| 00:06.02 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 00:09.13 | ChannelZ | hmm ok here's a stupid one, is there a way to see what MOH file is currently playing on a channel? |
| 00:15.59 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 00:23.48 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip68-2-227-192.ph.ph.cox.net) |
| 00:38.12 | *** join/#asterisk datruth (datruth@gurutechie.org) |
| 00:50.30 | *** join/#asterisk gg608f (~gg608f@c-67-180-129-182.hsd1.ca.comcast.net) |
| 00:55.13 | *** join/#asterisk iq (~chatzilla@ip24-252-27-163.om.om.cox.net) |
| 01:09.23 | *** join/#asterisk sustav (~vpp@76.73.166.16) |
| 01:13.04 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 01:13.04 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 01:15.39 | *** join/#asterisk GGD (~deberle@pool-173-72-206-210.clppva.fios.verizon.net) |
| 01:20.21 | *** join/#asterisk Weezey (~ohno@wap54g-02.loit.ca) |
| 01:20.34 | Weezey | Hey party people |
| 01:20.58 | Weezey | I set up DUNDi for the first time today, pretty delightful stuff that. |
| 01:25.58 | *** join/#asterisk aross42 (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 01:27.13 | WIMPy | Unless you accidentally give it a wrong key. It won't give you a hint why it's failing. |
| 01:29.33 | WIMPy | Other than that it's a great thing. |
| 01:30.00 | carrar | But is it better then toast! |
| 01:30.14 | carrar | or Toast with Bacon! |
| 01:32.48 | Weezey | WIMPy: yeah, forgot to reload res_crypto and couldn't figure out what was going on |
| 01:36.09 | *** join/#asterisk slingr (~phunguy@will.one.day.hack-the-pla.net) |
| 01:41.10 | *** part/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 01:45.59 | gg608f | hi group anyone installed LUA module in asterisk succesfully? |
| 01:46.31 | Weezey | I'm having an issue with PHP AGI |
| 01:48.26 | Weezey | the $agi->get_data("soundfile", 15000, 15) isn't playing the file or waiting 15000 milliseconds |
| 01:48.58 | Weezey | the verbose output shows the Playback("soundfile") line |
| 02:00.35 | *** join/#asterisk eicto (~eicto@eicto.broker.freenet6.net) |
| 02:08.16 | *** join/#asterisk mzb (~mzb@2001:44b8:512d:7501:f66d:4ff:fe90:9629) |
| 02:13.53 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 02:16.57 | *** join/#asterisk [ctrl][alt][del] (~Jabber@70.116.68.114) |
| 02:24.54 | *** join/#asterisk jackmcbarn (47c77dd2@gateway/web/freenode/ip.71.199.125.210) |
| 02:29.00 | jackmcbarn | im having a strange issue: i have an asterisk server behind nat with the sip port forwarded. when a sip client outside the nat and a sip client inside the nat try to talk (no matter which end the caller is), there's no audio either way. when either the outside or the inside talks to a dahdi client, it works fine. i set directmedia=no for all the sip clients |
| 02:32.05 | datruth | would love to get a working asterisk server |
| 02:36.40 | *** join/#asterisk infinity- (infinity@ultra30.tptp.cc) |
| 02:37.53 | infinity- | Hi, I have a laptop that doesn't come with very accurate echo cancellation software... I'm trying to be able to make calls without a headset or headphones on, although when I do an echo test I hear echos, that progressively get louder, and sound like water sort of. |
| 02:38.06 | infinity- | Is there anyway I can fix this at the asterisk level or do I have to pay for better echo cancellation software? |
| 02:41.31 | infinity- | I was looking at solicall although that is $42 and I'd rather fix it at the asterisk level -^.^- |
| 02:44.31 | infinity- | AFK - will idle for reponze -^.^- |
| 02:45.05 | Weezey | infinity-: tried Blink? |
| 02:46.04 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 02:46.16 | infinity- | Not yet Weezey |
| 02:47.08 | infinity- | Will inztall now I"ve been using 3cxphone |
| 02:49.02 | infinity- | wow their download server is soooooo laggy can some1 mirror it |
| 02:50.54 | infinity- | Ah I found it on softpedia |
| 02:51.30 | infinity- | I can't download it Weezey |
| 02:51.33 | infinity- | can you please email it to me |
| 02:51.44 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 02:52.41 | infinity- | It uzez teh same mirror |
| 02:52.45 | infinity- | Their site is down for me |
| 03:01.51 | *** part/#asterisk ruben23 (~John_Carl@121.97.111.142) |
| 03:33.40 | *** join/#asterisk asteriskmonkey (~philip@74-51-38-204.telnetcommunications.com) |
| 03:34.29 | asteriskmonkey | hey im running asterisk 1.10 i have an odd issue, I have a macro that recieves a fax and i also have a h exten in there.. when the fax is received it dosnt seem to be calling the h (hanup) exection stuff |
| 03:35.47 | asteriskmonkey | can anyone shed light on that? |
| 03:37.03 | ChannelZ | is it in the same context? |
| 03:37.14 | asteriskmonkey | yes its in the macro context |
| 03:37.54 | asteriskmonkey | old examples of fax show exection of lines after an rxfaxl line, i found this to be false also |
| 03:38.29 | asteriskmonkey | ive got a macro call rx-fax.. code like this |
| 03:38.34 | asteriskmonkey | exten => s,n,ReceiveFAX(/tmp/${FAXFILE}.tiff) |
| 03:38.35 | asteriskmonkey | exten => s,n,NoOp("Fax Came in") |
| 03:38.42 | asteriskmonkey | also |
| 03:38.44 | asteriskmonkey | exten => h,1,NoOp("got the fax now going to process it") |
| 03:38.46 | asteriskmonkey | for tracing |
| 03:38.52 | asteriskmonkey | its not hitting the noop on either |
| 03:38.53 | asteriskmonkey | :( |
| 03:39.28 | ChannelZ | it should be in the context you called the macro from. Macros are special. |
| 03:39.57 | asteriskmonkey | I need the hangup or after action to live in the macro |
| 03:40.31 | *** join/#asterisk Zopsi (1818afbd@gateway/web/freenode/ip.24.24.175.189) |
| 03:40.36 | ChannelZ | why |
| 03:40.48 | asteriskmonkey | have other h' rules elsewhere |
| 03:41.14 | asteriskmonkey | so with macros then you cant have anything after an action that cuases a success? |
| 03:41.30 | asteriskmonkey | i remember this working in prior versions |
| 03:41.33 | asteriskmonkey | perhaps im mistaken |
| 03:43.19 | ChannelZ | I want to say it was broken behavior in the past but I could be wrong. Maybe someone else who uses macros more than me will have a definitive answer. |
| 03:45.54 | asteriskmonkey | no worries, will code an ugly work around, seems n stat not called either from parent context :? |
| 03:48.03 | *** join/#asterisk mcf37821 (~mcf3782@adsl-065-012-184-148.sip.asm.bellsouth.net) |
| 03:50.23 | *** part/#asterisk infinity- (infinity@ultra30.tptp.cc) |
| 03:53.38 | Zopsi | Has anyone configured a polycom spectralink 8020 without the dual charger? |
| 03:53.53 | *** join/#asterisk ilj (~ilj@sourcemage/grimoire/apprentice/ilj) |
| 04:33.16 | *** join/#asterisk twanny796 (~twanny@46.11.14.192) |
| 04:35.16 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 04:47.46 | *** join/#asterisk darksk1ez (~mhb@fsf/member/darkskiez) |
| 05:18.59 | *** join/#asterisk gg608f (~gg608f@c-67-180-129-182.hsd1.ca.comcast.net) |
| 05:19.13 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-gijuzkrerbtuxqso) |
| 05:20.49 | *** join/#asterisk cleron (~cleronx@d75-154-225-254.bchsia.telus.net) |
| 05:22.44 | cleron | hi can anyone help me? i have FreePBX installed on cent os, im trying to get the incoming fax to email working. can some one tell me weather this is possible or do i need other third party modules installed? |
| 05:24.20 | cleron | is anyone awake in here? |
| 05:28.21 | Nugget | Nope. |
| 05:28.37 | Nugget | You might have better luck asking in a FreePBX channel |
| 05:34.52 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 05:45.39 | *** join/#asterisk l2trace99 (~l2trace@rrcs-71-43-104-238.se.biz.rr.com) |
| 05:59.48 | *** join/#asterisk ThomasLocke (~ThomasLoc@pdpc/supporter/active/thomaslocke) |
| 06:01.31 | *** join/#asterisk fling (~fling@unaffiliated/fling) |
| 06:04.44 | *** join/#asterisk vlad_starkov (~vlad_star@ppp91-79-91-232.pppoe.mtu-net.ru) |
| 06:11.52 | *** join/#asterisk bn-7bc (~bjarne-im@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 06:22.00 | datruth | can anyone here point me to a how to for internet phones and asterisk? |
| 06:22.00 | *** join/#asterisk RJV123456 (~RJV123456@2002:4ca7:72df:0:b81d:93c3:39d0:b4d5) |
| 06:24.36 | *** part/#asterisk RJV123456 (~RJV123456@2002:4ca7:72df:0:b81d:93c3:39d0:b4d5) |
| 06:38.35 | *** join/#asterisk s[x] (~sx]@eth589.qld.adsl.internode.on.net) |
| 06:42.00 | ChannelZ | ~thebook |
| 06:42.00 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 07:04.00 | *** join/#asterisk bad53 (~bad53@94.98.214.89) |
| 07:18.51 | *** join/#asterisk eject_ck (~eject@109.86.138.135) |
| 07:28.07 | *** join/#asterisk bad53 (~bad53@94.98.214.89) |
| 07:28.50 | *** join/#asterisk v0lZy (~chatzilla@mail.silk-group.net) |
| 07:28.59 | v0lZy | lo |
| 07:29.07 | bad53 | hi guyz, i am looking for asterisk developer who is willing to work part time |
| 07:36.32 | datruth | hi v0lZy can you help me with a ip based configuration setup? |
| 07:37.15 | *** join/#asterisk djinni (~djinni@li125-242.members.linode.com) |
| 07:37.44 | *** join/#asterisk troyt (~troyt@2001:1938:240:3000::3) |
| 07:38.21 | v0lZy | i can try |
| 07:40.09 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 07:40.22 | datruth | awesome |
| 07:40.47 | datruth | So I have asterisk going I've configured the asterisk.conf/manager.conf/http.conf |
| 07:40.59 | datruth | So far I am unable to see the configure via the web site |
| 07:41.11 | datruth | but I can access certain commands |
| 07:41.45 | datruth | Can I configure this with out the web gui? If so what configuration files will I need to edit as there are many? |
| 07:44.43 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 07:47.30 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 07:51.19 | *** part/#asterisk eject_ck (~eject@109.86.138.135) |
| 07:57.31 | v0lZy | hold on |
| 07:57.35 | v0lZy | which webgui? |
| 07:58.26 | v0lZy | well, at the minimum, you will need to check sip.conf file and extensions.conf |
| 07:59.33 | v0lZy | in sip.conf you define your provider, and each phone, in extensions.conf you set what should happen when certain numbers are dialed |
| 07:59.40 | v0lZy | all these files should have examples in them already |
| 07:59.56 | v0lZy | as for the webgui, i dont think im the right person to ask. i use askoziapbx... |
| 08:02.32 | *** join/#asterisk jkroon (~jkroon@196.38.199.207) |
| 08:09.07 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 08:09.15 | *** join/#asterisk fskrotzki_ (~fskrotzki@cpe-67-253-245-174.rochester.res.rr.com) |
