is a signaling server in the context of asterisk |
| 23:03.00 | p3nguin_ | ~sipnat |
| 23:03.01 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 23:03.21 | SeriousMatters | well, nat=yes for both global and sip friend |
| 23:03.35 | norrec | did u add externip and localnet? |
| 23:03.50 | *** part/#asterisk grey-monkey (~ericshell@75-148-103-190-Utah.hfc.comcastbusiness.net) |
| 23:03.53 | *** join/#asterisk grey-monkey (~ericshell@75-148-103-190-Utah.hfc.comcastbusiness.net) |
| 23:04.07 | *** join/#asterisk lsenft (~levisenft@75-148-103-190-Utah.hfc.comcastbusiness.net) |
| 23:04.22 | Kobaz | it seems like DEVICE_STATE only returns a non NOT_INUSE, if it's actually bridged to another device |
| 23:04.26 | SeriousMatters | externip=87.xxx.xxx.xxx at [general] |
| 23:04.32 | Kobaz | being in dialplan dosn't count |
| 23:04.48 | norrec | u need to add in the localnet as well |
| 23:05.01 | *** part/#asterisk jelly-bean (~mikesmull@75-148-103-190-Utah.hfc.comcastbusiness.net) |
| 23:05.04 | hardwire | anybody have a good inline sip trace storage solution? |
| 23:05.15 | norrec | localnet = 10.10.10.0/255.255.255.0 for example |
| 23:05.22 | norrec | otherwise it thinks all clients are local |
| 23:05.23 | hardwire | something I can throw in as a bridge or on a monitor port? |
| 23:05.24 | *** join/#asterisk codershop (~ryanoldha@75-148-103-190-Utah.hfc.comcastbusiness.net) |
| 23:05.27 | *** join/#asterisk jelly-bean (~mikesmull@75-148-103-190-Utah.hfc.comcastbusiness.net) |
| 23:05.29 | Kobaz | hardwire: /var/log/asterisk ? |
| 23:05.34 | hardwire | Kobaz: |
| 23:05.37 | *** part/#asterisk grey-monkey (~ericshell@75-148-103-190-Utah.hfc.comcastbusiness.net) |
| 23:05.39 | *** join/#asterisk grey-monkey (~ericshell@75-148-103-190-Utah.hfc.comcastbusiness.net) |
| 23:05.39 | Kobaz | tcpdump? |
| 23:05.45 | *** part/#asterisk jelly-bean (~mikesmull@75-148-103-190-Utah.hfc.comcastbusiness.net) |
| 23:06.02 | SeriousMatters | localnet=192.168.1.0/225.225.255.0 <== is this correct? |
| 23:06.35 | SeriousMatters | with sip debug on, I can see the sip exchanges, |
| 23:06.46 | Kobaz | hardwire: carrier pidgons? |
| 23:06.49 | norrec | SeriousMatters: yeah thats correct |
| 23:07.02 | norrec | SeriousMatters: make sure u reload after the change |
| 23:07.21 | norrec | SeriousMatters: also make sure ur firewall isnt doing anything to the packets |
| 23:07.41 | SeriousMatters | sip messages exchanges are fine until client sending the password to asterisk. |
| 23:07.45 | voipmonk | turn of sip alg |
| 23:07.47 | p3nguin_ | Do not enable any of the NAT settings on the remote phones, either. |
| 23:07.48 | voipmonk | turn off |
| 23:07.58 | voipmonk | turn off sip alg if your router has it on |
| 23:08.24 | *** join/#asterisk jelly-bean (~jelly-bea@75-148-103-190-Utah.hfc.comcastbusiness.net) |
| 23:08.26 | visz | Kobaz, i'm afraid there are some significant latency issues with ip over carrier pigeons (rfc 1149) |
| 23:08.33 | SeriousMatters | just as a test, I use the exact same sip friend and secrect, but the md5-ed passwords are different when they arrive asterisk. |
| 23:08.54 | p3nguin_ | Try it will plain secrets. |
| 23:09.30 | norrec | well it sounds like the phone isnt hashing them properly |
| 23:09.31 | p3nguin_ | Get as basic as possible and get it working. Then add extra things onto it. |
