| 00:01.47 | *** join/#asterisk highvoltz (rogers@bling.bling.org) |
| 00:15.44 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 00:26.07 | jdoe | nevermind, I'm an idiot. |
| 00:33.10 | *** join/#asterisk eugeneoden (~goden@99-62-173-58.lightspeed.austtx.sbcglobal.net) |
| 00:33.40 | *** join/#asterisk TJ_stamm (~TJ@c-98-213-150-32.hsd1.il.comcast.net) |
| 00:41.22 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 00:44.17 | *** join/#asterisk JuStIcIa_ (~justicia@190.52.236.133) |
| 00:48.06 | *** join/#asterisk neurosys (~neurosys@c-65-34-190-58.hsd1.fl.comcast.net) |
| 00:50.45 | *** join/#asterisk nicoAMG (~nicoAMG@201.237.49.131) |
| 01:08.50 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 01:09.52 | *** join/#asterisk jblack (~jblack@71.181.209.104) |
| 01:39.15 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 01:39.48 | *** join/#asterisk Guest47796 (Smegma@unaffiliated/coil) |
| 01:41.57 | *** join/#asterisk jblack (~jblack@71.181.209.104) |
| 01:41.57 | *** join/#asterisk chazzam (~chazz@173-24-239-247.client.mchsi.com) |
| 01:41.57 | *** join/#asterisk skrusty (~skrusty@93-97-20-22.zone5.bethere.co.uk) |
| 01:41.57 | *** join/#asterisk Obeliks (obeliks@gentoo/contributor/Obeliks) |
| 01:41.57 | *** join/#asterisk nemik (~cyanact@c-67-173-76-34.hsd1.il.comcast.net) |
| 01:41.58 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 01:41.58 | *** join/#asterisk madduck (~madduck@debian/developer/madduck) |
| 01:41.58 | *** join/#asterisk heffer (~felix@fedora/heffer) |
| 01:41.58 | *** join/#asterisk niekie (~niek@CAcert/Assurer/niekie) |
| 01:41.58 | *** join/#asterisk darkskiez (~mhb@darkskiez.ipv6.darkskiez.co.uk) |
| 01:41.58 | *** join/#asterisk Woody2143 (~Woody2143@machine76.Level3.com) |
| 01:41.58 | *** join/#asterisk Iamnach0 (~Iamnacho@ip174-70-139-208.ks.ks.cox.net) |
| 01:41.58 | *** join/#asterisk zxvff (shahid@dev.hockingits.com) |
| 01:41.58 | *** join/#asterisk kaldemar (~kaldemar@unaffiliated/kaldemar) |
| 01:41.58 | *** join/#asterisk _Raptor_ (raptorblue@andariel.informatik.uni-erlangen.de) |
| 01:41.58 | *** mode/#asterisk [+o Qwell] by verne.freenode.net |
| 01:50.25 | *** join/#asterisk underdog (~whyareyou@abel.33ad.org) |
| 01:51.19 | *** join/#asterisk seanbright (~sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 01:52.05 | *** join/#asterisk coppice (~chatzilla@m121-202-50-135.smartone-vodafone.com) |
| 01:54.20 | *** join/#asterisk jetlag (~jetlag@pool-173-61-243-204.cmdnnj.east.verizon.net) |
| 01:58.24 | pigpen | I have a quick question: Is there anything "magical" in 1.6 or 1.8 to handle faxes (ie: replacement for my current iaxmodem & hylafax) ? |
| 01:58.56 | pigpen | I have looked around a bit but haven't found anything that jumps out at me. |
| 02:05.37 | *** join/#asterisk TJNII (~TJNII@207.189.199.62) |
| 02:08.56 | TJNII | Okay, I broke my phone server somehow. If I do 1,Answer() 2,MusicOnHold() I get no sound. With 1,Answer() 2,Ringing() I also get no sound. However, 1,Answer() 2,VoiceMailMain() works as expected. Any hints as to why? |
| 02:09.17 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 02:11.39 | TJNII | If I throw a playback in there (1,answer() 2,Playback()) I get sound. |
| 02:11.42 | TJNII | Curious... |
| 02:13.17 | p3nguin | Ringing() doesn't play ringing for any specific amount of time. |
| 02:13.41 | p3nguin | If you wanted to hear ringings for 4 seconds, you would need Ringing() then Wait(4). |
| 02:17.01 | TJNII | Okay, well that's not how I thought that command worked, then. |
| 02:17.08 | TJNII | I don't get why I didn't hear MOH, though. |
| 02:17.21 | p3nguin | Maybe your moh broke. |
| 02:17.30 | TJNII | Nope |
| 02:17.38 | TJNII | Local MOH extension works fine |
| 02:17.47 | p3nguin | Great, then it works perfectly and there is no problem. |
| 02:17.50 | p3nguin | NEXT! |
| 02:17.53 | jdoe | haha. |
| 02:19.56 | TJNII | Yea, reading the docs closer I see Answer(), Ringing() will do nothing of value. |
| 02:20.22 | p3nguin | Answer() will bring the call Up. |
| 02:20.48 | p3nguin | Ringing() will play ringing until the next command finishes. |
| 02:23.09 | TJNII | Right. I have 2 servers watching a line, so the primary server answers() the call immediately to keep the other from picking it up. Problem I'm having is that if I do 1,Answer() n,Dial() I get no ringing until someone answers. the m and r options to dial don't fix this. If I throw a 2,playback() in there, it works. I'd like to understand why it is needed, though. |
| 02:23.38 | TJNII | *no ringing/no sound (ringing or otherwise)/ |
| 02:24.31 | TJNII | And that 1,Answer() 2,MusicOnHold() doesn't play MOH just confuses me. I'm guessing I don't understand the details of something I'm doing. |
| 02:25.13 | p3nguin | Where's the verbose output of a failed call? |
| 02:25.58 | TJNII | Lost in the copy buffer. I'll make another. |
| 02:27.50 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 02:28.15 | TJNII | Crap what are the pastebins again... |
| 02:28.17 | TJNII | ~pastebin |
| 02:28.17 | infobot | [~pastebin] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 02:30.53 | TJNII | http://pastebin.ca/1967601 |
| 02:33.40 | p3nguin | Are you using the same music class on the other call where moh is working? |
| 02:33.45 | TJNII | Yes |
| 02:33.47 | TJNII | Default |
| 02:34.15 | TJNII | And if I do that exact thing and call it as a local extension (Not my ITSP dropping a call in) it works as expected. |
| 02:34.38 | TJNII | Curiouser and curiouser. |
| 02:35.35 | *** join/#asterisk razu (~razu@razu.data.ee) |
| 02:39.01 | TJNII | ~book |
| 02:39.01 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 02:39.44 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 02:48.40 | *** join/#asterisk trelane` (~trelane@funtoo/staff/trelane) |
| 02:50.18 | *** part/#asterisk QbY (~Kelvin@96.176.19.11) |
| 02:53.36 | *** join/#asterisk dewinda (~dewinda@123.231.85.27) |
| 02:54.41 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 03:01.17 | dewinda | I have TDM800P clone card, It worked fine previously but in new main board IRQs do not show in /proc/interrupts tried disabling all other IRQs what could be the issue |
| 03:09.51 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 03:11.13 | ChannelZ | toasted, slots disabled in bios, who knows |
| 03:14.03 | dewinda | machine is new dual core |
| 03:14.44 | dewinda | I used this card on atom board previously |
| 03:38.25 | dewinda | slot are enabled in Bios |
| 03:39.00 | dewinda | do you know how to change the PCI irq in centos |
| 03:40.03 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 03:40.45 | dewinda | hi all |
| 03:41.06 | dewinda | i need help to sort out pci irq problem |
| 03:44.30 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 03:46.14 | joako | Is there somewhere I can find a working example of how to configure odbc voicemail with mysql? |
| 03:46.56 | *** part/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 03:50.08 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-qbmftltumpcspvmy) |
| 04:10.28 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 04:26.51 | *** join/#asterisk timholum1 (~chatzilla@98-125-77-219.dyn.centurytel.net) |
| 04:28.37 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 04:32.29 | timholum1 | Does anyone know of a sip client that is entirely web based? I have tryed to set up a jain-sip applet phone with no sucsess, And if there is not any good one's out ther does anyone know of a client that I could configure the settings and then deploy to people in a single file ( exe jar ext... ) |
| 04:33.57 | timholum1 | essencialy I have a custom crm type solution that I want to embed voip into |
| 04:51.39 | *** join/#asterisk jetlag (~jetlag@pool-173-61-243-204.cmdnnj.east.verizon.net) |
| 04:59.28 | *** join/#asterisk shamelessn00b (~chatzilla@58-65-172-114.nayatel.pk) |
| 05:00.53 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 05:05.05 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 05:08.15 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 05:13.59 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 05:17.39 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
| 05:19.57 | *** join/#asterisk bbryant (~brett@c-76-26-221-79.hsd1.sc.comcast.net) |
| 05:22.58 | *** join/#asterisk sshock (~sshock@63.248.133.83) |
| 05:42.28 | *** join/#asterisk deonv (~adium@pixfirewall.itn.com.na) |
| 05:44.26 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 05:54.30 | sshock | got a little problem: calls from my voip provider (gizmo) are coming in on extension 1747####### (my phone number) |
| 05:55.12 | sshock | so then I do an exten => 1747#######,1,Goto(s,1) in my dialplan |
| 05:55.45 | sshock | but anyway, that's not the problem; the problem is the first thing I say when they call is "Press 1 for X, Press 2 for Y, ..." |
| 05:56.40 | sshock | because 1 matches the 1747#######, it waits 3 seconds before jumping to extension 1 |
| 05:57.20 | sshock | I don't want that; how can I fix it? |
| 05:58.46 | kaldemar | sshock: what do you not want? the 3 second wait? |
| 05:58.47 | sshock | oh duh, I'm so dumb; just create another context! |
| 05:58.54 | sshock | kaldemar: correct |
| 05:59.29 | kaldemar | core show function TIMEOUT |
| 06:00.29 | *** join/#asterisk ferdna (~yup@cpe-24-92-114-97.elp.res.rr.com) |
| 06:02.42 | sshock | well, I don't want to just shorten the timeout; I want it to go away completely by not having it match other stuff |
| 06:03.29 | sshock | anyway, adding another context should fix it... |
| 06:12.18 | sshock | yep, that fixed it |
| 06:14.56 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 06:23.15 | *** join/#asterisk deonv (~adium@pixfirewall.itn.com.na) |
| 06:26.49 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 06:29.32 | sshock | now I just need to get outbound calls working... |
| 06:29.42 | sshock | getting a "Failed to authenticate on INVITE" |
| 06:30.40 | *** join/#asterisk deonv (~adium@pixfirewall.itn.com.na) |
| 06:33.59 | sshock | I think the problem is that I'm telling it to dial SIP/proxy01.sipphone.com/${EXTEN} |
| 06:34.16 | *** join/#asterisk oej (~olle@static-213-115-251-100.sme.bredbandsbolaget.se) |
| 06:34.31 | sshock | but I should be telling it to dial SIP/myphone@proxy01.sipphone.com/${EXTEN} |
| 06:34.45 | sshock | or something stupid like that |
| 06:37.38 | kaldemar | the problem is that your asterisk sends wrong credentials. there are many solutions to that. see the sample sip.conf for some sample dial strings. |
| 06:38.10 | sshock | I have my credentail for my voip provider both in the sip.conf |
| 06:38.27 | sshock | both in the register => line |
| 06:38.46 | sshock | and inside the secret in the [proxy01.sipphone.com] section |
| 06:39.24 | kaldemar | the register line has nothing to do with dialing. |
| 06:40.47 | *** join/#asterisk WWGD (~WWGD@208.79.14.130) |
| 06:41.18 | sshock | oh; ok, that is good to know |
| 06:42.16 | sshock | so what are these "sample dial strings" you speak of in sip.conf? |
| 06:42.30 | *** join/#asterisk FuriousGeorge (~chatzilla@ool-4354d13c.dyn.optonline.net) |
| 06:42.35 | FuriousGeorge | hey all |
| 06:42.37 | sshock | you mean extensions.conf? |
| 06:42.56 | *** join/#asterisk SeTTleR (~bernd@p5DDECDC9.dip.t-dialin.net) |
| 06:42.59 | SeTTleR | hi |
| 06:43.17 | FuriousGeorge | im having problems registering freeswitch to asterisk... here is what i get in asterisk console: Registration from '<sip:mmdllc@anterra.gotdns.com;transport=udp>' failed for '67.83.55.85' - No matching peer found |
| 06:43.33 | sshock | these should be equivalent, right? SIP/provider/${EXTEN} and SIP/${EXTEN}@provider |
| 06:43.45 | FuriousGeorge | it must be that transport=udp is being considered part of the username, no? |
| 06:43.49 | kaldemar | sshock: if i say sample sip.conf, i mean sample sip.conf. not extensions.conf. sip.conf.sample in the source package or here under your version: http://svn.digium.com/svn/asterisk/tags/ |
| 06:44.01 | kaldemar | sshock: no |
| 06:44.36 | kaldemar | FuriousGeorge: it should now. what are you expecting to match it? |
| 06:44.48 | kaldemar | s/now/not/ |
| 06:44.56 | *** join/#asterisk bbryant (~brett@c-76-26-221-79.hsd1.sc.comcast.net) |
| 06:44.58 | FuriousGeorge | infobot: ill pb it |
| 06:45.17 | *** join/#asterisk mpe (~mpe@94.127.49.1) |
| 06:45.46 | *** join/#asterisk shamelessn00b (~chatzilla@58-65-172-114.nayatel.pk) |
| 06:45.56 | sshock | kaldemar: my sip.conf doesn't have any "sample dial strings", unless you are talking about the comments at the top like: |
| 06:45.59 | FuriousGeorge | kaldemar: i made a pb |
| 06:46.04 | sshock | ; You may also use SIP/username@domain to call any SIP user on the Internet |
| 06:46.08 | FuriousGeorge | kaldemar: i expect it to match this user in sip.conf http://pastebin.ca/1967732 |
| 06:46.27 | sshock | there's also a comment that says these should both work: SIP/proxyhostname/user or SIP/user@proxyhostname |
| 06:46.59 | sshock | kaldemar: you said those weren't equivalent but sip.conf makes it sound like either one should work |
| 06:47.36 | sshock | but anyway, I tried both ways and I'm still getting "Failed to authenticate on INVITE" |
| 06:47.42 | kaldemar | FuriousGeorge: "host=mmdllc.gotdns.com" probably makes it not match. |
| 06:48.38 | sshock | could it be the [authentication] section that I missed? I need to add auth= in there? |
| 06:48.40 | kaldemar | FuriousGeorge: if you register, you might as well define host=dynamic. |
| 06:49.48 | FuriousGeorge | kaldemar: changed to host=dynamic and still get Registration from '<sip:mmdllc@anterra.gotdns.com;transport=udp>' failed for '67.83.55.85' - No matching peer found |
| 06:49.53 | sshock | based on the comments in there, I thought I shouldn't need that if I already have like secret and username defined in my [proxy01.sipphone.com] section |
| 06:50.00 | *** part/#asterisk tzafrir (~tzafrir@212.179.75.202) |
| 06:50.08 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
| 06:50.58 | kaldemar | sshock: you don't, if your peer is correctly defined. |
| 06:51.31 | kaldemar | FuriousGeorge: pastebin a sip debug with verbosity |
| 06:51.35 | sshock | ok, good; because the other reason I didn't want to put that in the [authentication] section is because I don't want it accidentally sending my password to other places |
| 06:52.56 | sshock | here's what my peer looks like: http://pastebin.com/q9rjJaLe |
| 06:53.28 | *** join/#asterisk karthik (~raghu@122.183.17.90) |
| 06:54.58 | *** join/#asterisk hehol (~Adium@ip-78-94-0-76.unitymediagroup.de) |
| 06:56.00 | sshock | anyone familiar with Gizmo to know if I'm missing something in there? |
| 06:56.20 | kaldemar | sshock: what version of asterisk are you using? |
| 06:56.37 | sshock | 1.4.21.2 |
| 06:57.06 | *** join/#asterisk sawgood (~sawgood@173-13-158-26-sfba.hfc.comcastbusiness.net) |
| 06:57.48 | FuriousGeorge | kaldemar: http://pastebin.ca/1967739 |
| 06:58.00 | FuriousGeorge | kaldemar: thats the sip debug when fs tries to register to * |
| 06:58.22 | sawgood | Hi: I added an Aastra phone to an * 1.4 box (as a SIP end point) ... the Aastra phone will not register (sip show peers) and Wireshark show the * box as giving out a 401 unauthorized return |
| 06:58.36 | sawgood | I am 100% sure the secret is correct on the Aastra phone |
| 07:02.26 | kaldemar | sshock: try adding fromuser parameter |
| 07:04.41 | sshock | kaldemar: thanks; I tried that and no luck, BUT |
| 07:04.50 | sshock | then I tried adding fromdomain, and that solved it! |
| 07:04.54 | sshock | silly gizmo |
| 07:06.14 | FuriousGeorge | kaldemar: did i paste enough of the sip debug output? |
| 07:07.51 | sshock | of course, then the audio wasn't working, but I just tried it again and this time the audio did work |
| 07:08.58 | kaldemar | FuriousGeorge: how does your peer look like in sip.conf? |
| 07:09.24 | kaldemar | FuriousGeorge: oh sorry, you pasted it already... |