| 08:12.11 | *** join/#asterisk los415 (~LOS@sfca-office.corp.race.com) |
| 08:19.08 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 08:28.34 | *** join/#asterisk vlad_starkov (~vlad_star@178.177.175.35) |
| 08:31.09 | *** join/#asterisk doolittlework (~doolittle@41-134-22-10.dsl.mweb.co.za) |
| 08:39.34 | *** join/#asterisk vlad_starkov (~vlad_star@178.177.175.35) |
| 08:57.50 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 09:05.47 | *** join/#asterisk eject_ck (~eject@109.86.138.135) |
| 09:06.50 | eject_ck | Hi all, I have 10 SIP peers and want use all of them using round-robin, how can I do that ? |
| 09:07.05 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 09:08.30 | Gugge | eject_ck: you could save the last used one in astdb, and choose the next one |
| 09:11.02 | eject_ck | Gugge: thank you very much ! |
| 09:12.48 | eject_ck | can this be done usingh queues ? |
| 09:13.27 | Gugge | maybe |
| 09:14.32 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
| 09:32.09 | *** join/#asterisk _zoom_ (~Eissa@196.1.219.122) |
| 09:33.46 | _zoom_ | hello, I'm looking for a asterisk based call solution |
| 09:55.15 | *** join/#asterisk vanguarde9 (~kica@dash.cnsk.sk) |
| 09:55.45 | vanguarde9 | Olla! |
| 10:16.35 | *** join/#asterisk jmls (~julian@host217-36-208-155.in-addr.btopenworld.com) |
| 10:28.22 | dax_roc | Can you initiate a background() before answer() for incomming sip/iax to replace the ring (progress) tone? |
| 10:40.12 | *** join/#asterisk Carlos_PHX1 (~Carlos@ip68-2-227-192.ph.ph.cox.net) |
| 10:49.02 | *** join/#asterisk mimage (~mimage@fedora/mimage) |
| 11:07.41 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 11:08.18 | *** join/#asterisk bad53 (~bad53@94.98.214.89) |
| 11:16.30 | vanguarde9 | Hello people is there anybody familiar with RTCP in asterisk ? |
| 11:16.48 | vanguarde9 | i have some question about computing of jitter |
| 11:20.20 | vanguarde9 | off course i have read rfc |
| 11:20.22 | *** join/#asterisk e7e5 (~rudenko@188.134.2.33) |
| 11:20.36 | vanguarde9 | but im still unsure how to compute jiiter |
| 11:21.17 | vanguarde9 | This jitter computation from asterisk sources |
| 11:21.20 | vanguarde9 | prog = (double)((timestamp-rtp->seedrxts)/(float)(rate)); |
| 11:21.20 | vanguarde9 | dtv = (double)rtp->drxcore + (double)(prog); |
| 11:21.20 | vanguarde9 | current_time = (double)now.tv_sec + (double)now.tv_usec/1000000; |
| 11:21.20 | vanguarde9 | transit = current_time - dtv; |
| 11:21.20 | vanguarde9 | d = transit - rtp->rxtransit; |
| 11:21.20 | vanguarde9 | rtp->rxtransit = transit; |
| 11:21.20 | vanguarde9 | if (d<0) |
| 11:21.20 | vanguarde9 | d=-d; |
| 11:21.20 | vanguarde9 | rtp->rxjitter += (1./16.) * (d - rtp->rxjitter); |
| 11:21.36 | vanguarde9 | * This is jitter computation from asterisk sources |
| 11:23.42 | leifmadsen | vanguarde9: |
| 11:23.44 | leifmadsen | ~pb |
| 11:23.44 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 11:27.24 | *** join/#asterisk jmls (~julian@host217-36-208-155.in-addr.btopenworld.com) |
| 11:29.22 | vanguarde9 | I know sorry if I am floodin , but code was short |
| 11:29.40 | vanguarde9 | and people are lazy |
| 11:31.42 | vanguarde9 | Please if somebody can tell me what are rtp->seedrxts and tp->drxcore |
| 11:31.42 | kaspar | hello everybody! I've got a problem with setting CDR(userfield). I need to capture the sip qos data from a slave-channel |
| 11:32.29 | kaspar | so in h,n,* I set MASTER_CHANNEL(rtpstats) with the values, |
| 11:33.19 | kaspar | then I set CDR(userfield)=${rtpstats}. |
| 11:34.25 | kaspar | The asterisk log correctly shows the Set(CDR(userfield)=xxx), but userfield doesn't show up in Master.csv. |
| 11:34.45 | kaspar | loguserfield is set to yes, |
| 11:35.04 | kaspar | and directly setting CDR(userfield) on the master channel works, too |
| 11:37.31 | kaspar | I'm executing the h, stuff using the "e" flag for Dial(). |
| 11:37.44 | kaspar | full code of my h, is here: http://pastebin.ca/2174830 |
| 11:37.49 | kaspar | Any ideas? |
| 11:39.27 | leifmadsen | vanguarde9: your lack of using a pastebin was lazy |
| 11:40.17 | leifmadsen | kaspar: is cdr.conf configured to close out the CDR before the h extension is run? |
| 11:41.52 | vanguarde9 | leifmadsen: :-[ |
| 11:41.55 | leifmadsen | kaspar: this will also execute a NoOp() and never a Set(CDR(userfield)...): exten => h,n,GotoIf($[1=1]?end) |
| 11:42.52 | leifmadsen | and your first priority looks like it would set a null value to userfield if rtpstats is not set |
| 11:43.01 | leifmadsen | (prior to entering the h exten) |
| 11:43.07 | leifmadsen | your logic seems funky |
| 11:44.16 | kaspar | leifmadsen: endbeforehexten is at it's default value (no), so it's not closed out before h, right? |
| 11:44.30 | *** join/#asterisk italorossi (~Adium@201.76.151.254) |
| 11:44.43 | *** join/#asterisk Bullmoose (~Bullmoose@71-33-18-80.bois.qwest.net) |
| 11:45.14 | leifmadsen | kaspar: right -- in this case I think your logic is a little.... off |
| 11:45.41 | kaspar | leifmadsen: h gets executed twice: once for the slave (SIP) channel, once for the master channel. I want the slave channel to set the master channel's rtpstats, then exit, |
| 11:46.02 | leifmadsen | well that is highly dependent on who hung up too |
| 11:46.12 | kaspar | leifmadsen: and the master channel should just set the rtpstats |
| 11:46.13 | leifmadsen | as to which 'h' will get executed and such |
| 11:48.07 | WIMPy | Did anyone ever do a truth table or something for when h is executed on which channel? |
| 11:48.22 | leifmadsen | I never did |
| 11:48.57 | WIMPy | I just put h extensions everywhere to observe what's happening, but it doesn't really seem consistant to me, yet. |
| 11:54.36 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 11:55.35 | kaspar | I'm using the "e" flag on Dial(SIP/*,...) so that h gets executed for both the master channel and the SIP channel |
| 11:55.58 | *** join/#asterisk italorossi (~Adium@201.76.151.254) |
| 11:56.01 | kaspar | in my case, only the master channel has the variable "has_sip_peer" set |
| 12:01.24 | WIMPy | Oh, and while we're at documentation: Does anyone have an idea, when DIALSTATUS would be set to ANSWER? Does that only hppen with g? Other options? |
| 12:05.07 | v0lZy | WIMPy: |
| 12:05.12 | v0lZy | issue that we talked about yesterday |
| 12:05.17 | v0lZy | works on aastra phones now |
| 12:05.21 | v0lZy | but not on vigor draytek phones |
| 12:05.28 | WIMPy | cool |
| 12:05.29 | v0lZy | aastra updates after pickup |
| 12:05.30 | v0lZy | vigor doesnt. |
| 12:05.33 | v0lZy | native |
| 12:05.38 | v0lZy | on 1.8.4.4 |
| 12:05.55 | v0lZy | sendrpid=pai rpid_update=yes |
| 12:06.00 | v0lZy | thats all thats needed i think |
| 12:06.05 | WIMPy | Bad luck. Had the impression the Vigorpohones were't that bad. |
| 12:06.06 | v0lZy | but the phone has to support it |
| 12:06.21 | v0lZy | WIMPy: well.. they arent bad.. they do regularly update firmware |
| 12:06.30 | v0lZy | but the problem is they outsource people for it apparently when they get enough requests |
| 12:06.33 | WIMPy | Yes. That's the usual issue with SIP. You never know wich details will work. |
| 12:06.36 | v0lZy | so its a bit irregular in terms of updates |
| 12:06.44 | v0lZy | on the second hand |
| 12:06.52 | v0lZy | my solution that we came up yesterday |
| 12:06.55 | v0lZy | has 1 major drawback |
| 12:07.08 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 12:07.17 | v0lZy | when i get called i store the number and pass it on when i do transfers etc |
| 12:07.25 | v0lZy | but when i then make calls later, people see the last number that called me |
| 12:07.28 | v0lZy | trying to work that out |
| 12:07.33 | v0lZy | i think i have a solution that might work |
| 12:07.40 | v0lZy | but man, im tearing up my dialplan as i go .D |
| 12:08.12 | WIMPy | Is there anything left from that monster you still need , then? |
| 12:08.59 | *** part/#asterisk ickmund (~ickmund@cli-5b7e85f8.bcn.adamo.es) |
| 12:09.04 | *** join/#asterisk italorossi (~Adium@201.76.151.254) |
| 12:12.08 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-dntthspvlkqacmnm) |
| 12:12.08 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 12:12.32 | v0lZy | refresh rate is a bit slow |
| 12:12.40 | v0lZy | wrong win |
| 12:12.47 | v0lZy | i dont know |
| 12:19.22 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 12:22.00 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 12:22.43 | *** join/#asterisk italorossi (~Adium@201.76.151.254) |
| 12:22.49 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 12:35.48 | *** join/#asterisk n3hxs (~ed@ann100fw01.answernet.com) |
| 12:40.51 | *** join/#asterisk thecardsmith (~quassel@pdpc/supporter/student/thecardsmith) |
| 12:43.05 | *** join/#asterisk _Corey_ (~chatzilla@64.215.11.114) |
| 12:43.28 | carrar | must |
| 12:43.29 | carrar | get |
| 12:43.31 | carrar | coffee |
| 12:45.55 | WIMPy | Coffee is no solution, but alcohol is. |
| 12:47.18 | *** join/#asterisk italorossi (~Adium@201.76.151.254) |
| 12:47.20 | Roelt | coffee is an intermediate step |
| 12:47.42 | Roelt | (coffee & work) => money => alcohol. |
| 12:48.43 | WIMPy | puts some "will work for beer" sings up for sale. |
| 12:48.54 | *** join/#asterisk din3sh (~din3sh@41.212.202.31) |
| 12:49.01 | din3sh | hello all |
| 12:49.21 | carrar | seems I am working in reverse then |
| 12:49.31 | carrar | alcohol was the gateway to coffee for me |
| 12:49.44 | n3hxs | alcohol in the coffee |
| 12:49.56 | n3hxs | nah, coffee in the alcohol... |
| 12:49.57 | din3sh | how do we use the [*81 Roll call]/ [/* *82 Eject all non-admins */] etc in meetme? |
| 12:50.35 | carrar | I'm guessing once I move off of coffee it will be prune juice or something |
| 12:51.26 | *** join/#asterisk italorossi (~Adium@201.76.151.254) |
| 13:00.54 | [TK]D-Fender | Coffee is a solution, and alcohol is a molecule |
| 13:02.28 | *** join/#asterisk aross42 (~aross@CPE009400809a9c-CMb89bc9d2e1a5.cpe.net.cable.rogers.com) |
| 13:03.25 | zorzar | hey, i'm confused about asterisk's behaviour regarding T.38 in perticular values set in T38FaxMaxDatagram. i have two devices that set T38FaxMaxDatagram to 244 and asterisk sets it to 960. shouldn't asterisk use the common minimum? |
| 13:07.11 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 13:07.28 | *** join/#asterisk s[x] (~sx]@ppp59-167-157-96.static.internode.on.net) |
| 13:07.58 | s[x] | Hey guys, anyone got any experience with Vegastream Products |
