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| 00:32.53 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 01:39.33 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 02:35.14 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 02:57.05 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 02:57.19 | *** part/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
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| 03:09.23 | *** join/#asterisk A-KO (as@unaffiliated/a-ko) |
| 03:10.43 | A-KO | I'm troubleshooting an asterisk issue and unfortunately I know very little about SIP/etc. I need an incoming connection from a provider to asterisk, right? Asterisk doesn't make an outgoing connection? |
| 03:12.08 | [TK]D-Fender | Asterisk does whatever you set it up to do |
| 03:12.17 | [TK]D-Fender | You have to do any specific kind of thing. |
| 03:12.18 | A-KO | unfortunately I did not set it up... |
| 03:12.29 | [TK]D-Fender | I used Asterisk to make coffe, and as a jukebox. |
| 03:12.37 | Kobaz | asterisk doesn't give me hugs at night :( |
| 03:12.56 | [TK]D-Fender | Kobaz: Then you're clearly doing it wrong :p |
| 03:13.03 | Kobaz | oooO |
| 03:13.19 | [TK]D-Fender | A-KO: Then you should consider setting it up. It works much better when it's set up |
| 03:13.28 | Kobaz | heh |
| 03:14.10 | Kobaz | 9 out of 10 agree |
| 03:15.59 | [TK]D-Fender | And we nabbed that 10th guy in the back alley afterwards... |
| 03:17.14 | A-KO | [TK]D-Fender: it was set up, but I had to re-IP the network, and now it doesn't work :P that's kind of the point... |
| 03:21.04 | [TK]D-Fender | A-KO: Well now you're at least vaguely on the path of telling us what's wrong so maybe we can help you..... |
| 03:21.37 | *** join/#asterisk pdtpatr1ck (~pdtpatric@ip184-179-74-42.oc.oc.cox.net) |
| 03:21.47 | [TK]D-Fender | A-KO: you should probably be showing us SIP DEBUG for some failed communications.... |
| 03:21.53 | [TK]D-Fender | "sip set debug on" |
| 03:21.54 | [TK]D-Fender | ~pb |
| 03:21.55 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 03:21.56 | [TK]D-Fender | ^^^^ |
| 03:23.04 | *** join/#asterisk gajini (~root@61.12.17.171) |
| 03:25.22 | A-KO | sorry I'm not suuper sure why it's broken, but I'll have to revisit this tomorrow night... |
| 03:25.30 | A-KO | since it's late |
| 03:25.44 | A-KO | nothing like redoing a rat's nest of cables |
| 03:26.00 | A-KO | I'll be on tomorrow evening to figure it out. |
| 03:26.02 | A-KO | thanks |
| 03:26.03 | *** part/#asterisk A-KO (as@unaffiliated/a-ko) |
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| 07:16.09 | ChannelZ | Was that like 'areola'? |
| 07:16.21 | din3sh | helo ppl |
| 07:17.05 | ChannelZ | ahoyhoy |
| 07:17.20 | din3sh | badly need some help, using asterisk 1.4.x, when there is double legged transfers, Caller can hear Callee but Callee only hears MOH |
| 07:17.23 | din3sh | :/ |
| 07:18.06 | *** join/#asterisk ayrjola (~ayrjola@89.18.236.11) |
| 07:19.40 | *** join/#asterisk timahvo1 (~rogue@41.80.76.146) |
| 07:24.00 | ChannelZ | Are they all transfers within the same Asterisk or what is the path? And you'd probably have to look at sip debug to try and figure out whose not getting invites and why |
| 07:24.30 | din3sh | same asterisk |
| 07:29.06 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 07:38.01 | *** join/#asterisk wonderworld (~ww@dsdf-4db5f246.pool.mediaWays.net) |
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| 08:11.51 | *** join/#asterisk Erwan (~kvirc@AMontpellier-652-1-3-125.w90-27.abo.wanadoo.fr) |
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| 08:12.24 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 08:14.52 | lwbnet | ARNING[21187]: pbx.c:4218 pbx_extension_helper: No application 'SetCDRUserField' for extension (macro-dialpstn, s, 2) |