| 07:09.40 | FuriousGeorge | kaldemar: but i changed it to host=dynamic since then |
| 07:09.46 | FuriousGeorge | kaldemar: other than that it's the same thing |
| 07:10.30 | kaldemar | FuriousGeorge: what version of asterisk are you using? |
| 07:10.35 | *** part/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 07:10.40 | FuriousGeorge | kaldemar: 1.4.20.1 |
| 07:11.48 | sawgood | got it ... darn Aastra has both a Global SIP settings section and a section for each LINE appearance |
| 07:11.54 | sawgood | it is registered now ... yeah! |
| 07:12.49 | FuriousGeorge | kaldemar: it keeps telling me no matching peer found, but I am able to dial into that server using the peer i set up |
| 07:13.24 | FuriousGeorge | kaldemar: i just can't seem to get that server registered with *. on the FS side I get: 2010-10-20 02:53:41.922467 [ERR] sofia_reg.c:1610 mompou Registration Failed with status Not found [404]. failure #4 |
| 07:14.45 | kaldemar | FuriousGeorge: actually there is no real point in registering if the other end is static. but, try changing the type from user to peer. |
| 07:15.03 | *** join/#asterisk af_ (~getsmart@78.134.21.185) |
| 07:16.29 | FuriousGeorge | kaldemar: that did it |
| 07:19.22 | sshock | any voip providers out there that will let me set my caller id to whatever I want? |
| 07:20.08 | sshock | stupid gizmo sets my caller ID to "Goog 411" |
| 07:20.12 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 07:20.42 | b14ck | sshock, flowroute.com |
| 07:20.50 | b14ck | They're the best in my opinion (for US users). |
| 07:21.19 | sshock | b14ck: cool, thanks; what are their rates like? |
| 07:22.08 | sshock | sweet, 1c/min |
| 07:22.17 | sshock | very reasonable |
| 07:22.36 | b14ck | They're pretty cheap compared to the other carriers. |
| 07:22.43 | b14ck | And very reliable (and transparent). I like using them. |
| 07:22.47 | b14ck | I'm sure you'll have a good experience with them :) |
| 07:23.04 | sshock | and they will let me set my caller ID to whatever I want? |
| 07:23.33 | sshock | or will they just automatically set it to the DID if I've purchased one? |
| 07:25.23 | b14ck | no, you can set it to anything |
| 07:25.28 | b14ck | even variable length numbers |
| 07:25.41 | b14ck | You can set it to '5', '666', or even '11111111111111' |
| 07:25.44 | b14ck | They're extremely flexible. |
| 07:25.53 | b14ck | Their web panel is also really nice. |
| 07:26.15 | sshock | sweet |
| 07:26.31 | sshock | and a caller ID can have both a name and a number, right? |
| 07:26.40 | b14ck | yep |
| 07:26.51 | b14ck | Well, that depends on who you're sending the call to though, if it'll show. |
| 07:27.00 | b14ck | A lot of people don't get caller ID name service anymore. |
| 07:27.06 | b14ck | So it probably isn't that useful. |
| 07:27.15 | sshock | and I don't have to purchase a DID if I don't need one, right? like if I just want to use it for outgoing that's ok |
| 07:27.21 | b14ck | Right. |
| 07:27.32 | creativx | whee core dumps |
| 07:27.39 | sshock | sweet, thanks for the suggestion |
| 07:28.07 | b14ck | No problem. |
| 07:31.17 | *** join/#asterisk jkroon (~jkroon@dsl-242-8-144.telkomadsl.co.za) |
| 07:35.32 | FuriousGeorge | kaldemar: thanks, everythings working well now |
| 07:38.53 | creativx | wtf core dumps on the old asterisk box too.. what the hell |
| 07:39.53 | creativx | is it infectious?! |
| 07:41.18 | *** join/#asterisk joobie (~joobie@CPE-138-217-158-153.lns6.lon.bigpond.net.au) |
| 07:42.21 | zamba | is it possible to restart polycom apps remotely? |
| 07:42.34 | kaldemar | zamba: yes |
| 07:42.42 | zamba | kaldemar: how? |
| 07:42.46 | zamba | kaldemar: through asterisk? |
| 07:43.59 | kaldemar | zamba: iirc, through asterisk with sip notify. or you could use the web interface if it has a reboot option. it's been a while since i took a look at it. |
| 07:44.48 | zamba | kaldemar: will this work if the app has crashed? |
| 07:45.27 | *** part/#asterisk WWGD (~WWGD@208.79.14.130) |
| 07:47.07 | *** join/#asterisk waschtl (~waschtl@3ed8a58a.d.d9tcloud.de) |
| 07:48.28 | kaldemar | zamba: don't know without knowing how it crashed. so, go try it. |
| 07:50.53 | *** part/#asterisk FuriousGeorge (~chatzilla@ool-4354d13c.dyn.optonline.net) |
| 07:54.57 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 08:01.31 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 08:02.49 | *** join/#asterisk Poincare (~jefffnode@v74.ampersant.be) |
| 08:04.24 | *** join/#asterisk schmidts (~schmidts@lmlo.sil.at) |
| 08:04.27 | schmidts | good morning |
| 08:04.52 | *** join/#asterisk felipe_ (~felipe@my.nada.kth.se) |
| 08:10.08 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 08:11.18 | *** join/#asterisk hrhrhr (~c1@unaffiliated/hrhrhr) |
| 08:12.34 | *** join/#asterisk ph8 (~ph8@unaffiliated/ph8) |
| 08:13.57 | *** join/#asterisk Arnoz (~Arnoz@91.183.59.105) |
| 08:15.00 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 08:21.47 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 08:30.57 | *** part/#asterisk sshock (~sshock@63.248.133.83) |
| 08:31.27 | creativx | ahah |
| 08:31.32 | creativx | error: bad! bad! bad! |
| 08:31.36 | creativx | I like it |
| 08:34.48 | *** join/#asterisk dlirit (~lirant@80.74.100.10) |
| 08:36.20 | *** join/#asterisk FlashDeluxe (~benedict@static-87-79-94-28.netcologne.de) |
| 08:38.48 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 08:42.52 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 08:52.01 | *** part/#asterisk karthik (~raghu@122.183.17.90) |
| 09:03.13 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 09:04.35 | SeTTleR | I have a Digium TDM400P with three FXS and one FXO module. dahdi_scan says FXS FAILED for every FXS port. what could that mean? |
| 09:05.13 | SeTTleR | is that some configuration mistage? |
| 09:05.22 | SeTTleR | s/mistage/mistake/ |
| 09:09.16 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 09:11.51 | *** join/#asterisk Tim_Toady (~moi@79.103.14.105.dsl.dyn.forthnet.gr) |
| 09:16.37 | *** join/#asterisk ruyo (~psantos@a83-132-152-91.cpe.netcabo.pt) |
| 09:21.59 | creativx | uh. i cant remember, where does queues look for files specified by "musiconhold=" and/or "music=" |
| 09:23.33 | *** join/#asterisk bjornts (~Adium@it010225.klientdrift.uib.no) |
| 09:29.06 | *** join/#asterisk z4nD4R (~zandar@pc-asnis9n5kqq6tftrd9hvs487uron66k.usr.iklub.sk) |
| 09:29.15 | z4nD4R | hi guis |
| 09:31.53 | *** join/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 09:34.09 | z4nD4R | can me somebody help with sip over tls? |
| 09:39.45 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 09:39.59 | *** join/#asterisk BoTheBlade (~BoTheBlad@56.3.ppp.dianet.info) |
| 09:40.21 | *** part/#asterisk deathnote[] (~lhorace@unaffiliated/evil-sheek) |
| 09:41.24 | creativx | i broke moh, but cannot figure out why.. i get no errors (even with debug 10), just "started moh, class 'ringing'" and then immediately "stopped moh" |
| 09:42.32 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 09:47.52 | *** join/#asterisk florz (nobody@2001:1a50:503c::1) |
| 10:10.15 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 10:15.21 | *** join/#asterisk silkcut (silkut@plox.tor.hu) |
| 10:15.30 | *** join/#asterisk sgimeno (~chatzilla@163.117.211.10) |
| 10:30.32 | *** join/#asterisk guilhermebr (~Guilherme@200.103.96.98) |
| 10:31.36 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
| 10:36.23 | creativx | heh odd. moh fixed itself |
| 10:37.03 | *** join/#asterisk TobSnyder (~schneider@dslb-088-073-204-184.pools.arcor-ip.net) |
| 10:40.44 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 10:42.43 | *** join/#asterisk euphor][a (~mallchin@mail.dataproservices.co.uk) |
| 10:42.51 | euphor][a | hi guys |
| 10:47.52 | euphor][a | can someone help me debug this sip exchange please: http://pastebin.com/C98AKMX0 |
| 10:48.06 | *** join/#asterisk radic (~radic@dslb-094-216-235-117.pools.arcor-ip.net) |
| 10:48.14 | euphor][a | it looks like an incoming call? my server replies with qa 404, but I can't see why |
| 10:48.20 | euphor][a | other incoming calls work fine |
| 10:48.26 | Naikrovek | what do you mean "help with"? what's the problem |
| 10:48.40 | schmidts | <PROTECTED> |
| 10:48.52 | euphor][a | Naikrovek: I don't get any calls from them |
| 10:49.04 | Naikrovek | "don't" or "can't" |
| 10:49.09 | schmidts | <PROTECTED> |
| 10:49.11 | euphor][a | schmidts: thank you. are they trying to call me? |
| 10:49.13 | kaldemar | euphor][a: it's not a call. |
| 10:49.21 | schmidts | <PROTECTED> |
| 10:49.22 | euphor][a | kaldemar: excellent, thank you |
| 10:49.23 | radic | how is the syntax to delete "/SIP/BLIST/373/ : 1" from the DB on the CLI? |
| 10:49.51 | schmidts | radic database del SIP/BLIST/373 |
| 10:49.55 | euphor][a | Naikrovek: don't, I have asked them for a test number to test dialling from there kit but as yet don't have one |
| 10:50.04 | euphor][a | Naikrovek: I expect it all works fine, just making sure |
| 10:50.32 | euphor][a | schmidts: great, just making sure :) |
| 10:50.38 | radic | bravo*CLI> database del SIP/BLIST/373 |
| 10:50.38 | radic | Usage: database del <family> <key> |
| 10:50.38 | radic | <PROTECTED> |
| 10:50.38 | radic | family and key. |
| 10:51.00 | schmidts | ah sorry that was database show then ;) |
| 10:51.08 | schmidts | ok try database del SIP BLIST |
| 10:51.11 | kaldemar | radic: database del SIP BLIST/373 |
| 10:51.23 | euphor][a | thank you guys :) |
| 10:51.38 | radic | kaldemar: does not exist |
| 10:52.14 | kaldemar | radic: database del SIP/BLIST 373 |
| 10:52.27 | kaldemar | radic: how did you put it there? |
| 10:53.28 | radic | kaldemar: it was a mistake in the dailplan as I was setting up a blacklist-function |
| 10:56.46 | *** join/#asterisk z4nD4R (~zandar@pc-asnis9n5kqq6tftrd9hvs487uron66k.usr.iklub.sk) |
| 10:58.06 | z4nD4R | somebody can help me wtih sip over tls? |
| 10:59.33 | *** join/#asterisk Weazel (~Weazel-@213.8.83.6) |
| 11:01.15 | *** join/#asterisk Weazel (~Weazel-@213.8.83.6) |
| 11:07.14 | *** join/#asterisk ukine (~ukine@14-145.97-97.tampabay.res.rr.com) |
| 11:11.17 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 11:11.53 | *** join/#asterisk z4nD4R (~zandar@pc-asnis9n5kqq6tftrd9hvs487uron66k.usr.iklub.sk) |
| 11:18.18 | *** join/#asterisk jeremy_g (~jeremey_g@gw-sthlm01.rebtel.com) |
| 11:21.50 | *** join/#asterisk bjornts (~Adium@it010225.klientdrift.uib.no) |
| 11:28.31 | jeremy_g | if chan_zap.so is missing, why would the meetme ztdummy still work. is chan_zap what provides zap abstraction |
| 11:31.15 | *** join/#asterisk z4nD4R (~zandar@pc-asnis9n5kqq6tftrd9hvs487uron66k.usr.iklub.sk) |
| 11:31.22 | *** join/#asterisk joel_oliveira (~chatzilla@alpes.nortenet.pt) |
| 11:32.36 | joel_oliveira | hello |
| 11:32.51 | joel_oliveira | I need some help related to asterisk configuration |
| 11:32.52 | kaldemar | jeremy_g: chan_zap provides the zapata channels, ztdummy provides the timing. |
| 11:33.36 | z4nD4R | somebody to help with sip over tls? |
| 11:35.54 | joel_oliveira | I am using version 1.4.21.2 and I need to configure the number of channels that a client of type friend may use. When I define call-limit=n I get n inbound connections and n outbound connections. is it any way that I get only n and each can be used to outbund-and-inbound. i.e. if I have defined a call-limit of two the user can: make 2 calls, recieve 2 calls or recieve 1 call and make 1 call.... |
| 11:35.55 | joel_oliveira | ...I understand that this may have been discussed already. |
| 11:38.04 | kaldemar | joel_oliveira: use GROUP functions in your dialplan |
| 11:38.33 | *** join/#asterisk af_ (~getsmart@78.134.21.185) |
| 11:39.41 | jeremy_g | kaldemar:isn't zapata a must to have ztdummy timing (kernel 2.6). Can i say that without chan_zap.so, it would be impossible to run meetme even if you have ztdummy and zaptel compiled and insmoded |
| 11:40.40 | kaldemar | jeremy_g: no. |
| 11:41.44 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 11:42.26 | *** join/#asterisk TobSnyder (~schneider@dslb-088-073-204-184.pools.arcor-ip.net) |
| 11:43.17 | jeremy_g | kaldemar:enlighten me |
| 11:43.46 | kaldemar | jeremy_g: ztdummy needs zaptel, not chan_zapata. |
| 11:45.06 | jeremy_g | kaldemar:right, how does asterisk and ztdummy talk to each other. |
| 11:45.22 | jeremy_g | kaldemar:doesn't asterisk need some abstract like zapata as a middleman |
| 11:57.15 | joel_oliveira | kaldemar: do you have some kind of example on the use of GROUP functions applied to my use case? I am new to Asterisk and on the book I don't see an example that is related to mine. thanks |
| 11:58.48 | jeremy_g | kaldemar:please answer me as i read in the book where this is illustrated with a figure |
| 11:59.15 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 12:02.13 | zamba | what administration consoles exist for asterisk? |
| 12:02.22 | zamba | web-based |
| 12:02.34 | zamba | setting up endpoints, ivrs, queues and so on? |
| 12:06.48 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 12:12.14 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 12:12.46 | jeremy_g | damn it was much easier to read ztdummy.c instead of all the books |
| 12:14.26 | *** join/#asterisk jkroon (~jkroon@dsl-242-8-144.telkomadsl.co.za) |
| 12:15.17 | jeremy_g | tzafrir_laptop:i am using 1.4. i was experiencing a funny problem with my prod. server. A 2xquad core amd, with ztdummy-1.4.12.1 and asterisk 1.4.19 freezing every after every x minutes. The kernel is 2.6.18-194 so ztdummy uses rtc. |
| 12:15.44 | jkroon | can someone please take a second look at https://issues.asterisk.org/view.php?id=17823 - yes, asterisk is correct but getting GS to fix their firmware will be next to impossible, so using the provided patch will happily work around it IMHO, but I could be wrong. |
| 12:16.56 | tzafrir_laptop | jeffrey, hmmm... dahdi required asterisk >= 1.4.22 . But should work better in that sense |
| 12:17.46 | jeremy_g | tzafrir_laptop:can't upgrade due to certain policy reasons. |
| 12:18.12 | jeremy_g | tzafrir_laptop:i am thinking of upgrading the kernel to use high res timer for ztdummy. would that be a good idea |
| 12:18.29 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 12:18.38 | jeremy_g | in x minutes , x is around 80mins |
| 12:19.11 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 12:19.37 | tzafrir_laptop | thinks it's funny that you can upgrade the kernel but not Asterisk |
| 12:20.01 | creativx | the kernel is not _that_ important tzafrir_laptop |
| 12:20.08 | creativx | i mean, telephony > kernel ! |
| 12:20.44 | jeremy_g | creativx:ztdummy.c checks if kernel version is higher, then use high res timer otherwise rtc |
| 12:21.16 | tzafrir_laptop | that's easy to override. But I don't think your kernel can use hires timers |
| 12:21.30 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 12:21.57 | jeremy_g | tzafrir_laptop:no it can't. i must upgrade to kernel verison >=2.6.22 |
| 12:22.16 | jeremy_g | in order to use hi res timer. |
| 12:22.25 | tzafrir_laptop | jeremy_g, if you want to backport all the timing fixes from dahdi to zt_dummy - feel free |
| 12:22.52 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
| 12:24.23 | *** join/#asterisk c0rnoTa (~c0rnoTa@178.177.194.233) |
| 12:25.26 | jeremy_g | tzafrir_laptop:no, i ll rather upgrade. |
| 12:26.51 | jeremy_g | i would like to run this problemative dual socket quad core amd node in full debug mode. so what do i have to do to enable debugging in kernel,ztdummy. For asterisk um gonna increase verbose and debug and run with -g though its not crash. Any instrumentation advice. |
| 12:28.18 | *** part/#asterisk nextime (~nextime@unaffiliated/nextime) |
| 12:30.19 | *** join/#asterisk pc-m (~pascal@modemcable094.94-70-69.static.videotron.ca) |
| 12:31.42 | *** part/#asterisk mcargile (~mikec@office2.vicidial.com) |
| 12:32.27 | *** part/#asterisk c0rnoTa (~c0rnoTa@178.177.194.233) |
| 12:40.15 | *** join/#asterisk Gohltron (~rooms@tmo-100-146.customers.d1-online.com) |