| 13:09.43 | *** join/#asterisk amessina (~amessina@2001:470:1f11:a4:d6be:d9ff:fe8d:7c1e) |
| 13:14.21 | Katty | herp. derp. |
| 13:14.40 | Nivex | derpy hooves? |
| 13:15.35 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
| 13:15.38 | n3hxs | Morning |
| 13:15.44 | Katty | hugs n3hxs |
| 13:15.56 | n3hxs | Katty, gets a hug back! |
| 13:17.35 | drmessano | merp |
| 13:19.12 | Katty | we've scared jaytee off |
| 13:19.16 | Katty | that's unfortunate. |
| 13:22.35 | *** join/#asterisk italorossi (~Adium@201.76.151.254) |
| 13:23.07 | *** join/#asterisk edwin_quijada (~macaruchi@190.122.124.128) |
| 13:23.16 | edwin_quijada | Hi! |
| 13:24.32 | edwin_quijada | Question:When asterisk do calls using callfiles if the lines are busy it trys to do the call, anyway or wait for available channel ? |
| 13:24.34 | *** join/#asterisk jsjc (~Adium@27.Red-83-59-180.dynamicIP.rima-tde.net) |
| 13:26.21 | [TK]D-Fender | edwin_quijada, No. |
| 13:26.50 | [TK]D-Fender | edwin_quijada, * will process it as-is and call the Channel: for X retries on the delay specified |
| 13:27.17 | [TK]D-Fender | edwin_quijada, Rather that directly calling out on an actual dialing tech, use a Local channel and put your redial logic there |
| 13:28.57 | edwin_quijada | so if all channels in the context are busy * will do a retry it will not wait for available channel |
| 13:28.58 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:28.58 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:29.42 | *** join/#asterisk italorossi (~Adium@201.76.151.254) |
| 13:30.09 | *** join/#asterisk serafie (~erin@nat/digium/x-iewjjtbtleshboth) |
| 13:30.10 | edwin_quijada | How can detect if all channels are busy ? |
| 13:30.40 | [TK]D-Fender | check to see if your call FAILED. |
| 13:30.54 | [TK]D-Fender | There are obvious vars created with that info |
| 13:32.19 | *** join/#asterisk italorossi (~Adium@201.76.151.254) |
| 13:32.53 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 13:33.02 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 13:34.36 | edwin_quijada | I tried to check FAILED in cdr but it is not recording the FAILED status |
| 13:34.57 | [TK]D-Fender | vars <- |
| 13:35.08 | [TK]D-Fender | "core show application dial" |
| 13:35.46 | carrar | -bash: core show application dial: command not found |
| 13:35.56 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 13:35.56 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 13:36.21 | carrar | smirks! |
| 13:36.26 | *** join/#asterisk italorossi (~Adium@201.76.151.254) |
| 13:36.37 | puzzled | WIMPy: ping |
| 13:38.26 | rolandow | carrar: in asterisk cli :) |
| 13:38.34 | carrar | OH!!! |
| 13:38.39 | carrar | SILLY ME |
| 13:39.03 | rolandow | oh |
| 13:39.07 | rolandow | missed your smirk |
| 13:39.11 | rolandow | SILLY ME\ |
| 13:39.22 | rolandow | doh |
| 13:39.28 | carrar | heh |
| 13:39.30 | carrar | good morning! |
| 13:39.55 | rolandow | well.. afternoon here already .. already had my best part of the day i suppose :) |
| 13:40.09 | carrar | Where is here? |
| 13:40.16 | rolandow | netherlands |
| 13:40.18 | carrar | < Seattle |
| 13:40.20 | carrar | oh |
| 13:40.25 | carrar | thats WAY over there |
| 13:40.30 | edwin_quijada | This is my context for failed extension is correct ? http://pastebin.com/Mp2i76pj |
| 13:40.31 | carrar | over and up |
| 13:40.32 | rolandow | yes |
| 13:41.28 | rolandow | but internet makes it a small world ;-) |
| 13:41.56 | n3hxs | Goedemiddag |
| 13:42.09 | rolandow | goedemiddag |
| 13:42.17 | rolandow | warm he? |
| 13:42.40 | carrar | indeed |
| 13:42.43 | n3hxs | The above was thanks to translation software. |
| 13:42.55 | rolandow | hehe :) |
| 13:42.59 | [TK]D-Fender | edwin_quijada, I see nothing in there that actually calls out and we have NO idea when you expect that dialplan code to even execute |
| 13:43.07 | [TK]D-Fender | edwin_quijada, You aren't loking at the whole process |
| 13:43.42 | [TK]D-Fender | edwin_quijada, and I've already told you that what you're looking for is a VARIABLE. NOT CDR |
| 13:44.26 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 13:44.27 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 13:45.03 | *** join/#asterisk italorossi (~Adium@201.76.151.254) |
| 13:45.37 | [TK]D-Fender | edwin_quijada, I also don't see when you'd even land on that "failed" extension in the first place so that part of the PB doesn't add up |
| 13:45.50 | edwin_quijada | [TK]D-Fender: I use callfiles. I do the callfiles from my db and use that context to manage the call |
| 13:46.04 | [TK]D-Fender | edwin_quijada, taht means nothing to what you';ve shown |
| 13:46.50 | edwin_quijada | :s |
| 13:46.56 | [TK]D-Fender | edwin_quijada, Nothing in that dialplan actually DIALS out, and I do not see how those 2 different extensions can even relate to your callfile |
| 13:47.36 | Roelt | rolandow, still no raspberry pi here.. farnell was lying when i ordered it last week.. :) |
| 13:48.21 | WIMPy | puzzled: pong |
| 13:48.32 | rolandow | Roelt: i received a mail yesterday that it is being shipped |
| 13:48.41 | rolandow | Roelt: did you receive that mail as well? |
| 13:48.44 | puzzled | WIMPy: hi, do you use CentOS? |
| 13:48.50 | WIMPy | No |
| 13:49.04 | drmessano | I got my rpi on Monday |
| 13:49.09 | drmessano | Well, two of them |
| 13:49.12 | [TK]D-Fender | edwin_quijada, And as usual we also don't see the call file itself |
| 13:49.13 | drmessano | :-D |
| 13:49.13 | rolandow | drmessano: when did you order it? |
| 13:49.34 | drmessano | Earlier this month |
| 13:49.41 | Roelt | rolandow, whut, you did? Nope, nothing.. :( order is still on processing.. (Wordt verwerkt) |
| 13:49.44 | rolandow | i ordered it last week or so .. they said it would take about 5 weeks.. and now it's being shipped out of their warehouse in a week |
| 13:49.44 | drmessano | I got mine from Newark |
| 13:49.57 | WIMPy | is kind of cured from mainstrem distros. |
| 13:50.07 | rolandow | Your Raspberry Pi has now been shipped from our warehouse. |
| 13:50.17 | Roelt | you stole my pi.. :( |
| 13:50.23 | Roelt | :D |
| 13:50.25 | rolandow | that's what the mail says.. but i still think it must be a mistake .. although i received the invoice today as well :) |
| 13:50.37 | puzzled | WIMPy: in that case never mind then :) I was looking for people interested in fixing the mISDNv2 backport to the CentOS latest 2.6.18 kernel. Karsten did the backport to 2.6.18 yesterday but it does not work on the latest CentOS 2.6.18 kernel |
| 13:50.54 | rolandow | hehe |
| 13:51.14 | WIMPy | puzzled: It wasn;t you who had the RH kernel issues a few weeks ago? |
| 13:51.21 | rolandow | do you have your sd card yet?? i just ordere 16gb class 10 for 12eu incl shipping at mycom :) |
| 13:51.28 | rolandow | which i think is a nice price.. hehe |
| 13:51.52 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 13:52.03 | puzzled | WIMPy: nope don't recall that. I did report some LCR issue a while back. Which mailing list did you see that? |
| 13:52.16 | Roelt | 9 euro at my local distributer.. :) |
| 13:52.26 | rolandow | ex vat i assume? |
| 13:52.43 | WIMPy | puzzled: It was here in the channel. Someone tried to get mISDN working for two fuill days without success. |
| 13:52.58 | rolandow | but yes that's better price even :) |
| 13:53.02 | rolandow | 30mb or 20 ? |
| 13:53.06 | datruth | Is there an how to on how to get a basic IP based configuration installed? I have asterisk installed I configured the http/mananger/asterisk.conf but still kinda confused as to how I activate everything |
| 13:53.06 | Roelt | ex-vat indeed |
| 13:53.12 | WIMPy | Then he replaced te -RH kernel with a vanilla one from kernel.org and magically it all worked instantaneousely. |
| 13:53.34 | puzzled | WIMPy: no that wasn't me. it doesn't suprise me he did not get it to work if he was on RHEL/CentOS 5.x as it does not even compile there |
| 13:53.36 | Roelt | rolandow, in saits 10 MB/s.. |
| 13:53.40 | *** join/#asterisk italorossi (~Adium@201.76.151.254) |
| 13:53.48 | puzzled | WIMPy: yes vanilla would work |
| 13:53.49 | Roelt | takeMS |
| 13:54.02 | rolandow | Roelt: ah ok .. i have class 10 at 20 mb/sec |
| 13:54.07 | WIMPy | puzzled: He did have a recent kernel. He tried different 3.x versions. |
| 13:54.21 | rolandow | so that could explain your lower price :) |
| 13:54.45 | puzzled | WIMPy: Karsten mentioned that latest 3 has all his stuff so it should work |
| 13:55.28 | WIMPy | puzzled: The vanilla kernels have always worked. Can't comment on 3.5, yet. |
| 13:56.04 | WIMPy | puzzled: In production I have never compiled the standalone mISDN, alsways took the kernel version. |
| 13:56.28 | WIMPy | thinks there should be some stable tarballs available. |
| 13:57.02 | puzzled | WIMPy: afaik 2.6.18 does not have mISDNv2 kernel modules, right? Or did I go blind and stupid? :) |
| 13:57.09 | *** part/#asterisk vanguarde9 (~kica@dash.cnsk.sk) |
| 13:57.19 | WIMPy | From 2.6.26 IIRC. |
| 13:57.36 | WIMPy | 2.6.27 |
| 13:58.20 | *** join/#asterisk sustav (~vpp@nat/digium/x-zvyowylywwzeabyh) |
| 13:58.48 | puzzled | WIMPy: phew :) |
| 14:00.07 | *** join/#asterisk edwin_quijada (~macaruchi@190.122.124.128) |
| 14:01.05 | *** join/#asterisk nighty- (~nighty@static-68-179-124-161.ptr.terago.net) |
| 14:01.33 | *** join/#asterisk SuperNull (~FreeManof@24-148-101-238.ip.mhcable.com) |
| 14:01.36 | SuperNull | BADGERS! |
| 14:01.39 | SuperNull | thats why. |
| 14:02.26 | drmessano | Was the question, why are we using memes from 2002 here in 2005? |
| 14:03.12 | *** join/#asterisk amessina (~amessina@2001:470:1f11:a4:d6be:d9ff:fe8d:7c1e) |
| 14:04.28 | SuperNull | it may have been .. |
| 14:04.35 | SuperNull | 2005 .. thats future man talk |
| 14:05.18 | Roelt | ring ring ring ring ring, bananaphone |
| 14:06.17 | *** join/#asterisk italorossi (~Adium@201.76.151.254) |
| 14:07.33 | datruth | Perhaps not |
| 14:07.34 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) |
| 14:08.55 | *** join/#asterisk italorossi (~Adium@201.76.151.254) |
| 14:10.08 | drmessano | Bananaphone? OMG, please let that one die too |
| 14:14.54 | *** join/#asterisk din3sh (~din3sh@41.136.85.105) |
| 14:15.09 | *** join/#asterisk zerohalo (~zerohalo@74-94-141-204-ma.hfc.comcastbusiness.net) |
| 14:15.14 | *** join/#asterisk bullium (~wbradshaw@216.68.250.27) |
| 14:15.30 | *** join/#asterisk eject_ck (~eject@109.86.138.135) |
| 14:16.03 | din3sh | in app_meetme.c (ast 1.8) i see options like - *81 Roll call - *82 Eject all non-admins, how do we enable them? |
| 14:18.05 | *** join/#asterisk Azrael808 (~peter@cpc17-walt12-2-0-cust657.13-2.cable.virginmedia.com) |