| 08:14.56 | lwbnet | anyone? :( |
| 08:15.17 | *** join/#asterisk bn-7bc (bjarne@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 08:20.07 | thebomb | lwbnet, http://www.asteriskguru.com/tutorials/no_application_for_extension.html |
| 08:20.28 | thebomb | lwbnet, This dialplan command does simply not exist in your version of asterisk |
| 08:20.36 | thebomb | hope this helps |
| 08:20.43 | lwbnet | righttt ok |
| 08:20.44 | lwbnet | thanks |
| 08:20.58 | lwbnet | yea - im trying to migrate from asterisk 1.4 to .18 |
| 08:21.01 | lwbnet | 1.8 * |
| 08:21.14 | lwbnet | might just install matching version on the new server |
| 08:21.45 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
| 08:22.12 | thebomb | SetCDRUserField could have fallen away in 1.8 or renamed to something else ? ? |
| 08:22.26 | kaldemar | not existing in a version of asterisk is a hasty assumption. it simply might not be compiled or installed or both. |
| 08:23.31 | kaldemar | thebomb: deprecated and removed, you should use func CDR: Set(CDR(userfield)=value) |
| 08:23.56 | lwbnet | ok, well that's only logging stuff to a mysql db anyway. i'll comment it out for now |
| 08:24.46 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 08:24.48 | schmidts | good morning |
| 08:29.42 | thebomb | lwbnet, there u go better answer form kaldemar :) |
| 08:42.25 | *** join/#asterisk TSM (~the_softw@78-105-6-158.zone3.bethere.co.uk) |
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| 10:00.47 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 10:02.23 | lwbnet | -- <SIP/245h-00000006>AGI Script /uks/agi/manglecli.pl completed, returning 0 -- Auto fallthrough, channel 'SIP/245h-00000006' status is 'UNKNOWN' |
| 10:02.28 | lwbnet | hmm |
| 10:14.52 | *** join/#asterisk wonderworld (~ww@dsdf-4db5f246.pool.mediaWays.net) |
| 10:22.18 | *** join/#asterisk gregor3005 (~Benutzern@85-125-11-10.static.xdsl-line.inode.at) |
| 10:36.05 | *** join/#asterisk cmendes0101 (~nn@pool-71-118-232-170.lsanca.fios.verizon.net) |
| 10:52.02 | lwbnet | any ideas? |
| 11:03.50 | kaldemar | lwbnet: regarding what? |
| 11:05.15 | lwbnet | -- <SIP/245h-00000006>AGI Script /uks/agi/manglecli.pl completed, returning 0 |
| 11:05.19 | lwbnet | -- Auto fallthrough, channel 'SIP/245h-00000006' status is 'UNKNOWN' |
| 11:05.25 | lwbnet | my agi script runs |
| 11:05.28 | lwbnet | then that happens |
| 11:11.57 | kaldemar | looks like your extension runs out of priorities. do you have an issue? |
| 11:26.29 | *** join/#asterisk wonderworld (~ww@dsdf-4db5ce23.pool.mediaWays.net) |
| 11:27.28 | lwbnet | kaldemar: basically i have an older asterisk server running 1.4 |
| 11:27.50 | lwbnet | and a new install running 1.8 |
| 11:28.02 | lwbnet | i am trying to move over to this new server |
| 11:28.13 | lwbnet | i'm testing a phone on it currently |
| 11:37.30 | Delido192123 | Hello :) if i used callcompletionrequest, is there an way to set the callerid(num) and callerid(name) for the phone who requestet the callback? (485 called 486; 486 is not reachable 485 make set callback (ccnr) 486 is now reachable and asterisk called 485 with its own callerid) |
| 11:46.56 | *** join/#asterisk goddva (~glarsen@77.40.154.242) |
| 11:50.03 | *** join/#asterisk Bullmoose (~Bullmoose@65-129-0-91.bois.qwest.net) |
| 12:03.21 | *** join/#asterisk EmleyMoor (phil@topdeck.tinsleyviaduct.com) |
| 12:04.33 | EmleyMoor | I've just had a nuisance call with a very sneaky caller ID - 0<the number called>0708 - I am wondering if there is any way to block that kind of abuse |
| 12:06.05 | EmleyMoor | 1 short of the maximum number length |
| 12:06.32 | EmleyMoor | Is there a way to match on a string of digits within a caller ID? |
| 12:07.49 | kaldemar | yes. what do you want to do? |