| 12:41.22 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 12:42.46 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 12:42.52 | *** part/#asterisk Gohltron (~rooms@tmo-100-146.customers.d1-online.com) |
| 12:43.47 | *** join/#asterisk SuPrSluG (~SuPrSluG@host-64-179-106-158.ind.choiceone.net) |
| 12:48.25 | *** join/#asterisk wphillips (~wphillips@64-19-240-7.caprock.net) |
| 12:56.07 | *** join/#asterisk SeTTleR (~bernd@p5DDECDC9.dip.t-dialin.net) |
| 12:56.11 | SeTTleR | hi |
| 12:57.26 | jsgoecke | hi |
| 12:57.33 | z4nD4R | hi |
| 12:58.39 | *** join/#asterisk a1fa (~awef@unaffiliated/a1fa) |
| 13:05.25 | WIMPy | lo |
| 13:07.21 | creativx | high |
| 13:07.26 | *** join/#asterisk malaiwah (~mbelleau@host-64-47-115-5.masergy.com) |
| 13:08.36 | malaiwah | Is there any flag that can be passed to EAGI to get audio in SLINEAR16 (wideband) format on file descriptor 3 on Asterisk 1.6? I've read a bit of the source code and I can't find it; I wonder if just changing the read format from SLINEAR to SLINEAR16 in the source code would do it. |
| 13:08.57 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:08.57 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:13.17 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 13:17.46 | *** join/#asterisk theHub (~karl@69.177.93.21) |
| 13:19.34 | *** join/#asterisk neurosys (~neurosys@173-9-159-177-miami.txt.hfc.comcastbusiness.net) |
| 13:21.05 | *** join/#asterisk clintc (~clintc@n128-227-82-65.xlate.ufl.edu) |
| 13:27.33 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 13:31.29 | *** join/#asterisk erinspice (~erin@nat/digium/x-szzhbtojdwcajpdb) |
| 13:37.11 | *** join/#asterisk timeshell_atwork (~chatzilla@gw.lusi.on.ca) |
| 13:38.17 | *** part/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 13:43.25 | *** join/#asterisk Victor_Yure (~victor@unaffiliated/victoryure/x-837844) |
| 13:47.58 | drmessano | Who's your dahdi? |
| 13:49.13 | schmidts | drmessano Dr. Zap tel |
| 13:49.43 | drmessano | ~zaptel |
| 13:49.43 | infobot | from memory, zaptel is zapata telephony interface. A low level interface designed to abstract hardware access to a variety of devices for BRI, PRI or analogue access. a phone card company |
| 13:50.10 | drmessano | ~zapupgrade is <reply> Don't be a ztdummy, upgrade to DAHDI and enjoy all the newness! |
| 13:50.10 | infobot | okay, drmessano |
| 13:50.11 | russellb | dahdi is not pronounced like daddy, btw |
| 13:50.15 | jeremy_g | has anyone every used asterisk -I option. Is it necessary to use if you are using ztdummy. |
| 13:50.50 | russellb | it's not necessary, but recommended (and the default in newer versions) |
| 13:50.55 | russellb | it should have never been optional, IMO |
| 13:50.58 | drmessano | russellb, it's not d-ah-di? |
| 13:52.08 | russellb | Dah is like the "do" in "don" ... Dah rhymes with saw |
| 13:52.49 | z4nD4R | somebody to help with sip over tls? |
| 13:53.10 | drmessano | It's daw-di? |
| 13:53.12 | schmidts | russellb like dodi? or more like dawdi |
| 13:53.28 | russellb | schmidts: i say those 2 the same way ... |
| 13:53.32 | russellb | ~dahdi |
| 13:53.33 | infobot | [~dahdi] Digium/Asterisk Hardware Device Interface (DAhdi). The new name of zaptel (more info at http://www.asterisk.org/dahdi ) and is pronounced "dah-dee" with a short A, or pronounced like http://www.russellbryant.net/dahdi.wav |
| 13:53.38 | russellb | i recorded it ^^^ :-p |
| 13:53.40 | jeremy_g | russellb:thanks. my dual socket quad core asterisk server is reporting a freeze for few minutes during operation. what could be causing this? asterisk1.4.19,ztdummy-1.4.21, kernel 2.6.18-194. |
| 13:53.52 | russellb | jeremy_g: a million things could cause that |
| 13:54.07 | russellb | ancient versions of everything isn't a good start |
| 13:54.33 | drmessano | russellb, Just as I thought.. like DAHDI LAMA |
| 13:54.47 | russellb | heh, yeah |
| 13:54.49 | jeremy_g | russellb:i cant upgrade to dhadi from some reason and i have an identical software,hw configuration working perfectly fine with only one less socket i.e. the other hw is 1x4 core, while the problematic one is 2x4 core. |
| 13:55.03 | russellb | i'm sorry. |
| 13:55.11 | drmessano | russellb, so "Who's your Dahdi?" is perfectly reasonable |
| 13:55.22 | schmidts | drmessano +1 |
| 13:55.26 | drmessano | russellb, especially if you are from up north |
| 13:55.28 | russellb | i guess ... as long as you don't say "daddy" |
| 13:55.47 | jeremy_g | russellb:can i cause ztdummy to stick to one processor |
| 13:55.57 | russellb | that doesn't make any sense |
| 13:55.58 | jeremy_g | what do you calll this affinity |
| 13:56.06 | russellb | i can't help you |
| 13:56.25 | jeremy_g | russellb:OK |
| 13:58.06 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 13:58.34 | *** join/#asterisk frigidzephyr (~rnewton@nat/digium/x-zormcbzeujfewdbp) |
| 14:01.30 | *** join/#asterisk DelphiWorld (~Delphi@41.200.3.40) |
| 14:01.32 | DelphiWorld | hi all |
| 14:01.36 | DelphiWorld | i have the folowing error: |
| 14:01.39 | drmessano | jeremy_g, enough time has passed that you are going to have a VERY hard time finding someone willing to troubleshoot an issue where Zaptel is involved. The oldest supported Asterisk branch is 1.4 and releases newer than several years old support DAHDI. You're just as likely to get help asking about Windows 95 |
| 14:01.40 | DelphiWorld | [Oct 20 10:01:00] NOTICE[362]: chan_iax2.c:9011 socket_process: Rejected connect |
| 14:01.40 | DelphiWorld | attempt from 41.210.64.75, request 's@default' does not exist |
| 14:01.49 | DelphiWorld | any idea? |
| 14:02.19 | *** join/#asterisk oej (~olle@static-213-115-251-100.sme.bredbandsbolaget.se) |
| 14:02.39 | DelphiWorld | this error while calling a DID routed to asterisk |
| 14:02.44 | DelphiWorld | i am using iax2 registration |
| 14:02.44 | [TK]D-Fender | DelphiWorld: Make s@default exist, or make the requester ask for something that DOES |
| 14:04.23 | anonymouz666 | anyone know why wireshark shows a negative packet loss while analyzing the RTP streams? 10.0.1.84 16470 10.0.0.170 11030 0x37453CB5 ITU-T G.711 PCMA 3739 -68 (-1.9%) 23.56 30008.72 1.49 X |
| 14:04.39 | anonymouz666 | -68? -1.9% ? |
| 14:05.13 | schmidts | anonymouz666 maybe something in between which sends an answer too |
| 14:05.52 | anonymouz666 | I do not understand |
| 14:06.15 | anonymouz666 | and I really don't know how to explain these values |
| 14:06.19 | [TK]D-Fender | anonymouz666: You'ev got Packet Found issues :p |
| 14:06.25 | anonymouz666 | hehe |
| 14:06.53 | russellb | redundancy? |
| 14:06.54 | russellb | :-p |
| 14:06.58 | schmidts | anonymouz666 what is in between the point you took this pcap and the other side of the rtp stream, something like routers, firewall ? |
| 14:07.04 | russellb | they want to make sure you really hear it |
| 14:07.19 | schmidts | russellb good idea, just send it twice :D |
| 14:07.22 | russellb | yup |
| 14:07.28 | anonymouz666 | schmidts: No Sir. It's just ATA -> LAN -> PABX -> LAN -> ATA |
| 14:07.33 | russellb | have you seen positive values before? |
| 14:07.43 | anonymouz666 | yes |
| 14:07.46 | russellb | k |
| 14:07.58 | russellb | packet found errors, then. :-) |
| 14:08.27 | schmidts | anonymouz666 i only know this behavior from ping if some routers in between swith over some metrics so its possible that you get an answer twice but over a different way |
| 14:08.36 | schmidts | s/swith/switch/ |
| 14:09.18 | schmidts | but icmp is different from udp so it shouldnt be possible |
| 14:10.12 | schmidts | anonymouz666 maybe one of the NICs are stupid and sends packages twice out of their send queue |
| 14:11.00 | schmidts | but you should see this in the log, packages with the same sequenz number |
| 14:11.17 | anonymouz666 | schmidts: did you already use tshark to view the RTP streams? |
| 14:12.03 | schmidts | anonymouz666 only with pcap files |
| 14:12.51 | schmidts | is wireshark = tshark? cause you said wireshark ;) |
| 14:12.57 | anonymouz666 | yes |
| 14:13.24 | anonymouz666 | you can run tshark from the console and got a nice output from rtp streams. you don't need to use pcap files. |
| 14:13.46 | jeremy_g | russellb:1.4.19 is not a 4 year old asterisk. |
| 14:15.08 | schmidts | anonymouz666 i see, i have never tried cause i have a sniffer in front of my servers which can make pcaps direct from switch monitoring which is just nice, cause you want loose any power on your server |
| 14:15.08 | drmessano | jeremy_g, 2.5 years.. Your two suggestions are (1) Upgrade to the latest 1.4 release so we aren't troubleshooting OLD code, and (2) upgrade to DAHDI so we're not troubleshooting old code |
| 14:15.47 | drmessano | jeremy_g, nobody like troubleshooting old, and possibly buggy code. Insisting on it will get you nowhere in here |
| 14:16.29 | drmessano | jeremy_g, if the solution to your problem after 3 hours of troubleshooting is "upgrade, this was fixed in a later release" then what? |
| 14:17.15 | schmidts | drmessano oh that would be easy, just write a backport :D |
| 14:17.48 | schmidts | anonymouz666 it this reproducable for you? if yes try to make a pcap file then you could easier find when and where it happens |
| 14:18.05 | drmessano | ~slap schmidts |
| 14:18.05 | infobot | ACTION slaps schmidts, keep your grubby fingers to yourself! |
| 14:18.05 | drmessano | Hmm.. stupid bot |
| 14:18.49 | drmessano | "trixbox pro" <--- oxymoron |
| 14:19.03 | schmidts | thanks to infobot about touching me, i am gonna tell my wife then you will be scared ;) |
| 14:19.11 | z4nD4R | how i can set cipher method on asterisk 1.8. rc3?... i use tlscipher=RC4-SHA |
| 14:19.24 | jeremy_g | drmessano:i already know what you are suggesting. But such a freeze out for a few instants seem to have something to do with the kernel-smp as well |
| 14:19.40 | jeremy_g | drmessano:um gonna strace it all |
| 14:19.52 | jeremy_g | to see where is the problem - curiosity is priceless |
| 14:20.09 | *** join/#asterisk fofware (~Fabian@host165.186-109-184.telecom.net.ar) |
| 14:20.22 | anonymouz666 | schmidts: yes it is. every tshark stats I can see negative values |
| 14:20.33 | anonymouz666 | on differents machines |
| 14:21.41 | jeremy_g | where the problem is. |
| 14:22.26 | schmidts | <PROTECTED> |
| 14:22.45 | *** join/#asterisk jhirley (~chatzilla@mail.mmdlaw.com) |
| 14:24.01 | anonymouz666 | schmidts: sure... I can start the pcap right now and send you after lunch |
| 14:24.42 | schmidts | anoymouz666 just use the emailadress from the dev list, i am going home soon ;) |
| 14:25.51 | *** join/#asterisk coppice (~chatzilla@112.203.17.210.dyn.pacific.net.hk) |
| 14:26.36 | anonymouz666 | ok! |
| 14:27.40 | [TK]D-Fender | trix |
| 14:27.54 | [TK]D-Fender | lol |
| 14:29.29 | z4nD4R | TLS handshake falider after serverHello ist send... any idee? |
| 14:29.47 | *** join/#asterisk OBCom (~Cadey@mail.officebroker.com) |
| 14:29.59 | Maliuta | disable TLS |
| 14:30.13 | Maliuta | or set it up properly in the first place |
| 14:30.33 | z4nD4R | Maliuta: but how? |
| 14:31.13 | z4nD4R | Maliuta: if you whant i show you my sip.conf |
| 14:31.28 | OBCom | Hi Peeps, Am i right in thinking when you listen to system messages with the AMI you do not see the channels that get created when you park a call? I see it Bridge and link a few channels but some of the ID's it bridges and links are do not have a Newchannel sent to say they are on the system? |
| 14:31.42 | z4nD4R | Maliuta: all looks good but tls still not work |
| 14:32.05 | Maliuta | Right now, I'm off to bed. It's 00:30 and I have a big day in about eight hours |
| 14:32.50 | z4nD4R | Maliuta: ok i am understant... maybe latter ;) |
| 14:33.25 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 14:34.07 | OBCom | sorry, im listening to system and call messages |
| 14:34.15 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 14:34.27 | p3nguin | |
| 14:34.55 | [TK]D-Fender | OBCom: the NewChannel should be for the new call placed by the park-er |
| 14:35.10 | OBCom | [TK]D-Fender : Hi again Fender |
| 14:35.15 | [TK]D-Fender | OBCom: what follows is an attended transfer hand-off |
| 14:35.40 | [TK]D-Fender | OBCom: You might have to track potential parking channels to match up events. |
| 14:35.42 | OBCom | [TK]D-Fender : I dont see this in the AMI though when i use a feature park (#72 for example) |
| 14:36.10 | [TK]D-Fender | OBCom: Oh.. Features.conf based... not sure exactly how that might look.. |
| 14:36.20 | [TK]D-Fender | OBCom: Stop doing those :p |
| 14:36.24 | OBCom | :) |
| 14:36.32 | OBCom | how else could i park a call ? |
| 14:36.49 | leifmadsen | Pre-orders for the next Asterisk book are now available: http://oreilly.com/catalog/9780596517342 |
| 14:37.49 | p3nguin | obcom: Most people transfer to the park extension. |
| 14:38.10 | OBCom | Xfer 700 or somthing ? |
| 14:38.24 | p3nguin | If you've set up 700 as the park extension, yes. |
| 14:38.28 | OBCom | or litraly send a redirect to the AMI ? |
| 14:39.02 | OBCom | both seem to work :) |
| 14:39.05 | OBCom | OK ill try that |
| 14:39.23 | *** part/#asterisk DelphiWorld (~Delphi@41.200.3.40) |
| 14:40.42 | *** join/#asterisk eugeneoden (~goden@rrcs-97-77-255-86.sw.biz.rr.com) |
| 14:42.57 | OBCom | [TK]D-Fender : same issue dude |
| 14:43.32 | [TK]D-Fender | ObYou should see the New Channel for the start, and ATXFER for the bridge |
| 14:44.25 | OBCom | Ill post the output and you will see what i mean |
| 14:44.48 | OBCom | http://pastebin.com/yus1bB9k |
| 14:45.07 | OBCom | line 246 is the park |
| 14:46.39 | OBCom | channel ID 1287588978.2655 <-- you dont see that get created |
| 14:47.25 | OBCom | teh closest you get is seeing it cloned at line 208 |
| 14:47.52 | OBCom | then renamed at 223 which is when you see the ID for the first tiome |
| 14:48.42 | *** join/#asterisk BANSAL (~bansal@117.199.117.111) |
| 14:48.45 | [TK]D-Fender | ObYou need to see the NeExten in there somewhere... |
| 14:48.52 | [TK]D-Fender | OBCom: Kinda has to happen |
| 14:49.29 | OBCom | so i need to listen for Dialplan events also right? |
| 14:50.00 | russellb | http://oreilly.com/catalog/9780596517342 <--- yay! |
| 14:50.19 | Katty | hmm. |
| 14:50.20 | OBCom | NeExten as in NewExten which is a dialplan message ? |
| 14:50.32 | Katty | so can you dial(SIP/foo&Zap/foo) at the same time? |
| 14:51.08 | OBCom | Katty : Queue/Group ? |
| 14:51.43 | Katty | local might work. |
| 14:51.52 | [TK]D-Fender | OBCom: NewExten. Yes, typo. |
| 14:51.56 | schmidts | katty yes that what happens by a enum dialplan injection |
| 14:52.09 | [TK]D-Fender | kayy Yes |
| 14:52.14 | [TK]D-Fender | Katty: even |
| 14:53.19 | anonymouz666 | schmidts: http://www.wireshark.org/lists/wireshark-users/201005/msg00059.html - it explains the negative values |
| 14:53.31 | anonymouz666 | but I think I didn't understand this well |
| 14:53.56 | schmidts | anonymouz666 a got it |
| 14:54.31 | schmidts | if you have for example 100 packets which all need 20ms to travel from start to you, and you have 10 which only need 15 ms you will have a negative delta |
| 14:55.17 | drmessano | I fly negative delta |
| 14:57.30 | anonymouz666 | schmidts: ooh yes. Understood |
| 15:04.02 | Katty | yay, it works. |
| 15:04.10 | Katty | also, boo |
| 15:04.31 | Katty | i really don't want this to happen |
| 15:05.04 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 15:08.01 | drmessano | russellb, I should have the 1.12 version ready here soon as well |
| 15:08.11 | russellb | drmessano: awesome |
| 15:08.29 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 15:08.46 | drmessano | russellb, Well, not true.. I guess that would be 2.2 since we're announcing 2.0 at Astricon |
| 15:08.55 | russellb | we are? |
| 15:08.59 | drmessano | Yep |
| 15:09.05 | russellb | i was not aware |
| 15:09.12 | drmessano | You didn't get the memo, I guess |
| 15:09.13 | russellb | why does nobody keep me up to date |
| 15:09.25 | drmessano | I will call Mark and find out what the hell is going on, and get you a copy of that memo |