| 14:18.11 | zorzar | hey, i'm confused about asterisk's behaviour regarding T.38 in perticular values set in T38FaxMaxDatagram. i have two devices that set T38FaxMaxDatagram to 244 and asterisk sets it to 960. shouldn't asterisk use the common minimum? |
| 14:26.30 | [TK]D-Fender | din3sh, dial them |
| 14:27.16 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 14:27.16 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 14:27.19 | *** join/#asterisk vlad_starkov (~vlad_star@wn1nat5.beelinegprs.ru) |
| 14:31.51 | SuperNull | Ring Ring Ring Ring Ring ... Nanner phone. RING RING RING RING RING DERP PHONE |
| 14:33.48 | *** join/#asterisk darksk1ez (~mhb@fsf/member/darkskiez) |
| 14:35.07 | *** join/#asterisk italorossi (~Adium@201.76.151.254) |
| 14:36.24 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 14:36.44 | *** join/#asterisk italorossi (~Adium@201.76.151.254) |
| 14:38.36 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 14:39.23 | *** join/#asterisk italorossi (~Adium@201.76.151.254) |
| 14:41.37 | *** join/#asterisk vanguarde9 (~kica@dash.cnsk.sk) |
| 14:41.59 | *** join/#asterisk italorossi (~Adium@201.76.151.254) |
| 14:42.42 | vanguarde9 | I have question about computing of RTCP jitter in asterisk |
| 14:44.10 | vanguarde9 | http://pastebin.com/ia6zhLA6 |
| 14:44.24 | *** join/#asterisk italorossi (~Adium@201.76.151.254) |
| 14:44.45 | vanguarde9 | I dont understand why there is following line |
| 14:44.54 | vanguarde9 | rtcpheader[5] = htonl((unsigned int)(rtp->rxjitter * rate)); |
| 14:45.29 | vanguarde9 | Why is tp->rxjitter multiplied by rate ? |
| 14:46.01 | *** join/#asterisk italorossi (~Adium@201.76.151.254) |
| 14:46.24 | vanguarde9 | does somebody know who is author of res_rtp_asterisk.c ? |
| 14:46.50 | vanguarde9 | or who is the person I can ask about this code ? |
| 14:47.07 | WIMPy | That question may be more appropriate in #asterisk-dev. Or the last two. |
| 14:47.10 | leifmadsen | asterisk-dev mailing list if you have specific code questions |
| 14:47.27 | vanguarde9 | thank you i wil try it |
| 14:51.10 | Katty | weeeee what a crazy morning |
| 14:51.16 | Katty | also, hi ^_^ |
| 14:52.28 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 14:53.44 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 14:54.49 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 14:57.57 | rolandow | hi! |
| 14:58.40 | rolandow | i just had somebody who unplugged my dect base station |
| 14:58.55 | rolandow | what's this wire for?? ahhhh probably not necessary |
| 14:58.58 | rolandow | *doh* |
| 14:59.19 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 15:02.40 | *** join/#asterisk italorossi (~Adium@189.124.200.92) |
| 15:03.52 | Katty | what's this wire for? *checks CAH deck* |
| 15:04.06 | *** join/#asterisk amessina (~amessina@2001:470:1f11:a4:d6be:d9ff:fe8d:7c1e) |
| 15:04.30 | Katty | Natalie Portman |
| 15:05.50 | pabelanger | yes please? |
| 15:06.17 | Katty | :> |
| 15:06.38 | vastina | morning Katty |
| 15:06.58 | Katty | ohai, how'rechu dear? |
| 15:07.24 | vastina | is not sure he read that right... |
| 15:07.33 | vastina | gets more coffee, must be the sleepiness |
| 15:07.49 | Katty | hee |
| 15:09.05 | *** join/#asterisk asteriskATmarmuD (~Mundt@193.158.65.23) |
| 15:09.18 | vastina | hell yes, donuts in breakroom |
| 15:09.30 | vastina | i'm good katty, and you? |
| 15:10.40 | Katty | doing super duper :> all caught up on tickets, so i'm knittin! |
| 15:11.01 | *** join/#asterisk bn-7bc (~bjarne-im@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 15:11.15 | [TK]D-Fender | kitten knittin! |
| 15:11.33 | Katty | more like stitchen n bitchen. |
| 15:11.36 | Katty | bitchin |
| 15:11.41 | vastina | bitchen! |
| 15:11.45 | vastina | :0 |
| 15:13.14 | Katty | 38 rows left on the tardis project. |
| 15:15.28 | *** part/#asterisk asteriskATmarmuD (~Mundt@193.158.65.23) |
| 15:20.20 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 15:21.26 | *** join/#asterisk Bullmoose (~Bullmoose@71-33-18-80.bois.qwest.net) |
| 15:22.06 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-zhxyvwktypyivmwf) |
| 15:22.06 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 15:35.25 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-ujfhfnwywcrtbdqp) |
| 15:35.25 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 15:36.40 | *** part/#asterisk vanguarde9 (~kica@dash.cnsk.sk) |
| 15:40.44 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-yhapfhxjmcezgmso) |
| 15:40.44 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 15:40.45 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 15:44.10 | *** join/#asterisk Leddy (leddy@krypton.evosurge.com) |
| 15:44.57 | jaytee | In SIP Session Timers what's the difference between UAC and UAS? |
| 15:45.14 | Qwell | client vs server |
| 15:48.37 | jaytee | Qwell, thanks. If I'm trying to prevent "hung" calls with Flowroute then I need to use Originate for session-timers and the sample configs say UAS is the default but I'm not sure whether to use UAS or UAC. If not properly disconnected calls through Flowroute can last a maximum of six hours and they won't adjust their settings obviously. |
| 15:49.04 | Qwell | why not just enable rtptimeout? |
| 15:49.26 | jaytee | hmmm, that's a thought |
| 15:50.38 | *** join/#asterisk brdude_ (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 15:50.53 | [TK]D-Fender | WHY NOT ZOIDBERG? |
| 15:50.55 | din3sh | [TK]D-Fender: dialing didnt work, anyway i figured it works on admin/moderator mode |
| 15:51.13 | [TK]D-Fender | din3sh, that makes no sense |
| 15:51.22 | [TK]D-Fender | din3sh, And you're showing nothing. |
| 15:51.23 | [TK]D-Fender | STILL |
| 15:53.18 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-cxbrnlhvbnfsktoa) |
| 15:53.18 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 15:53.21 | datruth | I have a asterisk's system setup and I have it where sip users can call each other and voicemail but how can I do conference calling? |
| 15:53.58 | Qwell | datruth: What version of Asterisk? |
| 15:54.15 | datruth | Qwell: 1.6 |
| 15:54.54 | Qwell | meh |
| 15:54.58 | Qwell | ~upgrade asterisk |
| 15:54.58 | infobot | Before requesting assistance, you should be running the latest version of a supported release branch. See the channel topic for the latest versions available in currently supported branches. |
| 15:55.02 | Qwell | but, you'd use app_meetme |
| 15:55.10 | Qwell | or app_confbridge, with a sane version of Asterisk |
| 15:55.19 | Qwell | ~book |
| 15:55.19 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 15:55.24 | Qwell | $10 says both are documented in the book. |
| 15:55.39 | drmessano | confbridge is the way to go |
| 15:55.44 | drmessano | Upgrade to 10 and be happy |
| 15:55.58 | *** join/#asterisk gbit (~no@unaffiliated/gbit) |
| 15:56.35 | [TK]D-Fender | datruth, "core show application meetme" <- |
| 15:57.15 | datruth | asterisk16-1.6.2.24_2 An Open Source PBX and telephony toolkit |
| 15:57.21 | datruth | Is what I have exactly |
| 15:57.23 | drmessano | That's very old |
| 15:57.49 | datruth | I used a ports system to install |
| 15:57.50 | [TK]D-Fender | Not THAT old |
| 15:57.52 | dax_roc | is reading that book... + freePBX |
| 15:57.55 | Qwell | [TK]D-Fender: pretty old |
| 15:57.59 | [TK]D-Fender | datruth, Anyway you lready have your answer |
| 15:58.07 | Qwell | old enough that nobody cares anymore |
| 15:58.13 | gbit | Hello, what technology should I use to get wireless continuous signal between AP's to avoid broken calls when moving around the area? |
| 15:58.14 | drmessano | It's obsolete and unsupported. |
| 15:58.19 | drmessano | and old |
| 15:58.30 | datruth | I guess I will have to compile from the source then :( |
| 15:58.52 | drmessano | If $version < 1.8 then up_grade |
| 15:59.23 | [TK]D-Fender | datruth, "core show application meetme" <- |
| 15:59.53 | drmessano | gbit: A working implementation of WDS |
| 16:00.02 | datruth | one sec |
| 16:00.23 | gbit | drmessano, wds + ap mode? |
| 16:00.45 | drmessano | Yes |
| 16:01.26 | gbit | drmessano, this will not get disconnected with I move myself around when talking? |
| 16:01.32 | datruth | e show application meetme |
| 16:01.33 | datruth | Your application(s) is (are) not registered |
| 16:01.33 | datruth | Command 'core show application meetme' failed. |
| 16:01.45 | drmessano | gbit: I saw your original question, and I answered based on that question |
| 16:01.59 | drmessano | [11:58] <gbit> Hello, what technology should I use to get wireless continuous signal between AP's to avoid broken calls when moving around the area? <-- WDS |
| 16:02.03 | gbit | drmessano, thank you, I will try that |
| 16:03.58 | datruth | damn back to the drawing board |
| 16:04.59 | drmessano | datruth: You need DAHDI installed for meetme to work. If it were me, I would upgrade to Asterisk 10 and use confbridge |
| 16:05.16 | [TK]D-Fender | datruth, You clearly didn't install DAHDI first |
| 16:06.20 | *** join/#asterisk jonnyATroot (~jonny@90.183.82.208) |
| 16:06.22 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v031-099.mobile.uci.edu) |
| 16:06.37 | datruth | drmessano: looking for information how to install asterisk 10 |
| 16:06.44 | datruth | For FreeBSD |
| 16:06.56 | *** join/#asterisk AlfE_ (~quassel@83-215-36-12.bruck.dyn.salzburg-online.at) |
| 16:14.11 | datruth | lastest version http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-10.6.1.tar.gz |
| 16:14.14 | datruth | ? |
| 16:15.42 | *** join/#asterisk mbrit (~mbrit@186.120.97.194) |
| 16:16.30 | datruth | and I need this too http://downloads.asterisk.org/pub/telephony/dahdi-freebsd-complete/dahdi-freebsd-complete-2.2.1-rc1+2.2.1.tar.gz? |
| 16:17.23 | ChannelZ | not if you use ConfBridge |
| 16:17.48 | datruth | Where can I get confbridge? |
| 16:17.52 | ChannelZ | That was sort of the point. MeetMe uses DAHDI for mixing/timing, while ConfBridge can be all software-based |
| 16:18.00 | ChannelZ | it's a part of Asterisk |
| 16:18.22 | datruth | hrmm |
| 16:18.33 | datruth | ChannelZ: I'm trying to install asterisk 10 |
| 16:18.42 | datruth | Do I just need the two tarballs? |
| 16:19.00 | ChannelZ | I'm saying you don't (necessarily) need dahdi-freebsd-complete.... |
| 16:19.11 | datruth | http://downloads.asterisk.org/pub/telephony/dahdi-tools/dahdi-tools-2.6.1.tar.gz < -- perhaps thats what I need that one? |
| 16:19.29 | ChannelZ | no that either |
| 16:19.55 | datruth | ChannelZ: so all I need is the asterisk 10.x tarball and I'm fine? |