| 12:08.44 | EmleyMoor | A match is required on any string over 11 digits, where 11 of them are my caller ID |
| 12:10.13 | kaldemar | yes but where do you want to put that? what do you want to do with the call? |
| 12:10.21 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 12:10.58 | EmleyMoor | In a "if caller ID matches this..." expression, and force the call either to voicemail or to a "get lost" message |
| 12:15.27 | kaldemar | $["${CALLERID(num)}" =~ "this"] |
| 12:15.42 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 12:21.33 | EmleyMoor | That should thwart them, at least until the next trick! |
| 12:24.40 | *** join/#asterisk albertoandrade (~albertoan@187.59.73.81) |
| 12:26.08 | kaldemar | EmleyMoor: so what's so sneaky and abusive about that? |
| 12:27.12 | kaldemar | EmleyMoor: are they getting into your voicemail by caller id? |
| 12:29.18 | EmleyMoor | kaldemar: No, they are trying to fool a way past my caller ID based blocking by putting my number within a fake caller ID |
| 12:30.58 | EmleyMoor | I have amended the routine to treat all caller IDs not 10-17 digits, or more than 11 but including my DDI range within, as void |
| 12:32.18 | EmleyMoor | I already treat some as void, including any I find that are not valid for returning calls |
| 12:34.10 | *** join/#asterisk mahiti-irc (~mahiti@119.30.38.44) |
| 12:34.15 | mahiti-irc | hi |
| 12:34.40 | mahiti-irc | i bought a digium te820 PRI card |
| 12:35.12 | mahiti-irc | i thought that ISP provides rj45, but now they say its rj48 |
| 12:35.26 | mahiti-irc | is rj48 supported in digium pri cards?? |
| 12:36.07 | [TK]D-Fender | mahThat is the norm |
| 12:36.12 | [TK]D-Fender | mahiti-irc, That is the norm |
| 12:37.05 | mahiti-irc | rj48?? |
| 12:37.15 | mahiti-irc | sure? [TK]D-Fender |
| 12:37.16 | [TK]D-Fender | mahiti-irc, Theya re the same physical plug, it's just a question of how the pins are wired on the connector |
| 12:38.03 | [TK]D-Fender | mahiti-irc, 1,2,4,5. those are the ones that are used, and 1/4, 2/5 are swapped for a crossover cable. |
| 12:38.27 | mahiti-irc | [TK]D-Fender, i dont get it |
| 12:38.43 | mahiti-irc | [TK]D-Fender, am not familar with wiring terms |
| 12:38.44 | [TK]D-Fender | mahiti-irc, those are the PINS on the jack that are used |
| 12:38.47 | mahiti-irc | ok |
| 12:38.52 | [TK]D-Fender | Its the SAME physical plug as normal ethernet |
| 12:38.54 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 12:39.07 | [TK]D-Fender | mahiti-irc, All you normally need is a straight-through ethernet cable from their junction box to your card normally |
| 12:39.12 | mahiti-irc | [TK]D-Fender, ya that i got to knw |
| 12:40.03 | [TK]D-Fender | mahiti-irc, But if you need a crossover cable then you will have to wire it yourself as it does not use an ethernet standard for the pins. Hence RJ48 vs RJ45 |
| 12:40.11 | mahiti-irc | u mean to say that we need terminate rj48 in a switch kinda thing and from there we use rj45? |
| 12:40.30 | [TK]D-Fender | mahiti-irc, No. |
| 12:40.54 | mahiti-irc | no [TK]D-Fender the ISP provides rj48 |
| 12:40.57 | [TK]D-Fender | mahiti-irc, RJ48 is a PIN specification. For most equopment you can use a regular RJ45, except when you need a CROSS-OVER |
| 12:41.16 | [TK]D-Fender | mahiti-irc, And you almost certainly won't requier one. |
| 12:41.22 | mahiti-irc | a cross-over is machine to machine |
| 12:41.26 | mahiti-irc | ? |
| 12:41.28 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-etulktfwyjxwxsna) |
| 12:41.28 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 12:41.30 | kaldemar | http://www.differencebetween.net/technology/difference-between-rj45-and-rj48/ |
| 12:41.40 | [TK]D-Fender | mahiti-irc, Yes. Often used when connecting channel-banks |
| 12:41.44 | mahiti-irc | ya kaldemar i saw tht |
| 12:41.53 | mahiti-irc | oh ok [TK]D-Fender |