| 15:09.30 | russellb | thanks |
| 15:09.34 | drmessano | anytime |
| 15:09.55 | Naikrovek | announcing 2.0 of what at Astricon |
| 15:10.03 | drmessano | Asterisk |
| 15:10.09 | drmessano | Shh.. don't tell russellb |
| 15:10.12 | Naikrovek | k |
| 15:10.13 | drmessano | I mean |
| 15:10.20 | drmessano | "Until he gets the memo" |
| 15:10.32 | russellb | you'd think i'd be at least casually made aware of such a thing |
| 15:10.42 | russellb | i guess that's what just happened |
| 15:10.45 | Naikrovek | instead the whole channel was notified |
| 15:10.46 | Qwell | russellb: we should probably get started on that if we're announcing it next week |
| 15:10.48 | Naikrovek | THEN you saw it |
| 15:10.52 | russellb | Qwell: no kidding |
| 15:10.57 | leifmadsen | Qwell: amen |
| 15:11.03 | russellb | s/1.8/2.0/ ? |
| 15:11.04 | russellb | done |
| 15:11.10 | drmessano | russellb, I hate when that shit happens.. I had 3 of my 11 divorces hit me like that |
| 15:11.18 | leifmadsen | svn mv $svn/1.8 $svn/2.0 |
| 15:11.18 | Naikrovek | lol |
| 15:11.31 | russellb | leifmadsen: make it happen |
| 15:11.42 | *** join/#asterisk n3hxs (~HAMming@63.68.135.4) |
| 15:11.43 | leifmadsen | laughs at his initial reaction to actually do it |
| 15:12.34 | drmessano | I do have one question about 2.0. Are we gonna rename anything else to dome D _ _ _ I name? |
| 15:12.38 | drmessano | some* |
| 15:13.00 | russellb | DOOBI |
| 15:13.10 | Naikrovek | hah! |
| 15:13.22 | russellb | Digium Object-Oriented B? Interface |
| 15:13.28 | drfreeze | Does * support multiple sip devices with the same username? |
| 15:13.32 | leifmadsen | no |
| 15:13.47 | russellb | well, sort of, if you match on them by IP address with type=peer |
| 15:13.57 | leifmadsen | ... |
| 15:14.05 | drfreeze | is there a way to get the same extension on multiple phones ? |
| 15:14.05 | russellb | ... |
| 15:14.09 | leifmadsen | "it won't work as you probably expect it to" :) |
| 15:14.13 | russellb | leifmadsen: +1 |
| 15:14.19 | leifmadsen | drfreeze: yes |
| 15:14.21 | russellb | sure, define multiple SIP accounts |
| 15:14.28 | russellb | and then Dial(SIP/account1&SIP/account2) |
| 15:14.29 | russellb | done |
| 15:14.34 | leifmadsen | drfreeze: exten => 100,1,Dial(SIP/foo&SIP/bar) |
| 15:14.39 | russellb | i totally beat you |
| 15:14.39 | Naikrovek | drfreeze: you want one extension to ring multiple phones? easy, see above |
| 15:14.44 | leifmadsen | russellb: you were not pedantic |
| 15:14.58 | russellb | leifmadsen: i'm going to pedactically punch you |
| 15:15.12 | drfreeze | well, actually I'm asking if Dial(SIP/500) can dial multiple phones |
| 15:15.13 | drmessano | leifmadsen, russellb has pedantic=no set |
| 15:15.15 | leifmadsen | s/pedactically/pedantically/ |
| 15:15.21 | drfreeze | ie, more than one phone with ext 500 |
| 15:15.26 | russellb | awesome typo |
| 15:15.32 | drmessano | leifmadsen, please change the config and reload |
| 15:15.34 | leifmadsen | drfreeze: yes, see above example |
| 15:15.34 | drmessano | kthx |
| 15:15.43 | leifmadsen | module reload res_awesome.so |
| 15:15.49 | leifmadsen | <PROTECTED> |
| 15:15.58 | drfreeze | leifmadsen: you mean, no, see above |
| 15:16.09 | russellb | Asterisk does not support, and never will support, SIP/500 mapping to more than one device. |
| 15:16.15 | drfreeze | thanks |
| 15:16.16 | russellb | ... unless SIP/500 goes to some forking proxy |
| 15:16.22 | leifmadsen | extension != device |
| 15:16.27 | leifmadsen | my answer is still correct |
| 15:16.32 | russellb | but Asterisk doesn't work properly with forking proxies, either, so don't do that |
| 15:16.45 | drfreeze | Dial(SIP/500) can dial multiple phones (if yes is your answer) |
| 15:16.58 | russellb | no, it can not |
| 15:16.59 | leifmadsen | SIP/500 is a device, not an extension |
| 15:17.12 | leifmadsen | so my answer still applies as yes, see the example above |
| 15:17.22 | drfreeze | oh, I see, sure |
| 15:17.30 | leifmadsen | a single extension can map to multiple devices, yes |
| 15:17.42 | drfreeze | my bad. I mean device |
| 15:17.51 | drfreeze | *meant |
| 15:18.01 | leifmadsen | and russell has since defined the answer :) |
| 15:18.18 | *** join/#asterisk bent_screwdriver (~socain00@74.255.249.66) |
| 15:20.42 | drmessano | If we're getting a 1.8 book, does that mean the 1.6 book has been scrapped, and does that further reduce the chances of a 1.0 book? |
| 15:20.49 | *** join/#asterisk lauris (~la@unaffiliated/lauris) |
| 15:20.58 | lauris | hi |
| 15:22.06 | [TK]D-Fender | drmessano: 1.8 is supposed to be LTS, and 1.6.1 is already DOA |
| 15:22.13 | lauris | i'm facing problem with 1.6.2.11 and cdr's |
| 15:22.14 | [TK]D-Fender | drmessano: Guessing this is a "no looking back" doc |
| 15:22.44 | lauris | i don't why but there is no cdr is call is redirected through asterisk manager |
| 15:22.52 | lauris | anyone familiar with this issue ? |
| 15:23.06 | drmessano | [TK]D-Fender, this LTS thing bothers me a little in that, what if we don't like 1.8.. what if it's the wrong color or smells funny? |
| 15:23.12 | drmessano | [TK]D-Fender, we are screwed |
| 15:23.25 | [TK]D-Fender | drmessano: You have the source :p |
| 15:23.38 | [TK]D-Fender | drmessano: Lift your skirt, grab your balls, and MAN UP :p |
| 15:23.43 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 15:23.46 | drmessano | [TK]D-Fender, I know, but I can't keep calling russellb at home for everything |
| 15:24.29 | *** join/#asterisk rgavril (~rgavril@78.96.142.35) |
| 15:24.32 | [TK]D-Fender | drmessano: You clearly need his office # too ;) |
| 15:25.15 | rgavril | I'm using dahdi 2.4.0 and I get this error : "wctdm24xxp 0000:0b:00.0: ERROR: Unable to service card within 25 ms and unable to further increase latency." ? Any known fix for this ? |
| 15:25.38 | drmessano | [TK]D-Fender, I am sure if I disturb him during dinner one more time he's gonna my calls to file or leifmadsen's houses and those guys just hang up on me |
| 15:27.33 | *** join/#asterisk DelphiWorld (~Delphi@41.200.3.40) |
| 15:27.35 | DelphiWorld | hi |
| 15:27.45 | DelphiWorld | [TK]D-Fender: in what format my carrier should send the did to me? |
| 15:27.59 | *** join/#asterisk timahvo1 (~rogue@41.72.215.94) |
| 15:28.48 | russellb | drmessano: OH, I meant to tell you something about 1.8! |
| 15:28.58 | [TK]D-Fender | DelphiWorld: Sorry, could yuo be a little more vague? |
| 15:29.05 | russellb | We have google talk / google voice inbound and outbound working for 1.8.0 :-) |
| 15:29.15 | DelphiWorld | [TK]D-Fender: i say i have a did from my carrier |
| 15:29.18 | DelphiWorld | through iax2 |
| 15:29.24 | DelphiWorld | but if i dial it is looking for exten S |
| 15:29.34 | [TK]D-Fender | DelphiWorld: Sogo MAKE IT |
| 15:29.42 | *** part/#asterisk TobSnyder (~schneider@dslb-088-073-204-184.pools.arcor-ip.net) |
| 15:29.49 | russellb | drmessano: hopefully that helps the smell |
| 15:29.55 | DelphiWorld | [TK]D-Fender: ? |
| 15:30.09 | *** join/#asterisk Deeewayne (~dwayne@75-150-14-49-Atlanta.hfc.comcastbusiness.net) |
| 15:30.09 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:30.14 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:30.14 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 15:30.22 | [TK]D-Fender | DelphiWorld: What part of MAKE THE DAMN EXTENSION is unclear? |
| 15:30.34 | [TK]D-Fender | DelphiWorld: "doesn't exist" <- go make it exist |
| 15:30.36 | DelphiWorld | [TK]D-Fender: no idea |
| 15:30.48 | DelphiWorld | [TK]D-Fender: i need it to look for a full did not for S |
| 15:31.05 | DelphiWorld | [TK]D-Fender: so my carrier should send in special format |
| 15:31.17 | *** join/#asterisk KavanS (~KavanS@unaffiliated/kavans) |
| 15:31.21 | russellb | drmessano: I am very disappointed in your lack of response. |
| 15:31.32 | [TK]D-Fender | DelphiWorld: Do you really? Is there really a chance of the exten you'll create being landed on by something else? |
| 15:31.55 | [TK]D-Fender | DelphiWorld: perhaps you should at what you TOLD them to send it to you as in your REGISTER statement |
| 15:32.53 | DelphiWorld | russellb: any explanation please? |
| 15:33.26 | russellb | i have no idea what you're talking about |
| 15:33.29 | [TK]D-Fender | russellb: Far from the first.. or last... |
| 15:36.32 | *** join/#asterisk nny (~Scott@174.107.201.103) |
| 15:38.55 | nny | ok I know this is probably a sore subject, but I am trying to educate a consulting client that is using asterisk w/freepbx in larger installations. I am under the impression that this scenario is the lesser desired of two possible scenarios, the other being asterisk alone. I don't want to start a flame war, I like freepbx and feel it has it's place. Thoughts? |
| 15:40.21 | DelphiWorld | my carrier should send the number to me but i don't know how to in iax |
| 15:40.25 | DelphiWorld | in sip is working but in iax no |
| 15:41.53 | [TK]D-Fender | [11:31]<[TK]D-Fender>DelphiWorld: perhaps you should at what you TOLD them to send it to you as in your REGISTER statement |
| 15:42.03 | russellb | just add ... exten => s,1,Goto(whatever) |
| 15:42.03 | drmessano | russellb, I saw that!! Joking about "being stuck with 1.8" aside, I think 1.8 is going to be a great release. 1.6.x and 1.8.x have implemented a ton of new features that I know have been much requested in the community, which is exciting stuff. |
| 15:42.15 | [TK]D-Fender | nny: It does. There. |
| 15:42.32 | russellb | drmessano: cool. :-) just wanted to make sure you saw since you asked me about it a while back. |
| 15:43.40 | nny | [TK]D-Fender: if anyone else had said that, I would demand proof. I just need to prove that you said it now. Many religions have gone horribly wrong over similar circumstances... |
| 15:44.20 | Qwell | drmessano: psst |
| 15:44.21 | Qwell | http://packages.asterisk.org/centos/5/beta/ |
| 15:44.30 | [TK]D-Fender | nny: It has its place. It is for all users. It isn't for multi-tennnent. It is for the joe-blow schmuck SMBs pretty much |
| 15:45.19 | [TK]D-Fender | nny: Think about getting too "interesting" and you can forget it. It has limits. Lots. But for largely homogenous basic scenarios its filne. |
| 15:45.35 | nny | [TK]D-Fender: yeah just trying to explain it to people using it as a work horse when it's not. A call center is not the best place for it |
| 15:45.54 | nny | [TK]D-Fender: we agree pretty much, just need to be convincing. Time to break out the pointy sticks |
| 15:46.04 | *** part/#asterisk DelphiWorld (~Delphi@41.200.3.40) |
| 15:48.46 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 15:50.32 | [TK]D-Fender | nny: "call center" is far too vague a term |
| 15:50.40 | [TK]D-Fender | nny: Its a question of call processing requirements. |
| 15:51.20 | [TK]D-Fender | nny: The GUI options for queue management may not let you handle things extly the way you want (VERY possible). Inboind vs outbound? Lost of questions there... |
| 15:53.10 | nny | [TK]D-Fender: it works fairly well, configuration using x_custom.conf is apt for adding in dialplan changes the GUI won't allow or doesn't offer. Sometimes adding those changes in requires more work and complexity than a vanilla system due to the abstraction layer. |
| 15:54.10 | nny | [TK]D-Fender: really it's that layer (as a needed evil for it's purpose) that also cause's some pain though. troubleshooting is a lot more difficult, as you have to discern what each piece is trying to do independently without being able to acrhitect them from the dialplan directly |
| 15:54.57 | *** part/#asterisk sekil (~sekil@80.93.247.26) |
| 15:55.37 | drmessano | nny: FreePBX is not designed for larger installations.. call center, setting up an ITSP, etc. I agree with all involved on that. This is where the distinction between "Using Asterisk to replace an office PBX" and "Using Asterisk has a highly configurable, and scalable telephony engine" comes into place |
| 15:56.21 | nny | drmessano: yeah we agree entirely. the alternative need is client configuration, which can be emulated on a much smaller scale with real time and users.conf |
| 15:56.31 | nny | it really depends on the scenario |
| 15:56.33 | drmessano | FreePBX's feature set aims to address a lot of Asterisk's features, but it's very much designed around the "Office PBX" paradigm |
| 15:56.43 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 15:56.43 | nny | i am just trying help someone who is trying to use only it as a tool |
| 15:56.46 | nny | and not both |
| 15:57.23 | drmessano | nny: I tell people all the time "Just because FreePBX lets you configure Queues, does not mean you can put 1000 agents in a callcenter on a FreePBX box" |
| 15:58.06 | leifmadsen | +1 |
| 15:58.18 | KavanS | drmessano, do people successfully deploy asterisk in said 1000 agent scenarios? |
| 15:58.27 | [TK]D-Fender | ~users.conf |
| 15:58.41 | drmessano | KavanS, people deploy Asterisk for large callcenters, ITSPs, etc |
| 15:58.48 | nny | kk that's good points, thank you. I am not trying to steer anyone away from using it. I just want to make sure that the people I deal with understand where it is designed to work, and where it can work but not the only option or suitable use. Right now they think just that.. |
| 16:00.38 | KavanS | drmessano, roger that, just wanted to know the differences from Asterisk and the defacto enterprise alternatives - I suppose I need to google a bit |
| 16:01.22 | leifmadsen | KavanS: it requires multiple servers and a good deal of experience with asterisk to scale it up to 1000+ seats |
| 16:02.36 | drmessano | KavanS, I have personally seen several scenarios at deploying asterisk large scale.. whether it be dozens of boxes trunked across multiple locations or multi-server systems in a central location with frontend/backends using Realtime |
| 16:02.45 | nny | on scalability, is DUNDI preferred over abstraction layers on the OS model for distributed systems. (Asterisk layer only) |
| 16:02.52 | nny | er ./? |
| 16:02.59 | *** join/#asterisk drudge` (drudge@unaffiliated/drudge/x-837452) |
| 16:04.25 | drmessano | KavanS, the difference between Asterisk and some of your "SIP Proxy" applications is that there isn't any sort of "clustering" in Asterisk.. but the same can be achieved, reliably, using facilities build into Asterisk |
| 16:05.16 | leifmadsen | nny: DUNDi is usually one aspect, yes |
| 16:05.21 | leifmadsen | DUNDi is not the whole solution |
| 16:05.38 | nny | leifmadsen: yes agreed |
| 16:05.55 | leifmadsen | many tools come into play when building multi-box asterisk systems |
| 16:05.58 | leifmadsen | heads to lunch |
| 16:07.45 | KavanS | interesting |
| 16:07.53 | KavanS | gives a google to asterisk clustering |
| 16:08.23 | fauxalliance | wishes asterisk worked like CARP w/ PF |
| 16:09.01 | fauxalliance | ^^effective 'load-balancing' but decidedly not clustering |
| 16:09.31 | *** join/#asterisk kerframil (~kerframil@gentoo/user/kerframil) |
| 16:09.32 | nny | fauxalliance: what layer does CARP work at? I assume above 3 if asterisk needs to know about it. I have read briefly on the subject |
| 16:09.57 | drmessano | fauxalliance, I prefer TROUT or PERCH for my HA solutions, but FLOSS, BRUSH, and SCRUB applications don't support these easily. |
| 16:10.19 | [TK]D-Fender | drmessano: I DON'T WANT NO SCRUBS |
| 16:10.35 | fauxalliance | chokes on his coffee |
| 16:12.04 | fauxalliance | drmessano, dont make me run asterisk 1.2 on an depricated OpenMOSIX cluster, with buggy at best SHM patch just because.... |
| 16:13.11 | drmessano | fauxalliance, throw in Zaptel and you got a deal |
| 16:13.23 | fauxalliance | can i use my X100pCLOWN |
| 16:14.22 | fauxalliance | throws 'realtime' in front of the first MACK truck that rolls in. |
| 16:14.28 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:15.44 | fauxalliance | http://www.openinnovation.it/c/document_library/get_file?p_l_id=14284&folderId=10401&name=DLFE-401.pdf @ KavanS |