| 16:20.19 | ChannelZ | Unless you need DAHDI specifically (you have telephony hardware, like a Digium card) |
| 16:20.49 | datruth | No I dont have that |
| 16:21.00 | datruth | I just want to call users and have conference calling |
| 16:23.25 | ChannelZ | Then get Asterisk 10 running and use the ConfBridge app for your conferences |
| 16:25.06 | datruth | hrmm okay I think I can get asterisk going still unsure on the confbridge |
| 16:25.41 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 16:27.38 | *** join/#asterisk sustav (~vpp@nat/digium/x-qmpvwzuqazezdnqy) |
| 16:28.25 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 16:28.52 | ChannelZ | it's a dialplan application like everything else |
| 16:29.00 | ChannelZ | ~thebook |
| 16:29.01 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 16:34.15 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 16:37.26 | *** join/#asterisk jkroon (~jkroon@41.244.38.32) |
| 16:41.14 | *** join/#asterisk Azrael808 (~peter@cpc17-walt12-2-0-cust657.13-2.cable.virginmedia.com) |
| 16:43.23 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-zyjlnvukzfdbqbpn) |
| 16:43.23 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 16:46.06 | *** join/#asterisk gusto (~gusto@ppp-62-216-202-176.dynamic.mnet-online.de) |
| 16:46.36 | gusto | hey, i have set an IAX2 peer context to sth, but when i am called it goes to 'default' and not to the context the peer is in, what is that? |
| 16:46.48 | *** join/#asterisk fjfalcon (~fjfalcon@fjfalcon.ru) |
| 16:48.36 | fjfalcon | Hello all. Can someone help me with required search words. I wish to make automatic phone registration system. User call system, automatic robot says welcome, gives some options, that some more.. then more... and then gives registration code for it. So only problem.. i don't know from where to start with this target. Maybe can help me with some search words? |
| 16:49.54 | jpsharp | An IVR? Interactive Voice Response? |
| 16:51.02 | fjfalcon | jpsharp: yea, exactly.. just don't know how name that. |
| 16:51.30 | fjfalcon | jpsharp: thanks, exactly what i need. |
| 16:52.03 | *** join/#asterisk HyperNerdV2 (~HyperNerd@71-95-164-90.static.mtpk.ca.charter.com) |
| 16:54.32 | *** join/#asterisk magicdrums (~magicdrum@ec2-184-72-144-223.compute-1.amazonaws.com) |
| 16:56.34 | gusto | heh, now there is no context |
| 16:56.46 | gusto | isnt there a parameter like regcontext or something like that? |
| 16:57.29 | *** join/#asterisk navaismo (~navaismo@189.144.246.122) |
| 16:57.40 | thecardsmith | fjfalcon: a couple asterisk specific things too could be the Asterisk application "read" (which reads DTMF digits) and the application "playback" (which plays back sound files) you can make an IVR with those and a little dialplan logic |
| 17:01.41 | [TK]D-Fender | fjfalcon, here |
| 17:01.42 | [TK]D-Fender | ~book |
| 17:01.43 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 17:01.50 | [TK]D-Fender | fjfalcon, Basic IVR stuff. |
| 17:02.20 | gusto | it looks like IAX2 is trying to find out what the context is on the other side |
| 17:02.34 | [TK]D-Fender | fjfalcon, WaitExten() & Read() for your inputs, Playback & Background for your prompts. There are external TTS apps, etc if you want things more dynamic, etc |
| 17:02.46 | gusto | so maybe i somehow have to tell him what context to use when calling out that peer, right? |
| 17:05.08 | gusto | do i need to have a type=user to specify that context? |
| 17:15.32 | [TK]D-Fender | No. You could also jsut SET the context ni the peer |
| 17:17.25 | *** join/#asterisk TSM (~the_softw@78-105-6-158.zone3.bethere.co.uk) |
| 17:19.28 | gusto | [TK]D-Fender: i have that set |
| 17:21.32 | gusto | i see it in iax2 show peer <mypeer> that 'context: <context>' is set |
| 17:21.44 | gusto | on both sides |
| 17:21.48 | gusto | however, it ignores it |
| 17:22.06 | Katty | hello my asterisk does not work at all how to fix plz |
| 17:22.08 | [TK]D-Fender | gusto, and if they set it in the dial? |
| 17:22.11 | vastina | I *should* be able to run two digium PRI cards in a single system, correct me if I'm wrong please? |
| 17:22.12 | datruth | my install failed due to running out of space |
| 17:22.13 | datruth | ;/ |
| 17:22.22 | gusto | [TK]D-Fender: i tried that too, does not hel |
| 17:22.23 | gusto | help |
| 17:22.33 | datruth | Katty: you will have to give more infor than that to get help. |
| 17:22.43 | [TK]D-Fender | gusto, You should probably be SHOWING us the problem and its backup |
| 17:22.49 | gusto | [TK]D-Fender: and there is no "they" there is just me |
| 17:23.00 | navaismo | gusto have you tried IAX2/trunkname/EXTEN@context? |
| 17:23.01 | Katty | datruth: ??? |
| 17:23.02 | [TK]D-Fender | they = sending side |
| 17:23.13 | gusto | navaismo: yes, exactly, does not help |
| 17:23.21 | Katty | hello my asterisk does not work at all how to fix??? answer plz. |
| 17:23.36 | gusto | navaismo: it does then say that there is a try with <number>@context, but the call is being still rejected |
| 17:23.47 | Katty | leifmadsen: how to fix asterisk plz??????? |
| 17:23.49 | navaismo | Katty, sarcasm? |
| 17:23.52 | dwayne | offers Katty oatmeal with honey, cinnamon, raisins, and milk |
| 17:23.57 | Katty | ! |
| 17:24.00 | leifmadsen | step 1) be awesome |
| 17:24.02 | leifmadsen | step 3) profit! |
| 17:24.05 | Katty | gets complete distracted, noms oatmeals. |
| 17:24.08 | ChannelZ | gusto: Are you sure it's matching the right peer? |
| 17:24.21 | Katty | leifmadsen: my step 2? KNIT! |
| 17:24.32 | gusto | i found the problem here http://www.voip-info.org/wiki/view/Asterisk+config+iax.conf as well as "Rejected connect attempt from 213.1.2.3, who was trying to reach '987654321@'" |
| 17:24.35 | vastina | Katty: do you run digium boards miss? |
| 17:24.49 | gusto | ChannelZ: i am looking at both of them |
| 17:25.56 | Katty | vastina: no, i run sangoma boards. |
| 17:25.59 | navaismo | gusto, can you show us your cli output and the iax debug for that call? |
| 17:26.00 | Katty | vastina: tho i do have a few digiums. |
| 17:26.00 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 17:26.00 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 17:26.59 | vastina | Katty: just wondering... you should be able to run more than one digium board on a single system, am i wrong? I'm hanging in startup on the dmesg of the second module being loaded |
| 17:27.15 | Qwell | vastina: 2 should work fine |
| 17:27.18 | vastina | doing a single span and a dual span for the first time |
| 17:27.36 | vastina | Qwell: that's what I thought, but it's hanging on the second module in dmesg |
| 17:27.37 | Katty | i've have 3 cards in one. |
| 17:27.41 | Katty | they all take analog lines tho. |
| 17:27.57 | Katty | vastina: and what happens if you pop the first out, and leave the second on in? |
| 17:27.59 | vastina | hmm, these are PRI boards |
| 17:28.09 | vastina | Katty: starts up fine |
| 17:28.18 | vastina | single span or dual span |
| 17:28.21 | Katty | each boot fine, solo? |
| 17:28.24 | vastina | yeah |
| 17:28.54 | ChannelZ | PCI unhappiness in that particular machine |
| 17:29.04 | *** join/#asterisk learath (47f6db23@gateway/web/freenode/ip.71.246.219.35) |
| 17:29.05 | vastina | hmm, guess I'll take this thing apart and try something else, just wanted to confirm with you all that I can run more than one PRI board in a system |
| 17:29.12 | vastina | ChannelZ: agreed. |
| 17:29.19 | *** join/#asterisk sustav (~vpp@nat/digium/x-zgfrsmjopuqwqcdr) |
| 17:29.54 | ChannelZ | Any additional slots to try? The two you're using might be shared and causing problems |
| 17:30.03 | ChannelZ | (is it PCI or PCIe?) |
| 17:30.04 | gusto | so here is the output on the cli with 'iax2 set debug on' http://pastebin.com/GypmkA4X |
| 17:30.28 | Katty | hmm, yeah |
| 17:30.34 | Katty | thinks |
| 17:30.43 | Katty | shared interrupts or something |
| 17:30.48 | ChannelZ | gusto: so it's not matching your peer it seems |
| 17:30.53 | Katty | would make it spew all over itself, mayhaps |
| 17:31.16 | gusto | the only way i ve seen this working is when i add an additional [bhsbhs] type=user and then it comes in as unauthenticated |
| 17:31.38 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-ehapcvxcwtljrfkv) |
| 17:31.38 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 17:31.48 | gusto | ChannelZ: yes, looks like it |
| 17:31.53 | vastina | well, I'll go through the BIOS before I crack this open and ensure ther IRQ's are distinct |
| 17:31.58 | gusto | ChannelZ: so what do we do about it? |
| 17:32.04 | vastina | this is a bit odd |
| 17:32.44 | [TK]D-Fender | user VS peer does not make a difference |
| 17:32.50 | [TK]D-Fender | if it matches it matches |
| 17:32.59 | navaismo | gusto, the default guest iax2 peer is disabled? |
| 17:33.04 | gusto | [TK]D-Fender: well, to me it makes a big difference ;-) |
| 17:33.12 | ChannelZ | I'm not even sure how iax matches, is it by IP like SIP |
| 17:33.14 | gusto | navaismo: yes, i removed that shit |
| 17:33.52 | ChannelZ | Or I should say if it matches by IP depending on the type like SIP |
| 17:33.53 | gusto | ChannelZ: by ip would work, because registrying works |
| 17:34.39 | ChannelZ | Well I can tell you how mine is setup from work<->home |
| 17:35.23 | ChannelZ | The peer on my home system to call work has username=farty; the peer on my work machine is called [farty] |
| 17:35.57 | ChannelZ | And vice-versa. But they also know each other's IP statically |
| 17:36.18 | gusto | but that is SIP, right? |
| 17:36.23 | ChannelZ | No, IAX |
| 17:37.01 | ChannelZ | Not sure if this is 100% true (you never know with voip-info) but this seems to mirror my experience: http://www.voip-info.org/wiki/view/Asterisk+IAX+authentication |
| 17:37.04 | gusto | to me it's the same but w/o the username= set, because username= does not work to me |
| 17:37.32 | gusto | voip-info says that i have to create another type=user - and that works to me |
| 17:39.08 | gusto | but here everyone is telling me that it is not right, and that i need only the peer section, but that does not work |
| 17:39.32 | gusto | of course - registrying works, but what good is that when you can not call through that |
| 17:45.04 | gusto | but the ~book says it as well |
| 17:45.43 | gusto | you have there that 'in iax.conf the meaning is much simpler - it only has to do with the direction of the phone calles |
| 17:45.49 | gusto | so there is no matching in IAX |
| 17:46.07 | gusto | well, at least we solved that |