| 12:42.39 | mahiti-irc | so then i can directly connect rj48 to the pri card? |
| 12:42.44 | [TK]D-Fender | yes |
| 12:42.52 | mahiti-irc | ok thats clears it |
| 12:43.09 | mahiti-irc | was worried that i may have to some kinda converter |
| 12:43.59 | mahiti-irc | ok great |
| 12:44.11 | mahiti-irc | thanks [TK]D-Fender and kaldemar :) |
| 12:45.09 | *** join/#asterisk timeshell (~chatzilla@gw.lusi.on.ca) |
| 12:48.37 | *** join/#asterisk blitzrage (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:48.37 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 12:50.27 | *** join/#asterisk nosaj (~jbarinas@gwb.anditel.com.co) |
| 13:00.20 | EmleyMoor | One day they'll get the message to leave me alone |
| 13:02.40 | *** join/#asterisk mintos (~mvaliyav@114.143.163.159) |
| 13:04.53 | [TK]D-Fender | EmleyMoor, nothing says "I love you" quite like a restraining order... |
| 13:12.04 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 13:16.44 | *** join/#asterisk serafie (~erin@nat/digium/x-romczpvbpprxknht) |
| 13:21.17 | EmleyMoor | In the meantime resistance is futile |
| 13:24.40 | *** join/#asterisk mirelab (~mirko@212.200.146.253) |
| 13:25.17 | mirelab | hello |
| 13:26.13 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 13:27.33 | mirelab | does anyone know how to use Asterisk cmd RealTime |
| 13:27.48 | mirelab | http://www.voip-info.org/wiki/view/Asterisk+cmd+RealTime |
| 13:28.18 | mirelab | if i use it as described it says ther is no RealTime application, like sawn in core show applications |
| 13:32.19 | *** join/#asterisk mntzn (~zion@82.131.53.231.cable.starman.ee) |
| 13:33.30 | mntzn | Hi, lots of peers got in UNREACHABLE state after a network outage in one of colocations, now their reconnecting 1 by 1, but this takes forever, can I speed things up or this is needs to be done from a phone ? |
| 13:35.18 | *** join/#asterisk jsjc (~Adium@51.Red-79-146-23.dynamicIP.rima-tde.net) |
| 13:35.19 | *** join/#asterisk timahvo1 (~rogue@41.90.24.226) |
| 13:35.22 | lifemadison | well it's just the phone doing a re-registration |
| 13:35.36 | lifemadison | or the qualify eventually hitting the device and it responding |
| 13:36.02 | mntzn | and can this be done quicker? |
| 13:36.10 | lifemadison | you can reboot the phone i guess |
| 13:36.21 | lifemadison | it's just how long it takes for the phones to become reachable |
| 13:37.19 | kaldemar | mntzn: if the future you can make then re-register faster by tweaking expiry settings in sip.conf. |
| 13:37.42 | mntzn | that's the stuff thanks |
| 13:38.45 | mirelab | exten => 11223344,1,RealTime(sipusers|name|12023243260|var_) <---- Is this depricated? |
| 13:40.25 | lifemadison | mirelab: very very very much so |
| 13:40.31 | lifemadison | it was deprecated in 1.4 and removed in 1.6.0 |
| 13:40.47 | mirelab | lifemadison: but why there is no alternative? :) |
| 13:40.51 | lifemadison | mirelab: there is |
| 13:40.52 | mirelab | it is usefull |
| 13:40.55 | lifemadison | use the REALTIME() function |
| 13:41.09 | lifemadison | there are other functions too, like REALTIME_FIELD() which is likely what you want |
| 13:41.20 | lifemadison | when something gets deprecated, it's because there is something better than it |
| 13:41.32 | lifemadison | core show functions like REALTIME |
| 13:42.12 | mirelab | ok, as I understood RealTime returned all fields as variables |
| 13:42.16 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 13:42.29 | mirelab | and with func i need to specify every field i guess |
| 13:42.37 | lifemadison | mirelab: no |
| 13:42.47 | lifemadison | REALTIME() returns everything |
| 13:42.59 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 13:43.03 | mirelab | ok thx :) |
| 13:43.21 | mirelab | lifemadison: back to more reading than :) |