| 16:16.11 | fauxalliance | "A reliable architecture for Asterisk Cluster" ^^ |
| 16:17.42 | drmessano | fauxalliance, I threw some X100Ps in a box and it was a real cluster. Does that count? |
| 16:17.58 | fauxalliance | s/cluster/clusterfuck ... and yes |
| 16:22.50 | OBCom | [TK]D-Fender : Hay Fender, do you know why there are so many "Bridge" and Unlink messahe when you park a call? |
| 16:23.01 | OBCom | i.e whats the reason? :S |
| 16:23.18 | fauxalliance | drmessano, http://30.media.tumblr.com/tumblr_kzq884RPWs1qa8pf4o4_500.jpg Caption: Load Balancing with a CARP |
| 16:23.21 | OBCom | it seems to bounce the bridge and then unlink, then bridge over and over? |
| 16:24.49 | drmessano | fauxalliance, HAHAHA |
| 16:32.01 | fauxalliance | http://blog.stackoverflow.com/wp-content/uploads/Ubuntu-Motivational.png |
| 16:32.14 | drmessano | OBCom, Asterisk isn't really good at parallel parking. Takes it a few tries |
| 16:32.22 | *** join/#asterisk Ta^3 (~tacvbo@gponr9-203-11-242.iusacell.net) |
| 16:33.40 | fauxalliance | http://goo.gl/dnYl @ drmessano |
| 16:36.47 | drmessano | HAHA |
| 16:36.51 | fauxalliance | ;-) |
| 16:41.58 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 16:42.07 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 16:44.00 | skrusty | anyone know how to check group count from an AGI? |
| 16:45.09 | *** join/#asterisk citywok (~kvirc@67-134-194-33.dia.static.qwest.net) |
| 16:46.22 | *** part/#asterisk nny (~Scott@174.107.201.103) |
| 16:48.51 | *** join/#asterisk Peaceful (~Peaceful@74-92-245-181-Utah.hfc.comcastbusiness.net) |
| 16:48.57 | *** join/#asterisk lirakis (~lirakis@ool-ad022bb1.dyn.optonline.net) |
| 16:49.07 | lirakis | sooo |
| 16:49.14 | lirakis | anyone else see this nortle 1535 phone? |
| 16:49.17 | lirakis | *nortel |
| 16:49.27 | lirakis | im curious as to why its so darn cheap now |
| 16:50.01 | lirakis | is it end of life ... liquidation? even if thats the case... it seems too cheap for a phone with its feature set |
| 16:52.19 | jdoe | "this"? |
| 16:52.22 | jdoe | which? |
| 16:52.47 | WIMPy | Non-standard? |
| 16:52.54 | lirakis | jdoe, saw it on nerd vittles |
| 16:53.06 | lirakis | WIMPy, it says its 3261 compliant SIP |
| 16:53.13 | GameGamer43 | lirakis: its discontinued by the manufacturer |
| 16:53.19 | coppice | its a SIP phone, so I guess it obeys no standards |
| 16:53.25 | lirakis | coppice, hehe |
| 16:53.48 | lirakis | GameGamer43, is that speculation or are you certain |
| 16:53.52 | WIMPy | Good point :-( |
| 16:53.58 | lirakis | GameGamer43, even if it is EOL ... its way too cheap |
| 16:53.59 | lirakis | haha |
| 16:54.08 | GameGamer43 | lirakis: I'm 100% positive |
| 16:54.13 | lirakis | okay |
| 16:54.32 | coppice | the manufacturer is discontinued |
| 16:55.14 | jdoe | lol |
| 16:56.01 | lirakis | yeah i knew that nortel sold off the cvas division etc. |
| 16:56.19 | lirakis | wasnt sure if they were going to stop doing the phone thing |
| 16:56.31 | lirakis | still though ... it does seem like a good deal |
| 16:56.41 | lirakis | i was curious if anyone has actually used one |
| 16:56.57 | lirakis | although for $40 ... seems worth the "gamble" |
| 16:57.04 | lirakis | a new toy :D |
| 16:58.28 | coppice | because they are out of business, software updates will be really hard to come by.... so pretty much like most other SIP phones |
| 17:00.12 | *** join/#asterisk wizhippo (~Adium@64.201.57.7) |
| 17:01.02 | wizhippo | is anybody using lua? How do I get the result from app.read() |
| 17:06.13 | *** join/#asterisk oej (~olle@2001:470:1f15:d79:225:ff:fe44:74ec) |
| 17:07.04 | cusco | one of the parametes is the name of the variable |
| 17:08.02 | *** join/#asterisk twanny796 (~twanny@80.71.106.213) |
| 17:09.16 | cusco | im not using lua, but I'm sure application read is teh same. |
| 17:09.56 | cusco | Read(variable,path/to/audio,9,,3,8); |
| 17:10.09 | cusco | noop(${variable} |
| 17:10.15 | cusco | that is 9 digits, 3 attempts 8 secs timeout |
| 17:10.56 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 17:14.39 | *** join/#asterisk wimt (wimt@freenode/staff/wikipedia.wimt) |
| 17:21.35 | Katty | ATTENTION |
| 17:21.38 | Katty | I HAVE CAFFEINE AND CUPCAKES |
| 17:21.42 | Katty | that is all. |
| 17:22.24 | [TK]D-Fender | Katty: mew! |
| 17:22.31 | Katty | [TK]D-Fender: ohai |
| 17:24.22 | *** join/#asterisk guyzmo (~guyzmo@nenya.mithrandir.net) |
| 17:24.27 | guyzmo | hi |
| 17:24.36 | Katty | ohai2u |
| 17:24.54 | guyzmo | I'm struggling to configure a cisco 7940g phone with asterisk 1.4.21 |
| 17:25.04 | Katty | i am sorry to hear this. |
| 17:25.17 | guyzmo | I upgraded it to 8.12.0 sip |
| 17:25.33 | guyzmo | and it authenticates to the asterisk server |
| 17:25.41 | Katty | Fancy. |
| 17:26.02 | guyzmo | but I can't receive or send a call, only getting ' |
| 17:26.04 | guyzmo | chan_sip.c: Call from '' to extension '4216' rejected because extension not found.' |
| 17:26.14 | guyzmo | (sorry for doing too much \n :) ) |
| 17:26.31 | Katty | who/what is extension 4216 |
| 17:26.37 | guyzmo | I tried to find an answer on the web, but I can't |
| 17:26.44 | Katty | ON THE INTERNETS |
| 17:26.48 | guyzmo | it's a valid extension that works with softphones |
| 17:26.52 | guyzmo | :) |
| 17:26.56 | Katty | that's not what i asked tho. |
| 17:27.00 | guyzmo | (and other sip phones) |
| 17:27.26 | guyzmo | what did you ask then ? |
| 17:27.58 | Katty | what's the price of rice in china. |
| 17:28.15 | n3hxs | Buck Two fifty an ounce. |
| 17:28.21 | coppice | about 50 yuan for an 8 kilo bag |
| 17:28.30 | Katty | giggles |
| 17:28.46 | n3hxs | that is the cheap stuff. |
| 17:29.11 | guyzmo | well, it seems it's not the good time to ask stupid technical stuff :) |
| 17:29.14 | Katty | does that include import tax? |
| 17:29.27 | Katty | guyzmo: that seems rather passive aggressive. |
| 17:29.32 | Katty | guyzmo: and that simply will not due sir. |
| 17:29.33 | coppice | import? they grow it in china. |
| 17:29.43 | Katty | coppice: yes but i'm not going to china to buy it |
| 17:29.52 | Katty | i guess that defeats the purpose of the question |
| 17:30.21 | coppice | you asked for rice in china, not exported chinese rice |
| 17:30.42 | guyzmo | Katty - sorry for being rude to ask something like that, but I'm fighting with those phones since 2 days, and when I finally got to configure them entirely |
| 17:30.52 | *** join/#asterisk Cadey (Cadey@host86-170-37-189.range86-170.btcentralplus.com) |
| 17:30.57 | guyzmo | asterisk does not let me send a single call |
| 17:31.05 | Naikrovek | anyone want google voice invite? I have 2 left, it says |
| 17:31.14 | *** join/#asterisk ferdna (~yup@cpe-24-92-114-97.elp.res.rr.com) |
| 17:31.18 | Katty | Naikrovek: ohai |
| 17:31.25 | Naikrovek | ohai |
| 17:31.25 | Katty | Naikrovek: i do not want a google voice invite. |
| 17:31.29 | Katty | Naikrovek: but i would take a hug. |
| 17:31.32 | guyzmo | and I usually ask on IRC before sending an email to the list, just to exclude stupid stuff I may have forgotten :) |
| 17:31.36 | Naikrovek | then do not ask for one. |
| 17:31.40 | Naikrovek | I will, however, hug you. |
| 17:31.42 | Naikrovek | hugs Katty |
| 17:31.44 | Katty | :>>>>>>>>>>>> |
| 17:31.46 | Katty | hugs on Naikrovek |
| 17:32.16 | Katty | guyzmo: i wish you the best of luck in finding a solution to your issue. |
| 17:32.26 | guyzmo | but please accept my apologies for being aggressive |
| 17:32.31 | guyzmo | I never wanted to look that way |
| 17:32.46 | n3hxs | thinks the channel has a bad case of digression. |
| 17:32.57 | Katty | this channel IS digression. |
| 17:33.02 | guyzmo | though, sometime, on some channels, people ask technical stuff, and other help :) |
| 17:33.14 | Katty | and i'm sure with patience someone will! |
| 17:33.16 | guyzmo | but that may be in a parallel timespace dimension |
| 17:33.16 | Katty | it just won't be from me. |
| 17:33.24 | guyzmo | that I understood ;) |
| 17:33.33 | Katty | excellent. |
| 17:33.43 | Naikrovek | beautiful thing about IRC is that you cannot talk over someone else, you just show up alongside |
| 17:33.50 | Naikrovek | it gets cluttered, but nothing is lost |
| 17:34.06 | Katty | what about sanity |
| 17:34.12 | guyzmo | Naikrovek - and even better, you can interleave digressions with intopic discussions |
| 17:34.14 | Katty | i think it might have the potential to get lost. |
| 17:34.19 | Naikrovek | sanity can be lost if you're not practiced in the art |
| 17:34.25 | guyzmo | though sometimes (actually most of the time) in-topic gets flooded |
| 17:34.28 | Naikrovek | and you learn to type quickly in IRC |
| 17:34.34 | Naikrovek | can't type fast = can't stay in conversation |
| 17:34.42 | Katty | you can consider me comic relief for the channel. |
| 17:34.42 | guyzmo | Naikrovek - for that, there is the dvorak layout :) |
| 17:34.54 | coppice | Katty: there's no sanity claus in the IRC service agreement |
| 17:35.06 | Katty | we have a service agreement? |
| 17:35.07 | guyzmo | engineered for IRC 20 years before it existed |
| 17:35.07 | Naikrovek | claus? lol |
| 17:35.08 | Katty | hot diggity |
| 17:35.25 | Naikrovek | never tried dvorak |
| 17:35.28 | Naikrovek | not sure i wanna |
| 17:35.32 | n3hxs | Sanity Clause? isn't he that guy from up North? |
| 17:35.40 | Naikrovek | 5% faster isnt' enough reason to switch |
| 17:36.00 | guyzmo | Naikrovek - take off all your keys, put them in a bag, ramble it, and send it over your keyboard, you get dvorak :) |
| 17:36.03 | Katty | n3hxs: may i facebook you? |
| 17:36.09 | n3hxs | yeah |
| 17:36.11 | Katty | n3hxs: :> |
| 17:36.29 | guyzmo | Naikrovek - maybe 5% faster, but the comfort of using it is not comparable |
| 17:36.29 | Cadey | Anyone in here knowledgable with AMI ? |
| 17:36.29 | n3hxs | gets a book in the face |
| 17:36.39 | Katty | n3hxs: status updated! |
| 17:36.46 | guyzmo | like this |
| 17:37.00 | Naikrovek | guyzmo: are you saying dvorak is more comfortable? |
| 17:37.05 | guyzmo | Naikrovek - indeed |
| 17:37.10 | Naikrovek | hmm |
| 17:37.16 | guyzmo | I'm using it since 2001 |
| 17:37.16 | Naikrovek | ok |
| 17:37.25 | Katty | i think talking is more comfortable |
| 17:37.26 | Naikrovek | my ergo keyboard is pretty comfy |
| 17:37.32 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 17:37.32 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 17:37.39 | Katty | HAI QWELL |
| 17:37.59 | guyzmo | anyway, that fscking cisco phone blowed my mind |
| 17:38.12 | Katty | it blowed your mind |
| 17:38.19 | guyzmo | litterally I mean |
| 17:38.27 | Katty | kinky. |
| 17:39.53 | carrar | hat |
| 17:39.54 | carrar | what |
| 17:39.54 | n3hxs | Cadey AMI as in T1? |
| 17:40.18 | Cadey | heheh no, as in The Asterisk Manager Interface (AMI) |
| 17:40.28 | n3hxs | LOL... |
| 17:40.42 | Katty | i am familiar with making cupcakes. if that will help. |
| 17:40.53 | n3hxs | OK send a couple this way. |
| 17:40.56 | guyzmo | Cadey - it's not the good time to ask technical questions |
| 17:41.00 | Naikrovek | I made two cheesecakes the other day |
| 17:41.02 | Katty | :P |
| 17:41.12 | Cadey | guyzmo : when is a good time? |
| 17:41.17 | Katty | it's always ag ood time to ask technical questions! |
| 17:41.27 | n3hxs | got an ugly cupcake from the vending machine the other day. |
| 17:41.37 | Katty | was it a golden one |
| 17:41.50 | Katty | orange? |
| 17:41.52 | Katty | dingdong? |
| 17:41.53 | lirakis | does asterisk actually respond to options pings for a user address based on whether or not that user is registered? or does it just send 200 for everything? |
| 17:42.01 | n3hxs | Yellow |
| 17:42.08 | Naikrovek | please do the needful. toss that rancid cupcake in the bin |
| 17:42.13 | Katty | hostess? |
| 17:42.18 | wizhippo | found it app.read("fred") app.noop("fred = " .. channel.fred:get()). thanks for the suggestion |
| 17:42.26 | n3hxs | No, I had to get it myself. |
| 17:42.26 | Naikrovek | "Please do the needful." <--- the phrase I hate to see most |
| 17:42.39 | Katty | i don't get it. |
| 17:42.48 | n3hxs | No hostess here |
| 17:42.57 | Katty | oh. |
| 17:42.59 | Katty | what country are you in |
| 17:43.10 | n3hxs | Foreign.... MD |
| 17:43.17 | Naikrovek | maryland? |
| 17:43.21 | Naikrovek | :_ |
| 17:43.22 | Naikrovek | :) |
| 17:43.25 | n3hxs | something like that |
| 17:43.34 | Katty | ^_- |
| 17:43.52 | n3hxs | but changing to DE :) |
| 17:44.05 | guyzmo | Cadey - I'd say, when the chan does not gets crazy :) |
| 17:44.10 | Katty | deleware or germany? |
| 17:44.16 | guyzmo | Cadey - I also came to ask something |
| 17:44.27 | n3hxs | LOL, Delaware, wish for Germany. |
| 17:44.43 | guyzmo | bt instead I think I'm gonna take my non-working cisco phone and burn it |
| 17:44.46 | Naikrovek | this channel is awesome |
| 17:44.49 | guyzmo | or slam it over a wall |
| 17:45.03 | Katty | have you tried cupcake therapy? |
| 17:45.04 | guyzmo | or punch it until the keys jump |
| 17:45.06 | Naikrovek | in ##java I got kicked for asking about programming in general (as opposed to java specific programming) |
| 17:45.10 | Katty | it does me a world of good. |
| 17:45.15 | n3hxs | I have heard others feel the same way about Cisco |
| 17:45.36 | Naikrovek | those keys can probably be cleaned |
| 17:45.54 | Naikrovek | i've taken apart many a polycom cleaning off thick grease |
| 17:45.58 | guyzmo | n3hxs - well, the first post I've read about cisco 7940g + asterisk began with (sic) : 'The guys at Cisco are good for one thing: making administration a nightmare.' |
| 17:46.11 | Naikrovek | lol |
| 17:46.12 | Katty | Naikrovek: wanna clean my polycom? |
| 17:46.25 | Naikrovek | Katty: ship it here and i will clean it. |
| 17:46.33 | Katty | :< |
| 17:46.35 | Naikrovek | i know |
| 17:46.38 | Naikrovek | not ideal |
| 17:46.42 | Katty | that totally went over your head, did it ;) |
| 17:46.48 | Katty | s/did/didn't/ |
| 17:46.49 | Naikrovek | must have |
| 17:46.58 | Naikrovek | everything goes over my head |
| 17:47.01 | Katty | 's cool |
| 17:47.06 | Katty | 's coo bro |
| 17:47.08 | Naikrovek | if it's not a technical joke i have no idea that you're making a joke |
| 17:47.21 | Katty | i'll clean your polycom in a minute? |
| 17:47.22 | Naikrovek | which is fortunate, i'm told that my feelings would hurt if i got them |
| 17:47.36 | Naikrovek | ... is that a clarification of the joke? |
| 17:47.39 | Naikrovek | i still don't get it |
| 17:47.46 | Katty | ehhehehe |
| 17:48.03 | Katty | that sir would be a sexual innuendo. |
| 17:48.07 | *** join/#asterisk [Outcast] (~anonymous@64.202.62.5) |
| 17:48.59 | n3hxs | checks Katty's pix again. |
| 17:49.01 | Naikrovek | really. |
| 17:49.10 | Katty | no, not really. |
| 17:49.11 | Katty | yes, but no. |
| 17:49.16 | Katty | ya know. |
| 17:49.16 | Naikrovek | a, but B |
| 17:49.19 | Naikrovek | got it, not really |
| 17:49.37 | Katty | well now that that's settled. |
| 17:49.54 | Naikrovek | :) |
| 17:49.59 | *** join/#asterisk wcselby (~wcselby@pdpc/supporter/professional/wcselby) |
| 17:50.03 | wcselby | o/ |
| 17:50.07 | Katty | :>>>>> |
| 17:50.10 | Katty | glomps wcselby |
| 17:50.15 | wcselby | lol |
| 17:50.47 | wcselby | you going to be at astricon next week Katty? |
| 17:50.47 | Cadey | anyone here from the UK ? |
| 17:50.47 | Katty | wcselby: sadly i am not |
| 17:50.51 | leifmadsen | is attending astricon, w00t |
| 17:50.52 | wcselby | Katty :( oh well, maybe next time |
| 17:51.03 | lirakis | sooo |
| 17:51.04 | wcselby | i got signed up last week, tmc called me to sign me up :) |
| 17:51.07 | lirakis | no info on the options behavior |
| 17:51.13 | lirakis | just ... cupcakes |
| 17:51.16 | lirakis | or whatever |
| 17:51.37 | Katty | one of the behavior options for asterisk is cupcakes?! |
| 17:52.15 | lirakis | one of the behaviours of this channel is cupcakes |
| 17:52.37 | Katty | i would say that is a valid statement. |
| 17:52.50 | wcselby | lirakis - what's the question, that was asked before I got here |