| 17:48.46 | *** join/#asterisk jsjc (~Adium@27.Red-83-59-180.dynamicIP.rima-tde.net) |
| 17:53.14 | *** join/#asterisk [ctrl][alt][del] (~Jabber@70.116.68.114) |
| 17:53.17 | *** join/#asterisk cecillincher (~clincher@ool-457d39d6.dyn.optonline.net) |
| 17:53.45 | datruth | How can I fix my voicemail issue? http://dpaste.com/775842/ |
| 17:56.21 | doolittlework | datruth: the problem is stairing you right in the face |
| 17:56.27 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 17:58.15 | doolittlework | datruth: i think you links for your call records(cdr's) is also wrong in you cdr.conf it says /pbx/var/log/asterisk//cdr-csv//Master.csv must be /pbx/var/log/asterisk/cdr-csv/Master.csv |
| 17:59.28 | datruth | so my links are messed ip? |
| 17:59.33 | doolittlework | datruth: Voicemail--> No entry in voicemail config file for '200' |
| 17:59.36 | navaismo | WARNING[101230]: app_voicemail.c:5652 leave_voicemail: No entry in voicemail config file for '200' |
| 17:59.38 | navaismo | to late |
| 18:00.11 | doolittlework | lol |
| 18:00.32 | doolittlework | i just was just saying why have logs if you do not read them |
| 18:00.53 | datruth | doolittlework: I have an entry |
| 18:01.11 | doolittlework | datruth: in voicemail.conf? |
| 18:01.45 | datruth | voicemail.conf http://dpaste.com/775843/ |
| 18:02.18 | navaismo | and you reload the voicemail config? |
| 18:02.24 | [TK]D-Fender | datruth, "ls -la /etc/asterisk" <- |
| 18:02.41 | datruth | There is nothing in pbx/var/log/asterisk/cdr-csv |
| 18:02.55 | datruth | navaismo: I reloaded asterisk yes |
| 18:03.21 | datruth | [TK]D-Fender: dont think I have a /etc/asterisk |
| 18:03.22 | navaismo | what shows the command: "voicemail show users for default" |
| 18:03.57 | [TK]D-Fender | datruth, dump the whole folder of wherever they are including the command called |
| 18:04.28 | datruth | navaismo: nothing ;/ |
| 18:04.42 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 18:04.42 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 18:05.43 | navaismo | voicemail reload and then again the: voicemail show users for default |
| 18:06.30 | navaismo | hmm what version of asterisk? |
| 18:07.03 | gusto | btw. is there a possibility to make that md5 challenge a bit longer? |
| 18:07.16 | datruth | now it doesn't want to respond to the console |
| 18:07.24 | datruth | navaismo: I just installed 10.6.1 |
| 18:07.25 | *** join/#asterisk learath_ (47f6db23@gateway/web/freenode/ip.71.246.219.35) |
| 18:08.33 | navaismo | maybe you have a dead lock |
| 18:08.54 | datruth | its back |
| 18:09.06 | datruth | navaismo: voicemail show users shows the mailboxes now |
| 18:09.12 | datruth | but now voicemail show users default |
| 18:10.37 | navaismo | and if you call now you can leave a oicemail? |
| 18:10.49 | datruth | trying |
| 18:10.50 | gusto | but the good news is, that it works now all like i wanted |
| 18:10.54 | datruth | x-lite not registering ;/ |
| 18:11.04 | gusto | where? |
| 18:11.29 | datruth | on my pbx system |
| 18:11.50 | navaismo | datruth, maybe you need to compile again asterisk with DEBUG_THREADS |
| 18:12.03 | navaismo | an when it happen do in the cli a core show locks |
| 18:13.22 | navaismo | and* |
| 18:14.07 | datruth | now the console is lagged |
| 18:14.12 | datruth | not responding to commands ;/ |
| 18:14.50 | datruth | core show locks is invalid |
| 18:15.00 | datruth | I guess I need to recompile |
| 18:17.10 | datruth | [Jul 26 14:16:23] WARNING[101230]: app_voicemail.c:1697 create_dirpath: ast_mkdir '/pbx/var/spool/asterisk/voicemail/default/100/tmp' failed: Permission denied |
| 18:17.18 | datruth | I need to manually create the mailboxes? |
| 18:19.05 | navaismo | and we back to the [TK]D-Fender question "ls -la /etc/asterisk" and related directories to asterisk |
| 18:19.44 | [TK]D-Fender | datruth, No, you're busted the pERMISSIONS of various folders and * can't WRITE to them |
| 18:19.48 | *** join/#asterisk italorossi (~Adium@189.124.200.92) |
| 18:19.57 | [TK]D-Fender | datruth, you've* |
| 18:20.43 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 18:21.36 | datruth | http://dpaste.com/775847/ new voicemail eerrors Im getting. |
| 18:21.44 | datruth | [TK]D-Fender: what should the permissions be? |
| 18:21.54 | datruth | oh |
| 18:21.57 | [TK]D-Fender | Accessible to the user you are running * as |
| 18:21.59 | datruth | one sec |
| 18:22.35 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 18:23.07 | navaismo | exist /var/spool/asterisk/? |
| 18:23.16 | datruth | directory listing http://dpaste.com/775848/ |
| 18:23.42 | datruth | navaismo: yes also when I was trying to leave a voicemail I heard no voicemail greeting or nothing |
| 18:23.42 | [TK]D-Fender | Asterisk should not be run as ROOT |
| 18:23.54 | datruth | :( |
| 18:24.11 | [TK]D-Fender | And I'm betting .. it ISN'T, and those folders are owned by it |
| 18:24.20 | [TK]D-Fender | "ports" can DIAF |
| 18:24.32 | datruth | I installed it as root from the source |
| 18:24.44 | [TK]D-Fender | ~asterisk-non-root |
| 18:24.45 | infobot | [~asterisk-non-root] Running * as non-root is covered in chapter 13 of Asterisk:The Future of Telephony 2nd Edition (~book) and an article written by Leif Madsen at http://www.taug.ca/node/115 , and for Debian : http://www.thinkdebian.org/archives/828 |
| 18:24.49 | [TK]D-Fender | It's also in the book |
| 18:24.54 | [TK]D-Fender | ~book |
| 18:24.54 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 18:24.57 | [TK]D-Fender | 3rd ed ^ |
| 18:25.08 | [TK]D-Fender | And fix all your perms |
| 18:25.15 | navaismo | ~buybook |
| 18:25.15 | infobot | You can buy "Asterisk: The Definitive Guide" at http://oreilly.com/catalog/9780596517342 so go buy it SERIOUSLY |
| 18:25.24 | *** join/#asterisk Tim_Toady (~fuzzy@188.4.44.30.dsl.dyn.forthnet.gr) |
| 18:25.52 | datruth | I have the book I guess I need to read the entire thing even for stuff I'm not going to use and hopefully figure out what mhy issues are. |
| 18:25.55 | datruth | Thanks for the info folks. |
| 18:27.56 | datruth | am I supposed to hear a vm greeting? |
| 18:27.59 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 18:28.09 | navaismo | yep |
| 18:29.27 | datruth | and if I dont how can I debug that? |
| 18:29.37 | datruth | I sit behind a firewall / nat |
| 18:29.51 | navaismo | rtp ports |
| 18:30.01 | navaismo | and nat setting for sip |
| 18:30.47 | datruth | rtp ports? and I have the nat settings already |
| 18:32.34 | datruth | nevermind I guess I gotta punch a bunch of holes in my firewall |
| 18:34.04 | *** join/#asterisk leandroaugusto (~leandroau@189.90.54.130) |
| 18:35.44 | *** join/#asterisk din3sh (~din3sh@41.136.82.21) |
| 18:36.31 | drmessano | Why dont you just configure your firewall |
| 18:36.52 | drmessano | Punch holes probably isnt necessary unless there is a ventilation issue, in which case, you probably want to relocate it anyway |
| 18:38.26 | datruth | drmessano: punching holes in the firewall is configuring the firewall |
| 18:39.28 | drmessano | I guess humor is completely lost on you |
| 18:39.40 | *** join/#asterisk sustav_ (~vpp@nat/digium/x-dlkkxptvdzlixnuc) |
| 18:40.40 | datruth | drmessano: nah its just irc so that is expected. |
| 18:40.47 | Roelt | probably the heat. :) |
| 18:41.10 | drmessano | lol |
| 18:41.14 | drmessano | Yeah, the heat |
| 18:43.00 | datruth | hrmm even with the ports allowed still no sound |
| 18:43.01 | datruth | ;/ |
| 18:43.50 | drmessano | you opened the rtp ports and sip port? |
| 18:43.51 | drmessano | UDP? |
| 18:45.17 | datruth | yup |
| 18:46.25 | datruth | maybe I need to redirect the ports |
| 18:53.53 | datruth | w00t its working!! |
| 18:54.04 | datruth | time to see if I can get this to actually mail me the .wav file |
| 18:54.10 | datruth | Do i need a mta for that? |
| 18:55.24 | *** join/#asterisk paulius (~paulius@unaffiliated/paulius) |
| 18:56.25 | paulius | Hello. I've got a quick question about timing. I have heard that the conference room and IAX need a timing source. But does SIP (and RTP) need timing? |
| 18:56.44 | paulius | I'm trying to determine which protocol would be better in a virtualized environment. |
| 18:57.09 | [TK]D-Fender | IAX trunk mode requires it. |
| 18:57.28 | [TK]D-Fender | And Meetme / Page |
| 18:57.28 | paulius | But SIP and RTP doesn't? |
| 18:57.31 | [TK]D-Fender | Correct |
| 18:57.50 | paulius | And would you know why? |
| 18:58.07 | [TK]D-Fender | because there is nothing to sync like that in SIP |
| 18:58.19 | [TK]D-Fender | IAX2 trunk mode combines multiple calls |
| 18:58.26 | [TK]D-Fender | SIP has no such construct |
| 18:58.30 | paulius | ah right, makes sense. |
| 18:58.46 | paulius | A new RTP session would be established for multiple calls/channels |
| 19:00.02 | [TK]D-Fender | correct |
| 19:00.17 | paulius | Well thanks for your help. |
| 19:04.45 | *** join/#asterisk doolittlework (~doolittle@41-134-22-14.dsl.mweb.co.za) |
| 19:17.41 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 19:23.34 | *** part/#asterisk doolittlework (~doolittle@41-134-22-14.dsl.mweb.co.za) |
| 19:23.40 | *** join/#asterisk doolittlework (~doolittle@41-134-22-14.dsl.mweb.co.za) |
| 19:43.02 | *** join/#asterisk anonymouz666 (~anonymouz@189-25-81-45.user.veloxzone.com.br) |
| 19:52.06 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 19:55.43 | *** join/#asterisk mun24 (~chatzilla@216.191.50.162) |
| 19:56.15 | mun24 | does anybody know why DID incoming call fails sometimes |
| 19:56.24 | *** join/#asterisk jsjc (~Adium@27.Red-83-59-180.dynamicIP.rima-tde.net) |
| 20:00.03 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 20:00.03 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 20:00.37 | Qwell | mun24: Because it's Thursday. |
| 20:00.43 | [TK]D-Fender | mun24, At least in most places |
| 20:00.49 | Qwell | without any information about it, that's all that can be said |
| 20:01.25 | mun24 | Qwwll: what information you need? |
| 20:01.38 | mun24 | Qwell: What information do you want? |
| 20:01.47 | Qwell | anything at all would be a start |
| 20:02.35 | [TK]D-Fender | DID delivered to your server HOW? |
| 20:02.38 | mun24 | We have asterisk server whcih registers with SIP service provider for long distance |
| 20:02.38 | bullium | I've done a fresh install of CentOS 6.3 and successfully compiled and installed asterisk following the documentation on wiki.asterisk.org. I'm able to start asterisk via "service asterisk start" then verify it's running with "service asterisk status". Even though I get a PID returned I can't connect to asterisk via "asterisk -r" |