| 13:45.14 | *** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk) |
| 13:49.02 | *** join/#asterisk alexoc (~alexoc@187.106.48.119) |
| 13:49.18 | *** join/#asterisk drknus (~user@blk-222-141-162.eastlink.ca) |
| 13:49.54 | alexoc | Are there any way to cnonect "3com 3102" (VCX/NBX SIP Phone) at Asterisk??? |
| 13:53.37 | [TK]D-Fender | alexoc, If it's SIP then I don't see why it wouldn't |
| 13:55.27 | alexoc | it's SIP, but I think that the phone download a soft in power cycle. |
| 13:55.50 | *** join/#asterisk nighty- (~nighty@static-68-179-124-161.ptr.terago.net) |
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| 14:48.19 | bobb_WU | can someone help me find some postgres-specific table defs for 1.8? |
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| 15:06.54 | kaldemar | bobb_WU: http://svn.digium.com/svn/asterisk/tags/1.8.9.3/contrib/realtime/postgresql/realtime.sql |
| 15:07.47 | kaldemar | bobb_WU: look under realtime/ for more tables for different DB's. |
| 15:10.35 | justdave | how do I set the document root for asterisk-gui's static files? |
| 15:11.33 | justdave | the packages in epel appear to drop all the files in /var/lib/asterisk/static-http/ but attempts to serve them from /usr/share/asterisk/static-http/ instead |
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| 16:07.30 | anonymouz666 | anyone in here using a 'manager reader' written in PHP? |
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| 16:09.44 | hff135 | hey |
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| 16:11.09 | hff135 | we are a hosted pbx service provider. i have 3 customers who have one-way audio issues on extension-extension calling. all 3 customers use a sonicwall. these customers have been running for months without any problems. all of a sudden, they each report this problem. |
| 16:12.05 | hff135 | we did make a change to our ASA/firewall last week. it had been stripping off non-standard SIP headers like "Call-Info" and "Alert Info". now, it is not. |
| 16:12.17 | hff135 | do u think this could cause the problem thru the sonicwalls? |
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| 16:22.13 | hff135 | any sonicwall gurus out there? |
| 16:24.11 | anonymouz666 | nobody from SEGA :( |
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| 16:34.58 | bbourdage | I am using a PAP2 device for faxing, I know it is not the best to do, but client would not pay for additional pots line. It works fairly well, but I am getting the following error, if the other side has an a pap2 also, can I tell asterisk not to support t.38 or disable something ? I am getting the following error NOTICE[3568]: res_rtp_asterisk.c:2238 ast_rtp_read: Unknown RTP codec 100 received from x.x.x.x WARNI |
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| 16:54.55 | p3nguin | bbourdage: The PAP2 does not support T.38. |
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| 16:59.43 | SuperNull | sooo i am not sure what password this .. box is using is there a way at the console to have it debug auth fails to other peers ? |
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| 17:13.20 | ruben23 | guys any idea when i tried asterisk command ---> asterisk -rvvvvvvvvvvvvv, asterisk -vvvvvvvvvvvgc and asterisk -> prompt messge ¨Illegal instruction¨ |
| 17:23.06 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v010-087.mobile.uci.edu) |
| 17:24.13 | p3nguin | supernull: sip set debug peer <peer name> |
| 17:25.16 | ruben23 | guys any idea why.>? ¨Illegal instruction¨ |
| 17:27.15 | pabelanger | ruben23: what are you trying to do |
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| 17:36.26 | ruben23 | pabelanger: after i compile asterisk i just run asterisk -rvvvvvvvvvv but i get this error message --> Illegal instruction¨ |