| 23:09.50 | Kobaz | visz: severe packet loss too, so i've heard |
| 23:10.17 | *** part/#asterisk grey-monkey (~ericshell@75-148-103-190-Utah.hfc.comcastbusiness.net) |
| 23:10.19 | *** join/#asterisk grey-monkey (~ericshell@75-148-103-190-Utah.hfc.comcastbusiness.net) |
| 23:10.22 | SeriousMatters | hm... how to use non-md5 secret? |
| 23:10.39 | p3nguin_ | secret=somethingsecret |
| 23:10.52 | jelly-bean | what is a signaling server in the context of asterisk |
| 23:11.03 | SeriousMatters | yea, that's what's in .conf |
| 23:11.12 | SeriousMatters | I never use md5secret. |
| 23:11.33 | SeriousMatters | but the sip debug shows md5 secrest anyway. |
| 23:11.40 | p3nguin_ | Then how are you receiving md5? |
| 23:12.04 | p3nguin_ | Did you enable something on the phone to do that? |
| 23:12.19 | SeriousMatters | http://pastebin.ca/1799940 ; see line 97 |
| 23:12.24 | raden_work | anyone use asterisk realtime with mysql that can tell me why they use myisam tables and not innodb ? |
| 23:13.03 | visz | Kobaz, atleast the range is better than with ip-over-bongo-drums |
| 23:13.28 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 23:13.32 | Kobaz | heh |
| 23:13.37 | Kobaz | is there an rfc for that? |
| 23:13.44 | SeriousMatters | I think, when the secret is sent through internet, they are md5-ed regardless. |
| 23:14.06 | p3nguin_ | Who is "they?" |
| 23:14.21 | SeriousMatters | but in sip.conf, we can specify an already md5-ed secret just to make sure even admin won't see the secret. |
| 23:14.40 | SeriousMatters | in case the secret is the same as billing account or something |
| 23:15.57 | *** join/#asterisk BCS-Satori (~BCS-Sator@75-148-21-113-WashingtonDC.hfc.comcastbusiness.net) |
| 23:18.09 | *** join/#asterisk simplydrew (~simplydre@pool-74-97-190-109.prvdri.fios.verizon.net) |
| 23:18.33 | BCS-Satori | Is there a site or wiki that explains the differences between 1.6.2; 1.6.1; 1.6.0 to 1.4. If building a new system should I start with 1.6.2 (sorry I am use to 1.4 currently and want to learn 1.6)? |
| 23:18.48 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 23:20.18 | p3nguin_ | I don't quite understand. I use plain secrets on my phones and sip debug still shows the digest algorithm is MD5. Based on this fact, I'm not sure why you think your system magically uses md5 secrets without configuring md5secret. |
| 23:21.25 | SeriousMatters | because all sip clients auto md5 all secrets before sending? |
| 23:21.41 | SeriousMatters | I don't know |
| 23:22.04 | p3nguin_ | I should capture some packets and see if I can find my plain secret in it. |
| 23:23.19 | p3nguin_ | I'll have to get to it later, though. |
| 23:23.20 | SeriousMatters | My asterisk box is now at home behind normal home router. |
| 23:23.29 | SeriousMatters | I don't even know if it has sip alg |
| 23:23.40 | p3nguin_ | Do you know the brand and model? |
| 23:24.04 | SeriousMatters | Thomson v585v7, rebranded as plusnet router |
| 23:24.38 | *** join/#asterisk xmitter (~xmitter@c-24-21-213-242.hsd1.or.comcast.net) |
| 23:24.59 | SeriousMatters | Thomson tg585v7, my apology |
| 23:26.35 | raden_work | whats the point of mysql tables being ROW_FORMAT=DYNAMIC ????? |
| 23:26.59 | *** join/#asterisk mythicalbox (~mythicalb@rrcs-64-183-110-250.west.biz.rr.com) |
| 23:27.10 | SeriousMatters | raden_work: lol, that sound kinda funny |