| 17:53.00 | n3hxs | wonders if Katty's other name is Cupcake. |
| 17:53.14 | n3hxs | or is that the puppy's name? |
| 17:53.15 | Katty | a cupcake by any other name would be just as sweet. |
| 17:53.16 | lirakis | wcselby, I want to know how asterisk handles options requests internally |
| 17:53.22 | lirakis | does it just respond 200 immediately |
| 17:53.24 | [TK]D-Fender | n3hxs: Not by anyone that still has their own teeth :p |
| 17:53.29 | lirakis | or does it check for user registration |
| 17:53.32 | lirakis | and respond accordingly |
| 17:53.38 | lirakis | ... it seems to be doing the former |
| 17:53.40 | Katty | my puppy's name is far from cupcake. it's Riddick |
| 17:53.59 | Katty | death by teacup. |
| 17:54.29 | wcselby | lirakis - i believe it confirms the user information first, but I haven't looked in a long time |
| 17:55.40 | lirakis | hrm .. yeah .. like i said, im testing it and it seems to respond 200 immediately. I will have to check it out though. I dont know what users are this box, but im getting 200 back for any thing i send it |
| 17:56.05 | wcselby | 200 is what, trying, or OK? |
| 17:56.09 | lirakis | 200 ok |
| 17:56.49 | wcselby | can you explain your scenario in a little more detail? |
| 17:57.02 | wcselby | you've got two asterisk boxes and you've setup qualify on the link between them, or? |
| 17:57.37 | Katty | scenario: step 1, drink lots of caffeine. step 2, eat cupcake. step 3, be unresposible for your giddy random behavior for the next 2 hours! |
| 17:58.36 | lirakis | wcselby, no. i have one asterisk box which i believe is insecure and a script I have written to enumerate users (by sending a options request) |
| 17:59.12 | lirakis | wcselby, I know i can enumerate users on asterisk by sending register requests and it will send 404 for users it doesnt know of |
| 17:59.16 | lirakis | but options is faster |
| 17:59.25 | lirakis | b/c it doesnt have to gen a nonce |
| 18:00.16 | lirakis | a lot of our users who use us for termination that manage their own pbx are getting hacked.. and nailed for $1k+ of termination. |
| 18:00.28 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.133.199.dsl.dyn.forthnet.gr) |
| 18:00.39 | lirakis | so .. we are doing "precrime" |
| 18:00.40 | lirakis | haha |
| 18:00.47 | wcselby | heh |
| 18:01.09 | wcselby | i'm not sure how an OPTIONS packet is going to show insecurity |
| 18:01.25 | wcselby | but I may just not be thinking along the right neurons right now |
| 18:01.30 | lirakis | wcselby, I have written a seperate script that will do brute force on digest authentication |
| 18:01.34 | lirakis | all i care about is enumerating users |
| 18:01.39 | lirakis | (at this point) |
| 18:01.47 | lirakis | my digest brute forcer works |
| 18:01.56 | lirakis | but i only want to spend time bruting users that I know exist |
| 18:02.15 | lirakis | I know sipvicious exists too .... i just want some thing a little more scriptable - which is why im writing my own |
| 18:02.20 | Naikrovek | you should learn about rainbow tables |
| 18:02.25 | wcselby | lirakis - have you looked at svwar.py? from sipvicious I think |
| 18:02.26 | lirakis | Naikrovek, that wont work |
| 18:02.32 | lirakis | Naikrovek, digest auth relies on a nonce |
| 18:02.36 | lirakis | so you cant precompute md5 |
| 18:02.36 | wcselby | lirakis - heh, nevermind then |
| 18:03.11 | Naikrovek | ah |
| 18:03.44 | [TK]D-Fender | [14:01]<lirakis>all i care about is enumerating users <- There is a way to always respond 401 so you can't enumerate |
| 18:04.25 | lirakis | [TK]D-Fender, right - but im guessing that is not setup on these customers |
| 18:04.37 | lirakis | i mean .. on registration .. just always send 401 |
| 18:04.41 | lirakis | thats what i do |
| 18:04.43 | [TK]D-Fender | lirakis: Quess? Why don't you KNOW? |
| 18:04.49 | *** join/#asterisk timahvo1 (~rogue@41.223.57.78) |
| 18:04.52 | lirakis | [TK]D-Fender, what? |
| 18:05.08 | [TK]D-Fender | Guess* |
| 18:05.48 | lirakis | oh ... yeah the Quess threw me ... but im not sure what im guessing about |
| 18:06.11 | lirakis | its not my job to manage their pbx's ... i just want to know if they are insecure.. |
| 18:06.55 | lirakis | at any rate ... ill go back to sending a reg. request. and detecting 404 before trying a digest brute. |
| 18:07.31 | lirakis | b/c i dont think asterisk responds to options requests "properly" |
| 18:10.05 | wcselby | how is it supposed to respond? |
| 18:10.52 | wcselby | i guess maybe I just don't see how an OPTIONS packet is supposed to enumerate users. it just seems to me like you're looking at the process backwards |
| 18:11.00 | lirakis | wcselby, its supposed to respond based on wether or not it can actually reach the user (per rfc) |
| 18:11.14 | lirakis | if i send OPTIONS to foo@bar.com |
| 18:11.56 | lirakis | bar.com should either ... forward the packet to the client and it will respond with what it supports, or asterisk should see if it has a registration for foo and reply to me with a generic set of options it supports |
| 18:12.01 | wcselby | so are you sending OPTIONS packets to asterisk boxes themselves, or to the peers connected to those asterisk boxes? |
| 18:12.12 | lirakis | wcselby, asterisk itself |
| 18:12.40 | wcselby | that was not my understanding of how OPTIONS packets worked, but I'll be the first to admit, I haven't done an in-depth stufy of them |
| 18:12.42 | lirakis | but it shouldnt say "everything is 200 OK" for a user it has no idea about. at any rate ... its not a big deal - i was just curious |
| 18:13.05 | lirakis | wcselby, yeah - the UAS is supposed to respond the options packet as if it were handiling an invite |
| 18:13.08 | wcselby | to me, it's responding saying that the box itself is 200 ok, not the end-user you're looking for |
| 18:13.16 | lirakis | wcselby, thats not how it works |
| 18:13.22 | lirakis | its not about the server. |
| 18:13.34 | wcselby | hmmmmm |
| 18:13.42 | lirakis | options are just .... sort of used in funny ways as a heart beat. they werent meant to be used for that. |
| 18:14.05 | lirakis | your phone would respond to an options packet with the methods it supports .... INVITE, REFER, CANCEL, etc etc. |
| 18:14.22 | lirakis | okay ... back to work... |
| 18:14.54 | wcselby | but you're not sending to the phone, you're sending to the asterisk box itself. it's not an INVITE packet, asterisk doesn't know to resend the packet down to the phone and respond with the answer. |
| 18:15.10 | wcselby | heh |
| 18:15.24 | lirakis | wcselby, it should respond the same as if it were an invite. it should look at the request URI |
| 18:15.29 | lirakis | and see how it would handle it |
| 18:16.06 | wcselby | i guess I just didn't realize that's how OPTIONS packets worked |
| 18:18.41 | [TK]D-Fender | * doesn't really ack OPTIONS packets IIRC. They are ALWAYS refused |
| 18:19.29 | lirakis | wcselby, "The response to an OPTIONS is constructed using the standard rules for a SIP response as discussed in Section 8.2.6. The response code chosen MUST be the same that would have been chosen had the request been an INVITE." |
| 18:19.53 | [TK]D-Fender | lirakis: RFC <- NO FUCKING COMMENT. |
| 18:19.56 | lirakis | just as an FYI |
| 18:20.07 | [TK]D-Fender | lirakis: Comlpiance = VOLUNTARY |
| 18:20.14 | [TK]D-Fender | ... "just FYI" |
| 18:20.26 | wcselby | that's not saying it forwards the request as though it received an invite, just that it responds with the same parameters that an INVITE would be responded with |
| 18:20.27 | lirakis | [TK]D-Fender, you MUST comply unless you SHOULD |
| 18:20.28 | lirakis | lol |
| 18:20.47 | lirakis | wcselby, i didnt say it has to forward the request |
| 18:20.53 | lirakis | just respond as if it were an invite |
| 18:21.06 | lirakis | and ... if you sent an invite to an unknown extension... * would respond with a 404 |
| 18:21.21 | lirakis | [TK]D-Fender, come on ... that was funny |
| 18:21.34 | lirakis | MUST unless you SHOULD .... |
| 18:21.37 | lirakis | tough crowd |
| 18:22.14 | wcselby | no, that's not the way I'm reading it |
| 18:22.29 | wcselby | ah hell, maybe it is |
| 18:22.34 | wcselby | i hate reading RFCs |
| 18:22.48 | lirakis | (shrug) |
| 18:23.04 | lirakis | asterisk definitely ... "does its own thing" .. |
| 18:23.26 | [TK]D-Fender | lirakis: As coppice stated SIP implies lack of compliance |
| 18:23.39 | lirakis | [TK]D-Fender, its definitely a wacky world |
| 18:26.26 | *** join/#asterisk seanjohn (~seanjohn@gateways.sheltoncomputers.com) |
| 18:28.08 | Naikrovek | lack of compliance should not be tolerated |
| 18:28.19 | Naikrovek | or, if you aren't compliant, you can't SAY you're compliant |
| 18:29.09 | Naikrovek | with ISO and ANSI this is enforced, as i understand it. RFCs haven't a governing body |
| 18:31.29 | lirakis | d/ling the source to see what goes on under the hood |
| 18:31.36 | Naikrovek | right |
| 18:31.39 | Naikrovek | that should not be required |
| 18:31.43 | Naikrovek | ideally |
| 18:31.53 | Naikrovek | SIP = SIP = SIP in my fantasyland |
| 18:32.18 | Naikrovek | when i go crazy and start seeing things, i'm going to see pleasant things like standards that are fucking adhered to |
| 18:32.49 | *** join/#asterisk fullstop (~fullstop@static-173-210-91-3.ngn.onecommunications.net) |
| 18:33.39 | *** join/#asterisk rfuster (~mjagger@67.159.45.46) |
| 18:33.44 | lirakis | interesting ... looks like they now are authenticating options req. at least 1.8 |
| 18:34.07 | fullstop | I have two servers set up, with extensions on both.. connected with iax2. Is there any way to set up a sip user on system B which has a voice mail box on system A? |
| 18:34.31 | fullstop | and still have the nice things such as message waiting indicators function? |
| 18:34.43 | *** join/#asterisk MACscr (~macscr@c-98-214-103-147.hsd1.il.comcast.net) |
| 18:35.44 | lirakis | fullstop, setup up 3 systems ... one as your voicemail server |
| 18:35.58 | rfuster | hello i have an asterisk server and i'm doing call termination prepaid, i'm trying to do the billing myself, but i need to know how to calculate the profit from the money my customer gaves me and the money i have to pay to the provider from that |
| 18:36.04 | rfuster | any ideas please? |
| 18:36.04 | lirakis | and route all calls to ... whatever your VM exten is to the 3rd system |
| 18:36.28 | fullstop | lirakis: As silly as this is, I'm looking to get the MWI light working. |
| 18:36.43 | fullstop | lirakis: The set can already call in and get the VM.. |
| 18:36.46 | MACscr | anyone using a wireless headset with their cisco 504g phones? Just got a few plantronics savi wo100 headsets and it appears they dont have a 2.5mm plug, so trying to figure out how to connect them to the phone without a handset lifter. Wondering if its possible to just go from rj9 to 2.5mm. Wonder if that would work. |
| 18:36.48 | Naikrovek | rfuster: duration of call in minutes * (what you're paid per minute - what you pay per minute) |
| 18:37.48 | Naikrovek | handset lifters are one giant warning sign about how things are designed. a "you're designing it wrong" icon, if you will |
| 18:37.56 | Naikrovek | they're necessary because of poor design, i'm saying |
| 18:37.57 | Naikrovek | anyway |
| 18:38.16 | Naikrovek | i don't know if 2.5mm <--> rj9 adapters exist, but i'm thinking they do |
| 18:38.31 | wcselby | fullstop - there's a parameter in voicemail.conf to run a script after voicemail is left, you can use that have the voicemail on server A notify the user on server B |
| 18:38.52 | rfuster | Naikrovek: could you be more specific? i'm doing prepaid and i don't know the duration of the call beforehand |
| 18:39.12 | Naikrovek | seems like this is something you would have worked out beforehand. |
| 18:39.22 | lirakis | HAHA how funny ... right in the source code above the handle_request_options function |
| 18:39.28 | Naikrovek | no one knows the duration of a call until the call is over, btw |
| 18:39.32 | lirakis | " An OPTIONS request should be answered like an INVITE from the same UA, including SDP" |
| 18:39.32 | *** join/#asterisk silvestre_id (~silvestre@189-19-248-200.dsl.telesp.net.br) |
| 18:39.34 | MACscr | Naikrovek: i know they exist, im just wondering if they would technically work. I would think so |
| 18:39.43 | Naikrovek | MACscr: yeah i would imagine so |
| 18:39.54 | rfuster | Naikrovek: they pay me and i need to put the money on their account and i have to nibble bill from that until they don't have credit and cut the call when it reaches 0 |
| 18:40.08 | Naikrovek | rfuster: ah prepaid okay |
| 18:40.20 | wcselby | rfuster - maybe you should look at a2billing, I think it does this right out of the box |
| 18:40.28 | Naikrovek | yeah i was in the middle of typing that |
| 18:40.31 | Naikrovek | what wcselby said |
| 18:40.41 | wcselby | what Naikrovek almost said |
| 18:40.43 | wcselby | :) |
| 18:40.45 | Naikrovek | lol |
| 18:40.55 | wcselby | never did get the new friend invite on steam |
| 18:40.58 | Naikrovek | technically correct. THE BEST KIND OF CORRECT |
| 18:41.04 | Naikrovek | wcselby: WHAT?! |
| 18:41.05 | Naikrovek | okay |
| 18:41.08 | wcselby | heh |
| 18:41.12 | wcselby | i wasn't just ignoring you |
| 18:41.12 | wcselby | :P |
| 18:41.15 | lirakis | yeah ill need to get a older version of the code.... this new stuff is forcing auth for any initial options request |
| 18:42.00 | fullstop | wcselby: can you explain a little bit more? What would the script do? That is, how would you notify? |
| 18:42.55 | wcselby | fullstop - i did this once with when I integrated an asterisk box to be the VM server for a cisco call manager. the script would trigger something on the call manager to indicate MWI |
| 18:43.00 | Naikrovek | wcselby: francis icon? |
| 18:43.09 | *** join/#asterisk deltalytic (~haryv@154.5.144.132) |
| 18:43.10 | wcselby | Naikrovek - yeah, I changed it recently |
| 18:43.28 | fullstop | wcselby: I'll look into this. Thanks. |
| 18:43.51 | Naikrovek | wcselby: added. i got l4d and l4d2 on your recommendation |
| 18:43.54 | *** join/#asterisk netrat (~netrat@2001:470:8:479::3) |
| 18:43.57 | Naikrovek | very happy with those purchases, btw |
| 18:44.08 | wcselby | Naikrovek - heh, sweet |
| 18:44.18 | *** join/#asterisk oej (~olle@2001:470:1f15:d79:225:ff:fe44:74ec) |
| 18:44.22 | fullstop | Naikrovek: Hopefully you got them for cheap.. they were on sale recently. |
| 18:44.32 | wcselby | fullstop - the option was called externnotify, btw |
| 18:44.32 | Naikrovek | yeah like $10.20 for both plus both DLCs |
| 18:44.46 | Naikrovek | steam sales are really hard to pass up |
| 18:44.53 | fullstop | Naikrovek: The DLCs are free, but $10.20 is almost giving it away. |
| 18:45.03 | Naikrovek | oh they are? nice |
| 18:45.18 | fullstop | Naikrovek: I believe we are "friends" on steam. I'm "Statler" |
| 18:45.18 | wcselby | fullstop, Naikrovek - lol, yeah, they went on sale TWO DAYS after I talked another buddy into buying them |
| 18:45.21 | wcselby | he wasn't pleased |
| 18:45.24 | Naikrovek | d'oh |
| 18:45.30 | deltalytic | any of you use asterisk gui? I think I came across some code with a misconfiguration in it. |
| 18:45.35 | Naikrovek | fullstop: ah yeah we are |
| 18:45.46 | deltalytic | more like wrong code in the html page |
| 18:45.47 | deltalytic | ;) |
| 18:45.48 | Naikrovek | deltalytic: dont' think asteriskgui is mantained anymore |
| 18:46.00 | Naikrovek | but there is an #asterisk-gui channel I think |
| 18:46.09 | deltalytic | I think you are right. I have recived almost no replies in the forums |
| 18:46.25 | Naikrovek | that's usually a good indicator |
| 18:46.35 | Naikrovek | #trixbox is turning into a ghost town as well, thankfully |
| 18:46.37 | wcselby | Naikrovek - yeah it's still maintained, they added the latest release to AN1.7 |
| 18:46.54 | Naikrovek | eats his hat |
| 18:46.55 | wcselby | i remember talking to the guy who maintains it at astricon last year |
| 18:47.07 | wcselby | but i think it may just be one or two guys that maintain it, along with other projects |