| 17:37.16 | pabelanger | ruben23: pb your output |
| 17:37.22 | pabelanger | is asterisk actually started? |
| 17:39.30 | ruben23 | <PROTECTED> |
| 17:39.49 | Qwell | ruben23: How long have you been using Asterisk now? |
| 17:39.55 | pabelanger | ruben23: ps aux | grep asterisk |
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| 17:40.32 | pabelanger | Anybody know when polycom is going to EOL their 3.3.x branch for SIP? |
| 17:40.52 | Naikrovek | will probably be a while, it just came out last year i think |
| 17:41.35 | pabelanger | Naikrovek: Ya, trying to see if they actually have a date listed some place |
| 17:41.42 | pabelanger | nothing in the release notes that I see |
| 17:41.57 | Naikrovek | yah they won't have a date until it's announced, and when it's announced it'll be a year out or so |
| 17:42.43 | Naikrovek | actual sunset date will be a long while after it's announced, i mean, and 3.3.x is the last supported version for 320 & 330. it'll be a long while |
| 17:43.42 | puzzled | it also helps that 4.x is stable with less bugs before they can think about EOL'ing 3.x |
| 17:43.57 | pabelanger | Understood |
| 17:54.08 | p3nguin | ps -C asterisk |
| 17:57.12 | bbourdage | p3nguin: I a not using t.38, I want to only use 711, anyway to tell asterisk not to use t.38 ? |
| 18:00.37 | p3nguin | Don't set any of the t38pt_udpt settings and values. |
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| 18:24.33 | din3sh | i have a problem using asterisk 1.4.x, when there is double legged transfers, Caller can hear Callee but Callee only hears MOH |
| 18:29.10 | *** join/#asterisk Meaty (cf86a622@gateway/web/freenode/ip.207.134.166.34) |
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| 18:31.51 | Meaty | Hi EveryBody! Is possible configure asterisk to make g729 outgoing calls on another sip proxy supporting g729 and making voicemail calls in g711 ? |
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| 18:33.23 | p3nguin | meaty: Yes. See the __SIP_CODEC variable. |
| 18:34.14 | Meaty | Thank i will look that! |
| 18:36.49 | p3nguin | For example, for fax over SIP, I use Set(__SIP_CODEC=ulaw) in the extension use to fax out to force the codec to ulaw (G.711u). |
| 18:38.12 | Meaty | ha cool |
| 18:40.21 | gatty | anyone familiar with the digium handsets? I need to know what the DHCP Option 66 value should be set to if the handset isn't on the same network as the Asterisk or Switchvox. |
| 18:41.39 | Qwell | gatty: You really should be using the autoprovisioning stuff with the Digium phones, rather than DHCP stuffs. |
| 18:41.42 | jpsharp | Shouldn't it be set to the IP address of your provisioning/asterisk box? |
| 18:42.43 | gatty | I tried that but it doesn't find the config file... I think it needs to be http://<ip>/<something> - I've tried (on the switchvox ones) http://<ip>/cp/ but still no joy. |
| 18:42.57 | Qwell | oh, different network - right, no avahi there. |
| 18:43.10 | Qwell | gatty: When the list comes up, select to enter the IP manually, and give it the Asterisk server IP. |
| 18:43.37 | Qwell | You are using res_digium_phone, right? You should be. |
| 18:43.50 | gatty | Qwell: well I could set up multicast routing, but would rather not and I don't know what avahi TTL is set to |
| 18:48.44 | gatty | and at the moment I'm sorting out the one on the switchvox because that's a little more urgent |
| 18:49.27 | *** join/#asterisk Defraz (~Defraz@70.36.76.167) |
| 18:49.47 | lifemadison | p3nguin: only a channel variable for that? No dialplan function? |
| 18:50.03 | lifemadison | I'm only asking because someone was asking about that recently and I couldn't find a function for that |
| 18:52.58 | Qwell | gatty: well, like I said - you can just enter the IP address, and not use option 66 at all |