| 23:27.15 | *** join/#asterisk fnordus (~dnall@70.70.0.215) |
| 23:27.33 | raden_work | SeriousMatters, ? |
| 23:28.44 | SeriousMatters | p3nguin, I think I do have SIP ALG, and I can dig some docs to disable it through telnet. |
| 23:29.41 | SeriousMatters | but man, that's too much work. I hope it really is a router problem and won't affect me when I move the asterisk box to the production environment. |
| 23:29.56 | *** part/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 23:30.39 | p3nguin_ | I'm setting an md5 hashed response for proxy authentication in a packet capture. |
| 23:30.49 | p3nguin_ | s/setting/seeing/ |
| 23:31.06 | p3nguin_ | That's interesting. |
| 23:31.17 | raden_work | lol |
| 23:31.36 | raden_work | p3nguin_, u mess with asterisk & mysql |
| 23:31.40 | p3nguin_ | I think your presumption of SIP clients hashing automatically could be correct. |
| 23:31.52 | p3nguin_ | raden_work: Nope, I just use flat files. |
| 23:32.31 | SeriousMatters | I'm assuming no unhashed password go through Internet. |
| 23:33.22 | SeriousMatters | hm... does it mean, my SIP ALG router tried to hash an already hashed secrect? |
| 23:33.49 | SeriousMatters | or somehow rehashed elsewhere. |
| 23:34.35 | SeriousMatters | because I used the same version of SIP client in and out of NAT and using the exactly same sip friend. |
| 23:35.26 | p3nguin_ | I would check for a mistyped secret on the failing client. |
| 23:35.27 | SeriousMatters | and asterisk sip debug shows both messages arrived but different md5. |
| 23:35.54 | p3nguin_ | Assume it is wrong and retype the entire password. |
| 23:36.05 | SeriousMatters | I double checked it. |
| 23:36.16 | p3nguin_ | Assume it is still wrong and retype the entire password again. |
| 23:37.21 | SeriousMatters | I am sure it is correct as I copy+pasted directly from sip.conf |
| 23:37.46 | SeriousMatters | and I also double triple octuple checked by typing and re-C+P |
| 23:39.36 | SeriousMatters | I will leave that for now. I will move it to somewhere else to check if its my stupid router. |
| 23:39.48 | SeriousMatters | thanks a lot anyway. |
| 23:39.53 | t_j | anyone heard of Read() just dropping though on 1.6 ? |
| 23:40.07 | t_j | context ->extensions_additional.conf:exten => 8001,n(READPIN),Read(PIN,enter-conf-pin-number,,,,) |
| 23:40.25 | t_j | it does not wait for the pin to be entered |
| 23:47.02 | fifer | Is there an easy way two match a list? IE something like exten => lista,n,Dial() where the ext is in lista |
| 23:47.26 | p3nguin_ | t_j: Try adding a timeout: Read(PIN,enter-conf-pin-number,,,,10) |
| 23:48.29 | fifer | I know this sounds like an odd thing to do rather than just match a pattern but we have a reason.... |
| 23:49.38 | t_j | p3nguin_: no difference does not play the sound file either |
| 23:50.13 | voipmonk | fifer: sure u can |
| 23:50.38 | fifer | me is looking through the functions list to see if there is anything obvious |
| 23:51.22 | Kobaz | you could do a macro/gosub pattern1,1,GoSub(foo), pattern2,1,GoSub(foo) |
| 23:55.47 | Kobaz | haha |
| 23:56.00 | Kobaz | i just locked up a polycom phone really bad by sending it one too many calls |
| 23:56.18 | p3nguin_ | t_j: What is before this particular line? |
| 23:56.48 | t_j | im seeing the same thing with other Read() calls |