| 18:47.10 | wcselby | so it's not heavily worked on |
| 18:47.18 | wcselby | since so many people seem to have moved to freepbx |
| 18:47.21 | deltalytic | BTW is there a up and running asterisk server that I can test my softphone here? The new ISP we swiched to seems to not allow my asterisk box to register. |
| 18:47.23 | Naikrovek | yeah |
| 18:48.11 | deltalytic | Ie, sip/rtp not passing though even though, I have port forwading configured on the telus router |
| 18:48.48 | lirakis | hmmmmm ... interesting ... it does call a res=get_destination() .... which would presumably lookup the user in the registration .... and then it responds based on res..... off to get_destination |
| 18:49.15 | wcselby | fullstop - here is where I got the basic idea for the externnotify script that triggers the MWI --> http://www.voip-info.org/wiki/view/Asterisk+Cisco+CallManager+Voicemail+Integration#comment_43861 |
| 18:49.24 | lirakis | deltalytic, just try free world dial up |
| 18:49.25 | wcselby | it's old, but you should be able to adapt it I would think |
| 18:50.10 | fullstop | wcselby: Thanks. It gives me something to start with. |
| 18:50.14 | wcselby | np |
| 18:50.15 | deltalytic | I have always edited the config files by hand but some times, it would be nicer just to edit via a gui once in a while especially when I am busy. BTW, people on #html page said the asterisk gui interface is riddles with errors. |
| 18:50.35 | lirakis | deltalytic, screw the gui |
| 18:50.39 | deltalytic | hehe |
| 18:50.52 | *** part/#asterisk rfuster (~mjagger@67.159.45.46) |
| 18:51.05 | deltalytic | I have never found one I liked |
| 18:52.15 | netrat | lirakis, is free world dialup still around? |
| 18:52.23 | lirakis | nemik, i dunno |
| 18:52.27 | lirakis | i think so yeah |
| 18:52.30 | lirakis | or .. pbxes |
| 18:52.36 | lirakis | some free sip service |
| 18:52.46 | Kobaz | http://imgur.com/SpNUx |
| 18:53.02 | lirakis | aha ... |
| 18:53.04 | lirakis | got it |
| 18:53.07 | Kobaz | http://i.imgur.com/SpNUx.jpg |
| 18:53.08 | lirakis | so .. asterisk is cheating a little |
| 18:53.22 | lirakis | instead of looking up the registration, it checks the dialplan |
| 18:53.37 | lirakis | and if it finds a valid context and extension it says everything is ok |
| 18:53.43 | lirakis | regardless of if the user is on the system |
| 18:55.04 | lirakis | ... so there you go ... for any one wondering about how asterisk handles options requests .. thats it. ha ha |
| 18:55.25 | lirakis | (well there is a little more in there for sequential requests - and newer version auth.. but thats basically it) |
| 18:55.40 | *** join/#asterisk timahvo1 (~rogue@41.223.57.77) |
| 18:58.07 | *** join/#asterisk jkroon (~jkroon@dsl-242-8-144.telkomadsl.co.za) |
| 18:58.18 | Naikrovek | what do you guys use for monitoring bandwidth usage |
| 18:58.44 | Naikrovek | last i think i used manageengine but if there's something better i would like to know before i start implementation |
| 18:58.45 | wcselby | a ruler |
| 18:58.53 | wcselby | oh, sorry |
| 18:59.15 | lirakis | bwm-ng |
| 18:59.16 | jkroon | Naikrovek, depends on what you want to measure. |
| 18:59.22 | Naikrovek | jkroon: netflow |
| 18:59.29 | Naikrovek | and general utilization |
| 18:59.50 | MACscr | Naikrovek: found polycom 2.5mm to rj9 adapters for $10 each. Much better than $50 for crappy handset lifters =P |
| 18:59.56 | Naikrovek | here here |
| 19:00.04 | Naikrovek | headset lifters are part of the problem, man |
| 19:00.30 | Naikrovek | managengine is pretty good |
| 19:00.35 | Naikrovek | probably just use that |
| 19:00.46 | jkroon | general utilization something like mrtg is quite useful, netflow ... well, i use a custom cooked solution for that because at the time nothing out there did what i wanted. |
| 19:00.49 | jkroon | not sure about now. |
| 19:00.55 | Naikrovek | jkroon: okay |
| 19:01.18 | Naikrovek | at yahoo we had a guy writing an app for netflow but he never finished it that I know of |
| 19:01.26 | Naikrovek | had a cool name too, but i forget what it is now |
| 19:01.45 | jkroon | hehe |
| 19:01.47 | deltalytic | Naikrovek what was your job at yahoo? |
| 19:02.03 | *** part/#asterisk lirakis (~lirakis@ool-ad022bb1.dyn.optonline.net) |
| 19:02.06 | Naikrovek | i was an ops manager for the southeast asia region |
| 19:02.12 | Naikrovek | i worked out of sydney, australia |
| 19:02.14 | *** join/#asterisk timeshell_atwork (~chatzilla@gw.lusi.on.ca) |
| 19:02.23 | deltalytic | nice location I suspect |
| 19:02.24 | Naikrovek | ops engineer/ops manager i forget now, actually |
| 19:02.29 | Naikrovek | there was only one of me so i guess both work |
| 19:02.34 | Naikrovek | yeah it was |
| 19:03.11 | wcselby | Naikrovek - i think solarwinds has some tools you could look at |
| 19:03.23 | Naikrovek | i was the sole datacenter guy. i was responsible for picking datacenter locations, deploying to them, maintaining to them |
| 19:03.26 | wcselby | for both netflow and utilization, etc |
| 19:03.36 | Naikrovek | telstra loved us, they painted our cage yellow and our racks purple (yahoo colors) |
| 19:03.43 | Naikrovek | wcselby: okay thanks |
| 19:04.33 | Katty | HHHHHHHHHELLLLLLOOOOOOOOOOOOO Nurse. |
| 19:04.44 | beek | hugs Katty |
| 19:04.59 | Katty | hugs on beek |
| 19:05.17 | deltalytic | Naikrovek I could have been your infrastuture person :) That has been more my calling in the past. Install and hookup. |
| 19:05.54 | Naikrovek | yeah i did all that. went to training in hong kong for the robotic tape backup and stuff like that |
| 19:06.07 | deltalytic | ahh nice |
| 19:06.10 | Naikrovek | tape robots the size of small ISO containers |
| 19:06.18 | deltalytic | nice |
| 19:06.22 | Naikrovek | netapp filers and whatnot |
| 19:06.24 | z4nD4R | nobody can help with sip over tls? |
| 19:06.37 | Naikrovek | z4nD4R: you are GOING TO HAVE TO phrase that differently if you want help |
| 19:06.51 | Naikrovek | "sip over tls" gives us NOTHING to work with |
| 19:07.13 | deltalytic | I prefer the physical hands on then say, changing code. Like the outdoors, moving around gear/equipment logistics working with GCs and such. |
| 19:07.18 | Naikrovek | we need error logs and configuration details to begin |
| 19:07.38 | z4nD4R | Naikrovek: ok |
| 19:07.44 | deltalytic | Just to bad I do not have my electricans licence then can do it all. |
| 19:08.06 | z4nD4R | tls handshake failed after clienthello message - somebody can help if.. i put here (pb ) my sip.conf |
| 19:08.19 | deltalytic | Did assist a DC power GC in the installation of 48 450 bls batteries in a telco once. |
| 19:08.32 | Naikrovek | nice! ever see any flywheel batteries |
| 19:08.55 | deltalytic | I have read about kentic energy storage in flywheel technoligy. |
| 19:09.05 | Naikrovek | yeah, on paper it's almost ideal |
| 19:09.07 | deltalytic | It is perhaps the most efficient form of storage? |
| 19:09.10 | Naikrovek | in practice, i dunno |
| 19:09.33 | Naikrovek | claims to be, yes, more efficient than any battery |
| 19:09.41 | deltalytic | Have the rotor floating on Neomagnets in a vacume and ballances to precise measurments. |
| 19:10.31 | deltalytic | I imagine the case its mounted in is pretty thick steel to prevent twisting. |
| 19:10.58 | Naikrovek | what would make it twist |
| 19:11.10 | deltalytic | quake |
| 19:11.20 | *** join/#asterisk deonv (~adium@196.1.28.226) |
| 19:11.25 | Naikrovek | oh that would probably make the flywheel shatter |
| 19:11.30 | deltalytic | true |
| 19:11.41 | Naikrovek | carbon fiber flywheels would not do much damage, they would turn to dust |
| 19:11.48 | Naikrovek | metal, on the other hand, would do some damage |
| 19:11.51 | deltalytic | unless there is a p wave detection system to put on its brakes really quick |
| 19:12.10 | deltalytic | Ohh make the box float on shocks |
| 19:12.15 | deltalytic | and springs ;) |
| 19:12.23 | Naikrovek | yeah i think they do that |
| 19:13.23 | deltalytic | I have seen carbonfiber flywheels in pics before before, and after the unit shatered into a ball of fibers. |
| 19:14.15 | *** join/#asterisk dmast (~dmast@exchange.newpointe.org) |
| 19:14.22 | fullstop | is sip_notify.conf documented anywhere? Is it intended just to send static notify messages to sip clients? |
| 19:14.35 | deltalytic | anyway, I need to test my router for sip/rtp passthough. does anyone know of a site with such a test server? |
| 19:14.57 | Naikrovek | deltalytic: you could set one up on an amazon EC2 server pretty quickly |
| 19:14.59 | deltalytic | full, should be lots of info online. |
| 19:15.12 | deltalytic | ? |
| 19:15.20 | deltalytic | what is a ec2 server? |
| 19:15.24 | Naikrovek | set up a virtual machine on amazon ec2 |
| 19:15.26 | fullstop | elastic cloud |
| 19:15.28 | Naikrovek | then connect your phone to it |
| 19:15.34 | deltalytic | yes virtual |
| 19:15.34 | deltalytic | ;) |
| 19:15.43 | Naikrovek | unless $0.03 per hour is too expensive for you |
| 19:15.45 | deltalytic | probebly true |
| 19:15.50 | deltalytic | hehe |
| 19:18.03 | deltalytic | trying iptel |
| 19:23.17 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 19:28.23 | *** join/#asterisk mercutioviz (~michaelco@freeswitch/developer/msc) |
| 19:28.49 | *** part/#asterisk mercutioviz (~michaelco@freeswitch/developer/msc) |
| 19:31.25 | leifmadsen | http://ofps.oreilly.com |
| 19:31.42 | leifmadsen | I love how people will just click anything |
| 19:31.42 | *** join/#asterisk unspin (~unspin@S0106001451226d9c.vc.shawcable.net) |
| 19:31.48 | russellb | leifmadsen: ! |
| 19:31.49 | z4nD4R | Guys, i what to use sip over tls on asterisk 1.8 rc3... tls is enable ( server listen on port 5061 ) but registration users failed .. becauseee tls handshake failure.... |
| 19:31.49 | z4nD4R | i read in tls RFC that is bacause server do not match witch any cipher which client offer... here my sip.conf > http://pastebin.com/d7mkggem .. can somebody help??? |
| 19:32.03 | russellb | ~thebook |
| 19:32.03 | infobot | it has been said that thebook is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://astbook.asteriskdocs.org |
| 19:32.17 | leifmadsen | russellb: are you trying to trump me with multipel links? |
| 19:32.19 | russellb | infobot: thebook is also http://ofps.oreilly.com |
| 19:32.19 | infobot | okay, russellb |
| 19:32.31 | leifmadsen | oh snap |
| 19:32.31 | russellb | leifmadsen: no. |
| 19:32.39 | leifmadsen | I see what you're doing there |
| 19:34.22 | unspin | i'm in the process of setting up some kind of connection quality monitoring to diagnose some quality problems we have in our Asterisk network and i haven't found a good free solution to gather network quality data |
| 19:34.37 | unspin | something that would emulate an rtp stream and graph the result |
| 19:35.14 | unspin | i played with smokeping for a bit, but its not really designed for sending 33/55 pps to several different destinations and giving accurate graph results |
| 19:35.59 | unspin | does anyone have any recommendations? i dont want to code my own front-end to rrdtool ;P |
| 19:36.07 | dmast | I've been trying to get unsolicited mwi working with 1.8 RC4 and Exchange 2010. Right now, when I leave a message in a mailbox, the mwi notification comes through, but it lights the mwi on both of my test phones instead of just the phone I left the voicemail message on. Is there a piece of config that I'm missing in order to get this working properly? |
| 19:37.55 | *** join/#asterisk jetlag (~jetlag@pool-173-61-243-204.cmdnnj.east.verizon.net) |
| 19:38.34 | unspin | mos scoring would be awesome, but i suppose thats even more of a long shot without a budget |
| 19:40.01 | *** join/#asterisk r0d3nt (~astrutt@cheshire.telephreak.org) |
| 19:40.21 | leifmadsen | unspin: sounds awesome, but I'm afraid you're likely to find you'll have to create such a tool |
| 19:42.51 | z4nD4R | Guys, i what to use sip over tls on asterisk 1.8 rc3... tls is enable ( server listen on port 5061 ) but registration users failed .. becauseee tls handshake failure.... i read in tls RFC that is bacause server do not match witch any cipher which client offer... here my sip.conf > http://pastebin.com/d7mkggem .. can somebody help??? |
| 19:46.54 | *** join/#asterisk c0rnoTa (~c0rnoTa@80.251.113.55) |
| 19:48.10 | kaldemar | z4nD4R: you have 2 [general] contexts in your sip.conf. don't. |
| 19:48.57 | z4nD4R | kaldemar: ok this is the problem? |
| 19:49.00 | kaldemar | z4nD4R: you also seem to have tcpenable=yes in the first [general]. |
| 19:49.20 | kaldemar | i don't know if that spcifically is your problem, but it is a problem. |
| 19:50.01 | pabelanger | z4nD4R: and delete the commented out lines. Expecting somebody to read thru +200 lines is crazy |
| 19:50.44 | z4nD4R | kaldemar: ok... right .. i try it |
| 19:50.54 | z4nD4R | w8 for me results :) |
| 19:51.43 | *** part/#asterisk c0rnoTa (~c0rnoTa@80.251.113.55) |
| 19:54.21 | *** join/#asterisk diegomad (~mad@190.147.221.138) |
| 19:54.52 | diegomad | hi, how to generate a syscompat event in queuelog? |
| 20:00.55 | z4nD4R | kaldemar: http://pastebin.com/WvWi1P1d now my sip.conf... but the nothings happen |
| 20:01.06 | z4nD4R | stil tls does not work |
| 20:02.11 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.81) |
| 20:03.09 | kaldemar | z4nD4R: general is supposed to be the first context in the file. |
| 20:03.40 | z4nD4R | ok |
| 20:04.43 | *** join/#asterisk TJ_stamm (~TJ@69.197.107.254) |
| 20:05.11 | z4nD4R | kaldemar: done... again..nothing |
| 20:06.01 | unspin | woah |
| 20:06.06 | unspin | so |
| 20:06.28 | unspin | mayhaps there is a free/open source solution to creating mos scores for sip/rtp calls |
| 20:06.57 | Qwell | computers cannot make proper opinions. |
| 20:07.05 | unspin | as a followup to my question earlier, http://www.voipmonitor.org/ seems to be able to generate mos scores for live traffic as well as pcaps |
| 20:07.07 | Qwell | (that's what the 'o' in 'mos' stands for..) |
| 20:07.30 | unspin | sure, but its just one tool of many |
| 20:12.00 | frigidzephyr | computer cannot make proper opinions? |
| 20:12.05 | frigidzephyr | can they make improper ones? |
| 20:12.13 | Qwell | sure |
| 20:17.36 | *** part/#asterisk diegomad (~mad@190.147.221.138) |
| 20:17.50 | [TK]D-Fender | WHAT ARE YOU DOING DAVE? |
| 20:18.57 | frigidzephyr | I was... just trying to convert these SIP calls to RJ-45.... |
| 20:19.50 | [TK]D-Fender | jettisons frigidzephyr out the airlock |
| 20:21.06 | Qwell | frigidzephyr: SIP only works over Qwell Communications CAT6ezomg cabling. |
| 20:21.25 | frigidzephyr | Does that support faxes? |
| 20:21.40 | Qwell | It supports FAX over email. |
| 20:22.55 | [TK]D-Fender | checkout time, BBL |
| 20:24.50 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 20:26.00 | z4nD4R | Guys, i what to use sip over tls on asterisk 1.8 rc3... tls is enable ( server listen on port 5061 ) but registration users failed .. becauseee tls handshake failure....i read in tls RFC that is bacause server do not match witch any cipher which client offer... here my sip.conf > http://pastebin.com/WvWi1P1d .. can somebody help??? |
| 20:27.54 | ariel_ | ~book |
| 20:27.54 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 20:29.11 | russellb | infobot: book is also http://ofps.oreilly.com/ |
| 20:29.11 | infobot | russellb: okay |
| 20:34.39 | *** join/#asterisk jimi_ (~jimi@unaffiliated/tuxguy) |
| 20:34.55 | jimi_ | Using the chanspy functions, is it possible to whisper not on the local side, but on the remote side? |
| 20:35.19 | z4nD4R | kaldemar: some ideas? |
| 20:35.58 | *** join/#asterisk chuckp (~chuckp@c-76-106-198-76.hsd1.fl.comcast.net) |
| 20:36.31 | jimi_ | Basically, i want to whisper to a customer without the local agent knowing |
| 20:36.59 | chuckp | Anyone able to point me in the general direction of fixing speakerphone issues? Whenever calls are on speaker the calls get poping sounds if the party on speaker does not talk and does not mute their phone. |