| 18:53.03 | anonymouz666 | is there a way to see if a member is paused through dialplan? I don't remember |
| 18:53.25 | p3nguin | lifemadison: I only know about the variables __SIP_CODEC, __SIP_CODEC_INBOUND, and __SIP_CODEC_OUTBOUND. |
| 18:53.42 | gatty | Qwell: hmm ok, not ideal... there's no local IT support at the remote site, and it's in time zone that's +8 hours from us |
| 18:53.43 | lifemadison | p3nguin: cool, thanks |
| 18:55.20 | gatty | guess I'll just have to move the switchvox to same network and do NAT on the SIP trunks, which again is rather nasty. |
| 18:55.32 | gatty | thankfully Juniper's SIP ALG actually works |
| 18:55.32 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 18:58.30 | bmoraca_work | bullshit :P |
| 18:58.43 | bmoraca_work | disabling the Juniper SIP ALG is the only solution :P |
| 19:00.23 | gatty | yeah, now I've said it works it'll probably cause me all sorts of pain tomorrow as retaliation :P |
| 19:01.59 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 19:04.02 | lifemadison | totally :) |
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| 19:16.20 | anonymouz666 | anyone in here using a 'manager reader' (client) written in PHP? |
| 19:16.27 | anonymouz666 | (sorry to ask twice) |
| 19:17.02 | anonymouz666 | I want to know if anyone has problems with the client losing connection to AMI when there are high load |
| 19:17.14 | anonymouz666 | (PHP sucks for that) |
| 19:22.48 | lifemadison | anonymouz666: just used Shift8 PHP framework for that, but not heavily so no idea if it has the same issue |
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| 19:50.24 | KavanS | any links/examples of a 3 tier dialout macro for failsafe/redundant dialing? |
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| 19:56.29 | lifemadison | KavanS: basically it's just Dial(), check DIALSTATUS, try next Dial() |
| 19:57.00 | KavanS | k, didn't know if there was a popular example, or something liek that I could use... |
| 19:58.38 | lifemadison | I'm sure you could find something on the google, but basically, ya, it's just a loop that tries Dial() a few times |
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| 19:59.54 | lifemadison | I'm sure you could easily create a GoSub() that does a While($[${EXISTS(${ARG${x}})}]) -- Dial(${ARG${x}}) |
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| 20:00.20 | lifemadison | Set(x=${INC(x)}) |
| 20:00.25 | lifemadison | something like that |
| 20:01.03 | KavanS | ok |
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| 20:24.17 | *** join/#asterisk sanman (~aaron@ip-216-93-249-210.twdx.net) |
| 20:24.20 | sanman | hello |
| 20:25.20 | sanman | does anyone know if it is possible to dial any arbitrary number with chan_gtalk ? I have it configured and I see how I can dial people that are my buddies, but I don't see a way to dial any old phone number |
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| 20:34.49 | [sr] | so hot in here! |
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| 20:37.12 | sirajperson | howdy all, I have trying to connect my soundpoint IP450 phone to asterisk |
| 20:37.33 | sirajperson | can get it to work |
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| 20:47.19 | thecardsmith | sanman: exten => _X.,n,Dial(Gtalk/accountname/+8028675309@voice.google.com) |
| 20:47.29 | thecardsmith | that's how I do it in my dial plan, to dial on google talk |
| 20:47.42 | thecardsmith | errr, google voice, however you wanna say it |
| 20:49.09 | thecardsmith | also that phone number is bogus, and you might need the +1, pretty sure you need the +1NPANNX1234 like that |
| 20:49.56 | ChannelZ | I called it and got a sex line. |
| 20:50.27 | dym | i called it and got a pizza delivery service |
| 20:50.39 | thecardsmith | i got a pizza delivery service sex line |