| 20:02.43 | [TK]D-Fender | Got a FAILED CALL to show us from * CLI? |
| 20:02.48 | [TK]D-Fender | PASTEBIN is your friend |
| 20:02.48 | [TK]D-Fender | ~pb |
| 20:02.49 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 20:02.50 | [TK]D-Fender | ^^^ |
| 20:03.16 | mun24 | but when we call that did number sometimes it fails |
| 20:03.28 | Qwell | and what do the Asterisk logs say? |
| 20:03.29 | bullium | oh version certified-asterisk-1.8.11-cert4 |
| 20:03.46 | mun24 | D-Fender: asterisk spits out too much log |
| 20:04.15 | mun24 | how can I narrow down incoming call |
| 20:04.16 | [TK]D-Fender | No it doesn't |
| 20:04.20 | [TK]D-Fender | It spits out "enough". |
| 20:04.24 | [TK]D-Fender | Show us |
| 20:04.44 | mun24 | how should I connect to console using asterisk -r |
| 20:04.49 | [TK]D-Fender | yes |
| 20:05.10 | mun24 | then try to make a incoming call |
| 20:05.22 | mun24 | and see the log you mean |
| 20:05.23 | ectospasm | mun24: if you want to capture it to a file, use "asterisk -r | tee /tmp/cli-capture.txt |
| 20:05.25 | ectospasm | " |
| 20:05.57 | ectospasm | then you can read through it with "less -R /tmp/cli-capture.txt" |
| 20:06.27 | ectospasm | ...assuming you've got a capable terminal emulator, the -R will interpret the ANSI escape sequences, in less. |
| 20:06.33 | ectospasm | makes it MUCH easier to read IMO |
| 20:07.58 | doolittlework | Learned something new today: DAHDI stands for "digium asterisk hardware device interface" i thought id stood for "David and Heidi(Klum) did it" |
| 20:12.35 | drmessano | DAHDI is who MUHMI sleeps with. Unless it's the MAHLMAN |
| 20:13.32 | ectospasm | DAHDI sleeps with the MAILMAN? |
| 20:14.01 | drmessano | Unless she is sleeping with the mailman. I guess I could have been more specific |
| 20:14.21 | Qwell | hey, don't judge who DAHDI does or does not sleep with |
| 20:14.33 | drmessano | But this is 2012.. let DAHDI sleep with the MAHLMAN |
| 20:14.46 | _Corey_ | lol |
| 20:14.56 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 20:14.56 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 20:14.59 | Roelt | and LibPRI is watching for the closet? |
| 20:15.25 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 20:15.25 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 20:15.28 | drmessano | As long as there is no mouse the in closet. The yell of "IAX" will startle them |
| 20:18.10 | drmessano | I am waiting for The Carrier Remote Origination Channel driver to be merged with DUNDI, so we can operate our own CLEC using CROC-DUNDI |
| 20:18.40 | Qwell | speaking of channel drivers... |
| 20:18.44 | Qwell | drmessano: Are you using chan_motif yet? |
| 20:19.11 | drmessano | I will when there's a beta. Trunk scares me |
| 20:23.29 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 20:23.29 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 20:23.44 | drmessano | http://blogs.digium.com/2012/07/19/asterisk-11-development-call-ids-for-asterisk-logs/ "The beta of Asterisk 11 is rapidly approaching" <--- LIES.. It's been a week. |
| 20:25.26 | Qwell | I could totally make a beta right nao. |
| 20:25.35 | Qwell | mjordan would probably kill me, but I could do it. |
| 20:25.41 | Qwell | don't make me do it. |
| 20:26.55 | Katty | let's not anger the mjordan |
| 20:27.04 | Katty | there would be glaring. :< |
| 20:27.25 | Qwell | oh, I anger the mjordan all the time I'm sure |
| 20:30.11 | drmessano | mv mjordan /cage/ |
| 20:30.14 | drmessano | There |
| 20:30.24 | drmessano | Wait |
| 20:30.42 | drmessano | mv mjordan /home/qwell/Cage/ |
| 20:31.25 | drmessano | too obvious |
| 20:31.28 | drmessano | mv mjordan /home/qwell/.Cage/ |
| 20:31.29 | drmessano | Better |
| 20:34.22 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 20:37.18 | *** join/#asterisk logicwrath_work (~no@74-94-239-197-Michigan.hfc.comcastbusiness.net) |
| 20:38.46 | drmessano | Asterisk 10 beta 1 was released on 7-22-11 |
| 20:38.59 | drmessano | It is now 7-26-12 |
| 20:39.06 | drmessano | Asterisk 11 is behind schedule |
| 20:39.28 | drmessano | I call for an immediate dev audit |
| 20:40.52 | mjordan | glares at Qwell |
| 20:41.08 | Qwell | it was drmessano |
| 20:41.23 | drmessano | Qwell just encourages me |
| 20:41.30 | mjordan | there's a number of people who put together patches and got them in before the deadline, and we're working with them to get their features baked in |
| 20:41.31 | drmessano | A LOT, mind you |
| 20:41.43 | mjordan | so its getting there :-) |
| 20:42.10 | drmessano | That sounds like corporate double-speak |
| 20:42.15 | drmessano | OOPS |
| 20:42.19 | drmessano | +!!!!! |
| 20:42.29 | mjordan | nah |
| 20:42.30 | drmessano | I left off the outrage |
| 20:42.40 | ectospasm | heh |
| 20:43.09 | mjordan | if it was corporate double speak, I would have said something along the lines of "all features are currently in and fully tested" |
| 20:43.22 | *** join/#asterisk fling (~fling@fsf/member/fling) |
| 20:44.16 | drmessano | or "We are WAY ahead of schedule and are working aggressively to release the best Asterisk yet!!" |
| 20:44.44 | drmessano | Which means, "it's broken, and we are tearing through code trying to figure out who did what" |
| 20:44.50 | mjordan | if you squint hard at the schedule. Those blue bars on the wiki page can get fuzzy from far enough away |
| 20:45.00 | drmessano | HA |
| 20:45.16 | drmessano | Thats not corporate, thats Microsoft |
| 20:45.38 | Qwell | the real truth is, we changed our version numbers to match the year it was released in |
| 20:45.45 | Qwell | unfortunately, we're about 2 years behind |
| 20:45.46 | mjordan | crap, we're a year behind |
| 20:46.02 | datruth | Does the codec control how loud the voicemail recording will be? |
| 20:46.02 | mjordan | Qwell: I keep on thinking that since Asterisk 10 was Asterisk 2.0, Asterisk 11 is Asterisk 3.0 |
| 20:46.19 | Qwell | mjordan: we're gonna blow everyones mind when we release Asterisk 100 next |
| 20:46.26 | mjordan | Exactly :-) |
| 20:46.35 | drmessano | Asterisk 11 should be 11.0.2600.0 |
| 20:47.51 | mjordan | datruth: check out the volgain setting |
| 20:48.10 | datruth | thx |
| 20:48.11 | mjordan | although that only applies to sending the messages |
| 20:48.18 | drmessano | I am very much looking forward to 11 |
| 20:48.30 | datruth | yeah when i get a voicemail i can barely hear it |
| 20:48.48 | drmessano | datruth: Tell them to speak up, or turn your handset volume up |
| 20:48.52 | Katty | https://sphotos-a.xx.fbcdn.net/hphotos-ash3/s720x720/552622_10100323910537857_493616274_n.jpg <- the knittin is comin along. |
| 20:49.11 | Katty | 33 rows to go! |
| 20:49.25 | drmessano | Katty: That's GREAT |
| 20:49.46 | Katty | ty :> |
| 20:50.20 | drmessano | Trying to figure out who goes from hosting content on an FTP to freakin mediafire |
| 20:50.23 | drmessano | Shoot me now |
| 20:50.25 | *** join/#asterisk mo_og (~av@dslb-188-098-098-083.pools.arcor-ip.net) |
| 20:54.26 | Katty | goes after water gun |
| 20:55.13 | drmessano | I have the download of all of my weekend shows automated, and it was working great.. until one of them moved off FTP to mediafire |
| 20:55.20 | drmessano | Now I have no idea how to grab this thing |
| 20:55.34 | drmessano | Their REST API sucks |
| 20:56.06 | mo_og | hi guys, I'm having trouble with asterisk 1:1.8.13.0 on Debian testing -- IAX won't bind to my PPP interface whatever I do |
| 20:56.33 | mo_og | even if I specify bindaddr=0.0.0.0 in iax.conf, it will only bind to the local interface |
| 20:56.58 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 20:57.22 | mo_og | I have to specify bindaddr=<address of ppp0> explicitly, then the IAX service is reachable from outside |
| 20:57.30 | mo_og | what am I doing wrong? |
| 20:58.27 | mo_og | (of course the ppp0 address changes regularly, so a static bindaddr doesn't make sense) |
| 21:07.10 | [TK]D-Fender | 0.0.0.0 = bind to all |
| 21:08.57 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 21:10.59 | drmessano | [TK]D-Fender: Before you came back: [16:56] <mo_og> even if I specify bindaddr=0.0.0.0 in iax.conf, it will only bind to the local interface |
| 21:12.14 | mo_og | [TK]D-Fender: sure, but obviously that doesn't work for my PPP interface |
| 21:13.47 | Qwell | mo_og: What did netstat say when it was bound to 0.0.0.0? |
| 21:16.13 | mo_og | Qwell: how do i find out? |
| 21:16.46 | ectospasm | mo_og: netstat -altun |
| 21:17.33 | Katty | noms apple |
| 21:18.25 | mo_og | netstat -altun says this: udp 0 0 0.0.0.0:4569 0.0.0.0:* |
| 21:23.49 | mo_og | any ideas? |
| 21:23.49 | ectospasm | mo_og: netstat -altunp (as root) will show what process is using port 4569... |
| 21:24.03 | ectospasm | ...but it's probably asterisk |
| 21:24.23 | mo_og | it says "19713/asterisk" -- no surprise there |
| 21:25.25 | mo_og | just to make things more clear: I have two network interfaces, eth0 connected to the internal network and ppp0, which is my (direct) internet connection |
| 21:25.37 | Qwell | then it's listening on all addresses |
| 21:25.49 | drmessano | What is the actual ISSUE? |
| 21:26.20 | mo_og | if I put bindaddr=0.0.0.0 in iax.conf or leave it out altogether, I can connect with an IAX client via eth0 only, not via ppp0 |
| 21:27.09 | mo_og | but if I place bindaddr=<IP of ppp0> in iax.conf, I can actually connect via ppp0 (i.e. from anywhere on the internet) |
| 21:27.21 | carrar | katty |
| 21:27.25 | mo_og | which tells me that it's not the ISP blocking port 4569 |
| 21:27.25 | carrar | needs some apple eating music? |
| 21:27.28 | carrar | http://youtu.be/YLO7tCdBVrA |
| 21:28.49 | mo_og | I could, of course, place the ppp0 IP in iax.conf, but that IP isn't static, so it wouldn't last long |
| 21:29.01 | mo_og | so -- how do I resolve this? |
| 21:29.56 | Qwell | show us debug of a failed call |
| 21:31.14 | [TK]D-Fender | Nothing is "obvious". I see no debug |
| 21:31.48 | carrar | Did you compile with --no-suck? |
| 21:32.10 | ectospasm | I get better performance with --less-suck |
| 21:33.58 | mo_og | Qwell: in "asterisk -rvvv", nothing shows up -- is that what you mean? |
| 21:34.09 | [TK]D-Fender | You nee just a little suck to ensure traction |
| 21:34.52 | datruth | I can't seem tp use volgain with the format wav in voicemail.conf? |
| 21:35.49 | ectospasm | datruth: I seem to remember that not working, but I can't remember where I saw it |