| 20:50.52 | thecardsmith | "what are you wearing? do you want extra cheese?" |
| 20:51.09 | ChannelZ | Only if you put it on my sausage |
| 20:51.12 | ChannelZ | errr |
| 20:51.15 | thecardsmith | bahahahaha |
| 20:51.24 | ChannelZ | moving along |
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| 21:14.43 | sanman | thecardsmith: I've got this exten => _+1NXXNXXXXXX,1,Dial(Gtalk/aaroncirillo@gmail.com/${EXTEN}@voice.google.com) |
| 21:15.26 | sanman | I wonder if there's anything I need to configure on the google voice side? |
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| 21:19.45 | sanman | when I dial from my soft phone I get an error in the asterisk terminal: chan_gtalk.c:2082 gtalk_parser: Remote peer reported an error, trying to establish the call anyway |
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| 21:19.51 | *** mode/#asterisk [+o angler] by ChanServ |
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| 22:55.29 | ChannelZ | I'm confused about Pickup() - I don't know if something changed or what |
| 22:57.21 | ChannelZ | I have an internal extension *72XX that does a Pickup of extension 2XX (whatever you dial.) I do Pickup(${EXTEN:2}@internal) |
| 22:58.16 | ChannelZ | Now someone calls in on DAHDI/1-1 and dials extension 200 which in turn does a Goto into my 'internal' context and dials SIP/Foo |
| 22:59.18 | ChannelZ | I'm at another phone and go to pick it up with *7200 but it tells me "No target channel found for 200" |
| 23:00.14 | ChannelZ | I swear this used to work as-is, I don't remember changing the context of the Pickup or anything. |
| 23:02.30 | ChannelZ | In any case am I wrong, I thought you want to Pickup the extension number someone dialed to cause the device to ring, right? Or should it be in the context that the original person dialed, not the context that actually did the Dial()? if that makes any sense |
| 23:05.05 | ChannelZ | oh.. maybe this is something else... |
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| 23:26.08 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 23:32.21 | Posi211 | hello to all, I'm new |
| 23:40.03 | p3nguin | ~ask |
| 23:40.04 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 23:40.06 | p3nguin | ~pb |
| 23:40.06 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 23:40.17 | p3nguin | posi211: You'll want to know those two things. |
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| 23:44.36 | ChannelZ | hmm ok I have no idea why this damn thing doesn't work |
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| 23:46.26 | *** join/#asterisk onixx99 (1000@bas1-stetherese38-2925306126.dsl.bell.ca) |
| 23:47.31 | onixx99 | hello, I am in desperate need of help. I have upgraded to asterisk 1.8 and since then, my app_alarmreceiver stopped working. it has to do with DTMF detection that behaves diffently than on 1.4 |
| 23:48.01 | ChannelZ | so Pickup wants the context that the ringing device is in as if it were making an outgoing call.. !? that seems different |
| 23:48.48 | onixx99 | on 1.4, each DTMF were decoded properly by asterisk (inband) with emulation messages for each |
| 23:49.26 | onixx99 | now on 1.8, I get errors like DTMF end '2' has duration 59 but want minimum 80, emulating on |
| 23:56.40 | ChannelZ | onixx99: did you build yourself from source? |
| 23:56.55 | onixx99 | ChannelZ: yes I did |
| 23:58.30 | onixx99 | ChannelZ: I am trying to understand what may be going on in channel.c |
| 23:58.57 | ChannelZ | hmm looking at the source the min duration actually seems the same between 1.4 and 10 anyway |
| 23:59.21 | onixx99 | ChannelZ: correct ! (hence why I am enven more confused.....) |
| 23:59.34 | *** join/#asterisk timahvo1 (~rogue@197.178.9.192) |
| 23:59.46 | onixx99 | ChannelZ: basically, it goes on channel.c: 4040 |