| 21:35.58 | doolittlework | where could I get some info regarding legality of asterisk call recordings, here in south africa i head a few roumors that the asterisk recordings will not stand up in court |
| 21:36.15 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-ehapcvxcwtljrfkv) |
| 21:36.41 | datruth | ;// |
| 21:36.52 | datruth | horriable volume on the voicemail msg |
| 21:37.54 | [TK]D-Fender | doolittlework: get it from your gov't |
| 21:38.40 | ectospasm | ...or your counsel |
| 21:38.46 | mo_og | I didn't compile asterisk, if that question was directed at me, it's in the Debian testing repository |
| 21:39.15 | vastina | ectospasm: be a renegade |
| 21:39.20 | ectospasm | datruth: you can use post-processing tools to increase the gain, like sox. |
| 21:40.39 | datruth | I keep getting sox failure msgs |
| 21:41.10 | ectospasm | do you have sox installed, datruth? |
| 21:41.52 | drmessano | I always ask myself that in the morning |
| 21:42.15 | datruth | ectospasm: ahh maybe thats it let me get that installed |
| 21:43.13 | mo_og | what can I do get this issue resolved> |
| 21:43.16 | mo_og | ? |
| 21:43.17 | datruth | do you usually need volgain with wav? or some sort of conversion? |
| 21:43.29 | ectospasm | datruth: sometimes. |
| 21:44.04 | datruth | ectospasm: converting wav to mp3 would that be better? |
| 21:44.21 | Qwell | anything to do with mp3 would be worse |
| 21:45.01 | ectospasm | datruth: no |
| 21:45.03 | ChannelZ | OK this is bizarre, my fax detection is only working on 2 of my 4 lines |
| 21:45.16 | ChannelZ | but they're all configured as a single group |
| 21:45.29 | ectospasm | ChannelZ: what kind of lines? |
| 21:45.31 | datruth | hrmm okay think I may have fixed it sounds better now |
| 21:45.33 | ChannelZ | POTS |
| 21:45.58 | ectospasm | try swapping the lines, see if the problem follows the lines or the ports? |
| 21:46.00 | ChannelZ | DTMF works so it's not like it can't figure out tones |
| 21:46.10 | ChannelZ | at least I think, double checking |
| 21:46.32 | mo_og | any takers on my bindaddr issue? |
| 21:46.38 | ChannelZ | yeah dtmf worked |
| 21:47.45 | Qwell | mo_og: https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 21:48.27 | ectospasm | mo_og: have you collected CLI debug, and shows what happens when an inbound call comes in ppp0? You may want to run "tcpdump -i all -w /tmp/tcpdump.pcap -s0" in parallel, just to see if the traffic is even getting to your machine... |
| 21:49.45 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 21:50.21 | *** join/#asterisk quid246 (~quid246@CPE00131078b0b5-CM000f9f7eff1e.cpe.net.cable.rogers.com) |
| 21:50.26 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 21:50.50 | autofsckk | hello everybody |
| 21:51.32 | autofsckk | i need to install asterisk on a phisical machine with an openvox card, which distro is the best for this? the more stable |
| 21:52.51 | quid246 | Hmm... I am looking for some help with IAX (Realtime). Running Asterisk 1.6.2.24 and I am using deny/permit columns in the RT database... ie. deny is set to 0.0.0.0/0.0.0.0 and permit is set to 1.2.3.4/255.255.255.255 Asterisk will refuse registrations from any other IP other than 1.2.3.4 but I am able to make calls (without registration) from other IPs using that accounts login/pass. What |
| 21:52.51 | quid246 | gives? |
| 21:53.13 | ChannelZ | argh. |
| 21:53.29 | [TK]D-Fender | quid246: show us |
| 21:54.30 | ChannelZ | Hmm. I guess I can try running fxotune when people leave.. the lines sound clean to me though |
| 21:54.33 | quid246 | TK: How? |
| 21:55.09 | ectospasm | ChannelZ: "sound" clean... your ear is a poor judge |
| 21:57.09 | ChannelZ | yeah but I'm confused what could possibly cause it to not distunguish a single 1001hz tone versus several mixed frequencies in a DTMF |
| 21:58.16 | ectospasm | could be faulty lines, FXO channels...who knows? |
| 22:01.04 | ChannelZ | Isn't there a debug that shows what * thinks it's hearing, besides dtmf debug? |
| 22:01.07 | ChannelZ | I'm losing my mind |
| 22:04.40 | *** join/#asterisk mbrit (~mbrit@186.120.97.194) |
| 22:05.02 | mo_og | OK, so I have a log file, what should I be looking for? |
| 22:05.18 | mo_og | yes, it does register connections from the IAX client, how do I tell what's wrong? |
| 22:07.02 | *** join/#asterisk alter-ego (~alter@xdsl-78-34-99-213.netcologne.de) |
| 22:07.32 | ChannelZ | hmm ok weird.. I just stopped * and restarted DAHDI, now it's working. o_O |
| 22:09.10 | ChannelZ | meh.. whatever |
| 22:11.01 | *** join/#asterisk brian98 (~brian98@188.141.15.112) |
| 22:16.31 | *** join/#asterisk Firass-z0r (~asadf@c-67-201-205-34.reshall.wwu.edu) |
| 22:20.26 | *** join/#asterisk fskrotzki (~fskrotzki@cpe-67-253-245-174.rochester.res.rr.com) |
| 22:22.34 | *** join/#asterisk gusto (~gusto@ppp-62-216-202-176.dynamic.mnet-online.de) |
| 22:28.16 | *** join/#asterisk bobb_WU (~bobb_WU@206.74.211.13) |
| 22:38.33 | *** join/#asterisk zerohalo (~zerohalo@74.61.196.236) |
| 22:50.31 | *** join/#asterisk sawgood (~sawgood@173-13-158-29-sfba.hfc.comcastbusiness.net) |
| 22:51.23 | sawgood | an incoming SIP call arrvies, but it is not processed by any context (simply get the NOTICE) that the number is not found in the 'default' context |
| 22:51.34 | sawgood | any tips for figuring out why the context was not used? |
| 22:54.09 | sawgood | Now, for the same account, outbound calls fail because the dialplan is not found in 'default' context |
| 22:58.08 | navaismo | you need to create rules for inbound and outbound calls |
| 22:58.13 | navaismo | in your default context |
| 22:58.21 | navaismo | or whatever context defined |
| 23:09.53 | sawgood | the account that is not working is setup 100% like the others on the box (except this peer is not another Asterisk box) |
| 23:10.13 | sawgood | This is the only account to not work (and it is one of the few no Asterisk solutions connecting) |
| 23:10.27 | sawgood | I guess I'll look up on the OEM site for any ASterisk config tips |
| 23:11.52 | navaismo | you need to provide, your dialplan for incoming and outgoing calls, the sip config for that peer and the cli output to help you |
| 23:11.57 | newtonr | sawgood: can you pastebin a sanitized sip.conf and extensions.conf ? |
| 23:12.16 | sawgood | yes, stand by for pastebin |
| 23:23.51 | *** join/#asterisk Bullmoose (~Bullmoose@71-33-18-80.bois.qwest.net) |
| 23:24.30 | sawgood | 2-3 more min for the pastbin (sorry) |
| 23:29.27 | *** join/#asterisk Champi (Champi@2001:6f8:319::11:102) |
| 23:34.33 | *** join/#asterisk ThinkGNU- (~ThinkGNU-@216.67.183.210) |
| 23:38.38 | sawgood | http://pastebin.com/ryFAQF9C |
| 23:38.59 | sawgood | customers A, B, and C ... work 100% ... customer D cannot send calls (customer D is not an Asterisk box) |
| 23:40.25 | sawgood | any call arriving from customer D (fails) with a NOTICE saying '4156671000' cannot be found in context 'default' |
| 23:42.06 | sawgood | I 'think' for some reason (since customer D is a non Asterisk box) ... when anything arrives to my Asterisk box from it ... it is not being 'processed' as Customer-D (even though the username/password and IP are correct on the customer D side) |
| 23:42.53 | newtonr | yeah if the call is looking in the default context, asterisk is likely not identifying it as Customer-D-SIP |
| 23:43.29 | sawgood | also, incoming calls from Asterisk to the customer D (do not make it out to Customer D) (no known channel to deliver it on) |
| 23:44.13 | sawgood | When I run sip set debug peer Customer-D-SIP .... I do get incoming frames, but only 1 or 2 for each incoming call attempt |
| 23:44.38 | sawgood | normally, there are lots of sip 'packets' for each call (Customer's A,B, and C) |
| 23:45.15 | newtonr | what kind of device or system is at Customer-D-SIP ? |
| 23:45.38 | sawgood | Cisco CUBE acting as a proxy |
| 23:46.58 | jpsharp | can you pastebin the sip debug traces? |
| 23:47.35 | sawgood | I can ... put they have IP information (Can I clean that out)? |
| 23:48.02 | jpsharp | Yeah, just change them enough to obscure them. |
| 23:48.23 | newtonr | hopefully the secrets and IPs in your pastebin are sanitized too |
| 23:48.26 | sawgood | the thing is Customer-D-SIP is gone for the day already |
| 23:48.35 | sawgood | they are all cleaned up |
| 23:49.00 | sawgood | I'll check my logs to see if the last sip debug was saved to /var/asterisk/log/full |
| 23:49.28 | newtonr | make sure its not just with sip debug turned on for that peer |
| 23:49.46 | newtonr | if asterisk doesnt identify the traffic as associated with that peer, it probably wouldnt show up |
| 23:49.57 | sawgood | I turned on sip set debug peer (IP) (and when they made a few test calls) only 2 frames of traffic arrived to my Asterisk box |
| 23:50.13 | sawgood | I am trying to see if those are saved in /var/asterisk/log/full |
| 23:50.21 | jpsharp | I'm willing to bet the Cisco box doesn't like authenitcating. |
| 23:50.35 | newtonr | yeah i would just turn on "sip set debug" and not specify the peer IP, in case you have something weird going on with the addresses |
| 23:50.57 | sawgood | I found a decent article about registering a CUBE with a Vitelity SIP trunk, but that did not help as I thought it might |
| 23:51.10 | newtonr | id bet on authentication too |
| 23:52.00 | newtonr | triple check that your peer name in brackets and secret match whats set on the Cisco |
| 23:52.17 | jpsharp | takes a WAG and say the debug trace shows the CUBE sending an INVITE, Asterisk responding with "Authentication Required", the CUBE saying "No can do", and then Asterisk trying to drop the call into [default] as an unauthenticated/unauth peer. |
| 23:52.20 | sawgood | I cannot check the CUBE side |
| 23:52.52 | newtonr | yar |
| 23:52.57 | sawgood | @newtonr: I've seen the exact behavior you mentioned before (exactly like what you wrote) ... and that was my feeling too |
| 23:53.49 | sawgood | I guess I'll work on this tomorrow when the CUBE customer-d-SIP is back at their office |
| 23:54.08 | sawgood | Thanks for your time and opinions! |
| 23:54.55 | sawgood | Also, incoming calls are dropped with a SIP error (they do not leave the box to Customer-D-SIP) |
| 23:55.23 | sawgood | maybe they have a firewall? |
| 23:55.32 | newtonr | could be |
| 23:55.53 | newtonr | you may consider running a packet trace and checking that out with wireshark to see whats going on |
| 23:57.05 | sawgood | I will ... I have an Ethernet TAP on two spots (and I can get some cool captures) ... |
| 23:58.22 | *** join/#asterisk devdvd (~Jason@c-67-163-163-124.hsd1.wv.comcast.net) |