| 20:37.18 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 20:37.19 | *** join/#asterisk DrDigi (~mmurphy@75-146-139-38-Stockton.hfc.comcastbusiness.net) |
| 20:37.33 | leifmadsen | jimi_: I don't think asterisk cares too much -- you just have to configure it to whisper to the channel you want |
| 20:38.19 | jimi_ | leifmadsen, ok, i will look at that... usually i just chanspy an extension |
| 20:38.44 | leifmadsen | jimi_: just assign a group to the incoming callers and then you can whisper just to them |
| 20:38.50 | leifmadsen | I'm sure there are a few ways tos olve that issue |
| 20:39.04 | jimi_ | i want to whisper to a customer who is currently talking to an agent |
| 20:39.22 | jimi_ | like barge, but muting the agent ... but the agent doesnt hear me |
| 20:39.26 | jimi_ | i am disgruntled |
| 20:39.35 | chuckp | lol |
| 20:39.42 | jimi_ | :) |
| 20:40.10 | chuckp | agent is going to think the person is crazy |
| 20:40.26 | jimi_ | chuckp, ah, you get it |
| 20:40.31 | jimi_ | chuckp++ |
| 20:42.01 | jimi_ | chuckp, i wanna play tt-weasels to the customer so the agent doesnt know |
| 20:42.13 | chuckp | you are disgruntled |
| 20:42.15 | chuckp | :P |
| 20:42.21 | jimi_ | sling blade quotes, etc |
| 20:43.11 | chuckp | need some "The Rock" quotes |
| 20:43.18 | jimi_ | hell yeah |
| 20:43.29 | chuckp | "Id take pleasure in gutten you boy!" |
| 20:43.37 | chuckp | sure that will go over well |
| 20:43.46 | jimi_ | "You aught not done that to him, he was just a boy" |
| 20:45.05 | chuckp | slingblade? |
| 20:45.54 | doolittlework | what gain should i set for increasing the volume for outgoing tx or rx and to how much |
| 20:47.30 | jimi_ | can we dynamically dynamic increase or decrease the volume for outgoing sound |
| 20:47.32 | jimi_ | ? |
| 20:48.33 | chuckp | have to be able to, it can be done in ChanSpy |
| 20:48.54 | chuckp | your not going to play something to blow people's ear drums out are you? |
| 20:49.04 | *** join/#asterisk erinspice (~erin@207.98.195.107) |
| 20:49.11 | jimi_ | just an increasing and decreasing buzz |
| 20:49.15 | jimi_ | on the customer's side |
| 20:49.29 | chuckp | you should play that tone that only people under 25 can hear |
| 20:49.44 | chuckp | *frequency |
| 20:51.52 | wcselby | wtf...i'm trying to install cdr-stats, and the install instructions simply say "install cdr-stats, then configure" |
| 20:51.57 | wcselby | but no where does it say HOW to install it |
| 20:52.05 | wcselby | i suppose I'm blind today |
| 21:04.57 | *** join/#asterisk [netman] (~netman@83.37.61.17) |
| 21:09.24 | z4nD4R | Guys, i what to use sip over tls on asterisk 1.8 rc3... tls is enable ( server listen on port 5061 ) but registration users failed .. becauseee tls handshake failure....i read in tls RFC that is bacause server do not match witch any cipher which client offer... here my sip.conf > http://pastebin.com/nVfMUZ63 .. can somebody help??? |
| 21:18.30 | *** join/#asterisk Orbixx (~Orbixx@213.229.88.67) |
| 21:18.49 | *** join/#asterisk cm_ (~chris@2a01:e35:2e8b:fa50:21e:64ff:fe0b:c46c) |
| 21:19.43 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 21:20.29 | Orbixx | I've installed Asterisk, both the latest RC and the latest trunk and on installation, neither of them load a handful of modules, including the channel driver for IAX. |
| 21:20.46 | Orbixx | Even though the module exists and exists in the location Asterisk references. |
| 21:21.12 | z4nD4R | gn |
| 21:21.37 | Qwell | Orbixx: Check in `make menuselect` |
| 21:21.51 | Orbixx | It's selected. |
| 21:21.57 | Orbixx | I made sure of that. |
| 21:22.10 | Qwell | then it will be built |
| 21:22.39 | Orbixx | It looks built to. |
| 21:22.42 | Orbixx | But Asterisk still fails to load it. |
| 21:22.50 | Qwell | what is the error? |
| 21:23.34 | Orbixx | [Oct 20 17:13:31] WARNING[14780]: loader.c:449 load_dynamic_module: Error loading module 'chan_iax2.so': /usr/lib/asterisk/modules/chan_iax2.so: undefined symbol: ast_aes_set_decrypt_key |
| 21:24.13 | Qwell | Orbixx: Whats the output of core show version ? |
| 21:24.48 | Orbixx | Asterisk SVN-trunk-r292377 |
| 21:25.12 | Orbixx | Also tried the latest RC as of today. |
| 21:25.26 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 21:26.51 | Orbixx | On a fresh OS, no less. |
| 21:27.20 | Orbixx | So it's not an old module from a prior build or anything. |
| 21:28.09 | pabelanger | Orbixx: res_crypto install and loaded? |
| 21:28.37 | Qwell | pabelanger: it should handle it gracefully |
| 21:28.58 | pabelanger | <PROTECTED> |
| 21:29.04 | Orbixx | pabelanger: Does not exist. |
| 21:29.39 | fullstop | Can I set IAX channel variables / headers to pass small amounts of information between two asterisk servers? |
| 21:29.47 | Qwell | russellb: ping! |
| 21:29.59 | russellb | Qwell: pong! |
| 21:30.02 | Qwell | see above |
| 21:30.32 | russellb | ummm |
| 21:30.34 | russellb | huh |
| 21:30.39 | Qwell | exactly O.o |
| 21:30.52 | russellb | what's in modules.conf? |
| 21:31.15 | Qwell | it should still load it fine though, right? |
| 21:31.22 | russellb | it ... should ... i think, yes |
| 21:31.23 | Qwell | even if crypto stuff isn't there, that is |
| 21:31.38 | russellb | but if it thought it was there at build time, but then it's not at run time maybe ... |
| 21:31.41 | russellb | not sure |
| 21:32.05 | Qwell | hmm, maybe. |
| 21:32.07 | Qwell | Orbixx: what OS? |
| 21:32.25 | Qwell | it might be one of those sucky ones without the support for optional_API |
| 21:35.09 | Katty | HAI QWELL |
| 21:35.14 | Qwell | ohai |
| 21:35.41 | Katty | ohio |
| 21:36.17 | Katty | i'm starving |
| 21:36.26 | Qwell | that is fixable |
| 21:36.34 | Katty | not while i'm at work it's not :< |
| 21:36.37 | Katty | and then i gots to go work out |
| 21:36.42 | Orbixx | Qwell: CentOS |
| 21:36.51 | Qwell | what version? |
| 21:37.05 | pabelanger | Qwell: shouldn't chan_iax2 use res_crypto, not crypto? |
| 21:37.07 | pabelanger | http://pastebin.com/pHPCy6wP |
| 21:37.12 | pabelanger | like chan_sip |
| 21:37.18 | Qwell | umm |
| 21:37.35 | russellb | probably. |
| 21:37.48 | pabelanger | I'm testing it now |
| 21:37.49 | russellb | unless it uses the libcrypto API directly .... |
| 21:38.02 | Qwell | no it uses ast_aes_* |
| 21:39.34 | pabelanger | Orbixx: give this a try, if your up for it |
| 21:39.36 | russellb | if it uses the res_crypto APIs, and res_crypto wasn't built, and it's a system that doesn't support the optional API stuff, then that would explain the problem. |
| 21:39.41 | pabelanger | http://pastebin.com/pHPCy6wP |
| 21:41.55 | Orbixx | pabelanger: I'll give it a shot, thanks. |
| 21:42.24 | Qwell | Orbixx: You'll probably have to do a make clean for it to work |
| 21:42.24 | pabelanger | Orbixx: trying to reproduce it locally now |
| 21:43.15 | *** join/#asterisk mpe (~mpe@0xd99d3f8f.customer.cybercity.dk) |
| 21:51.31 | fullstop | so I found IAXVAR, which appears to be what I want.. but I can't seem to get it to work. |
| 21:52.13 | fullstop | If asterisk receives a call from SIP and then passes that call along over IAX, I should be able to use IAXVAR() to set a variable for the channel, right? |
| 21:52.34 | fullstop | core show function IAXVAR shows that I have it, but I get an error when I try to use it.. |
| 21:53.09 | fullstop | No application 'IAXVAR' .. So I must be doing something dumb. Any ideas? |
| 21:53.32 | Orbixx | pabelanger: No change. |
| 21:54.07 | pabelanger | Orbixx: Heh, still trying to reproduce it |
| 21:55.59 | frigidzephyr | fullstop: Set(IAXVAR(BLAH)=SOMETHING) |
| 21:56.06 | frigidzephyr | fullstop: its a function, not an application |
| 21:56.26 | fullstop | frigidzephyr: just figured that out. Thanks! |
| 21:57.28 | frigidzephyr | no prob bob |
| 22:07.49 | pabelanger | Orbixx: Can't reproduce it, you'll need to open an issue on the tracker, post OS, modules.conf, etc |
| 22:09.19 | pabelanger | Orbixx: config.log too |
| 22:09.32 | *** join/#asterisk scottsmith7 (~scottsmit@64.201.141.80) |
| 22:10.06 | *** part/#asterisk scottsmith7 (~scottsmit@64.201.141.80) |
| 22:10.42 | *** join/#asterisk ManxPower (~manxpower@user-24-214-153-32.knology.net) |
| 22:12.21 | ManxPower | I'm having problems getting cdr_mysql to log the uniqueid. I have MYSQL_LOGUNIQUEID #defined at the top of the source file, made clean, make && make install, confirmed the file timestamp on the module is correct. Does anyone have any ideas? |
| 22:12.37 | ManxPower | (and restarted, not reloaded asterisk) |
| 22:13.31 | ManxPower | oh, and Asterisk 1.4.recent |
| 22:14.33 | ManxPower | the uniqueid field exists in the table |
| 22:19.54 | *** join/#asterisk scottsmith7_ (~scottsmit@64.201.141.80) |
| 22:20.27 | *** part/#asterisk scottsmith7_ (~scottsmit@64.201.141.80) |
| 22:26.00 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 22:43.47 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 22:44.53 | *** join/#asterisk bbryant (~brett@c-76-26-221-79.hsd1.sc.comcast.net) |
| 22:50.30 | Orbixx | Call rejected, CallToken Support required. If unexpected, resolve by placing address [IP HERE] in the calltokenoptional list or setting user [CLID HERE] requirecalltoken=no |
| 22:50.37 | Orbixx | Can somebody explain this. |
| 22:50.43 | Orbixx | It seems contradictory. |
| 22:50.46 | WIMPy | You just did. |
| 22:51.00 | WIMPy | How? |
| 22:51.14 | Orbixx | Call token support is required. |
| 22:51.23 | Orbixx | To resolve, I set requirecalltoken=no. |
| 22:51.32 | WIMPy | ... but wasn't provided but the peer. |
| 22:52.00 | Orbixx | ... What? |
| 22:52.00 | russellb | to resolve, make it not required |
| 22:52.03 | russellb | that's what it says |
| 22:52.12 | WIMPy | call token support. |
| 22:52.20 | Orbixx | That's the other odd thing. |
| 22:52.25 | Orbixx | The line is already there. |
| 22:52.41 | Orbixx | It's already set and it's definitely been loaded by Asterisk. |
| 22:54.49 | *** join/#asterisk erinspice (~erin@207.98.195.107) |
| 22:58.07 | *** join/#asterisk c0rnoTa (~c0rnoTa@80.251.113.55) |
| 22:58.19 | *** part/#asterisk c0rnoTa (~c0rnoTa@80.251.113.55) |
| 22:59.53 | tessier | What do you call it when you dial in from outside, press some special extension, enter a code, and then get a dialtone which you can use to dial out? |
| 23:00.03 | tessier | The name of that functionality escapes me at the moment... |
| 23:00.11 | tessier | Asterisk even has a dialplan function to implement it... |
| 23:00.16 | WIMPy | DISA |
| 23:00.53 | tessier | That's it, thanks! |
| 23:00.59 | tessier | direct inward system access |
| 23:02.18 | tessier | Trying to resolve a latency/delay issue. 5 second latency makes it a real pain to have a conversation. |
| 23:02.34 | tessier | But the ping time between our asterisk box and teliax is 25ms and zero packet loss. |
| 23:02.56 | tessier | We have 100mb or better connectivity end to end |
| 23:03.49 | ManxPower | Orbixx, I suspect THE FAR END does not support call tokens |
| 23:04.28 | ManxPower | tessier, set qualify=yes and look at the latency in sip show peers of iax2 show peers |
| 23:06.54 | tessier | ManxPower: Already done. 4ms. |
| 23:07.05 | tessier | teliax/edirect 8.3.252.23 N 5060 OK (4 ms) |
| 23:07.17 | tessier | That actually seems strangely low. |
| 23:07.26 | tessier | The ping time to them is 25ms. |
| 23:07.37 | tessier | oh, not it isn't. That's from the end of a dsl line. |
| 23:07.59 | tessier | This box is at a datacenter. Very close to them. |
| 23:08.20 | Chainsaw | Best-case ADSL latency is around 20ms (and that is with fast-path, aka interleaving disabled). |
| 23:08.29 | Chainsaw | So agreed, 4ms on DSL is beyond suspicious :) |
| 23:08.39 | Orbixx | ManxPower: It does, I have another Asterisk installation that works with it. |
| 23:08.47 | Orbixx | Oddly with a very similar configuration. |
| 23:09.04 | ManxPower | Orbixx, both the SAME version of Asterisk? |
| 23:09.20 | ManxPower | tessier, that qualify time (not ping time) is strangly low. |
| 23:09.32 | Orbixx | ManxPower: No, slight difference. |
| 23:09.40 | Orbixx | 1.6.2.2 and 1.6.2.12 |
| 23:09.53 | ManxPower | tessier, maybe there is a sip proxy involved? |
| 23:10.01 | ManxPower | Orbixx, should be close enough. |
| 23:10.08 | Chainsaw | IAX2 is so closely tied to the Asterisk core, it is recommended to have the same versions running at both ends of the link. |
| 23:10.10 | Orbixx | ManxPower: You would think so, eh :) |
| 23:10.25 | Orbixx | Chainsaw: One end is not within my control. |
| 23:10.32 | Orbixx | They provide trunking to PSTN. |
| 23:10.33 | Chainsaw | (Especially if you see strange problems, I would equalise that before spending any time on further troubleshooting) |
| 23:10.54 | Chainsaw | Orbixx: Would you not rather use SIP to talk to something you can't control? |
| 23:11.02 | Chainsaw | Orbixx: I would... |
| 23:11.04 | Orbixx | Chainsaw: My point is the IAX config block for one installation is exactly the same as the other installation. |
| 23:12.06 | Chainsaw | Orbixx: Then I would take the uncertainty out of it. Apply wireshark to both the working and the non-working install and see what is different on the wire. |
| 23:12.20 | ManxPower | I have up on IAX2 ten years ago and never looked back. |
| 23:12.31 | ManxPower | I GAVE UP on ..... |
| 23:12.48 | Chainsaw | Orbixx: If it is the other end that's screwed in interesting ways... you could spend a *lot* of time on this. Be certain of what is wrong. |
| 23:12.57 | Orbixx | Yeah. |
| 23:13.04 | Orbixx | That's why I'm seeking advice before I plough time into it. |
| 23:13.16 | Orbixx | Might be better to just make the switch to SIP perhaps. |
| 23:13.19 | Chainsaw | Orbixx: I would say wireshark is the next step if you are unwilling to equalise versions. |
| 23:13.26 | ManxPower | russellb gave you the advice already |
| 23:13.46 | Orbixx | scrolls up |
| 23:13.54 | Chainsaw | Orbixx: But yes, SIP sounds like a good bet. I always use that to talk to what I can't control. That stack is more robust in Asterisk. |
| 23:14.10 | Orbixx | ManxPower: As I said, it almost seems to ignore the config directive that makes it not required. |
| 23:16.51 | ManxPower | Orbixx, maybe it is. easy to test |
| 23:17.14 | ManxPower | iax2 has...issues.. where it will allow unauthenticated calls when you don't intend it to. dunno if that was fixed in 1.6 |
| 23:18.05 | ManxPower | in [general] set context=INVALID (or any context that does not exist). make sure you have all your peers set up with the correct context= line then as the LAST entry create a dummy entry with a context=INVALID |
| 23:18.34 | ManxPower | if you want to trap the unauthenticated calls create a [INVALID] context in extensions.conf |
| 23:19.10 | ManxPower | if your incoming call gets rejected then you were accepting uneuthenticated calls. |
| 23:19.49 | ManxPower | (the same was required for sip until the allowguest= option was added in 1.4(?) |
| 23:20.10 | Orbixx | Hmm. |
| 23:20.12 | Orbixx | Thanks. |
| 23:20.16 | Orbixx | I think I may just swing along with SIP. |
| 23:20.17 | Orbixx | :) |
| 23:21.28 | ManxPower | make sure you have allowguest=no in your sip.conf or you will accept unauthenticated calls |
| 23:22.21 | *** join/#asterisk jsidhu (~js@173-8-149-45-SFBA.hfc.comcastbusiness.net) |
| 23:22.38 | Orbixx | Thanks. |
| 23:23.14 | *** join/#asterisk ariel_ (~chatzilla@c-98-254-138-7.hsd1.fl.comcast.net) |
| 23:24.51 | jsidhu | hey guys, i just saw that digium has a fax for asterisk module. Once i get it installed, is there a front-end i can install and make use of for faxing? I tihnk what digium sells is just a module that enables asterisk to be able to send/recv faxes, what do u guys recommend/use for the front end (web based).. |
| 23:25.17 | Chainsaw | jsidhu: I have dialplan logic that just e-mails them into a shared folder. |
| 23:25.33 | jsidhu | what about sending faxes then? |
| 23:26.37 | Chainsaw | jsidhu: I have people operate a fax machine for that step. |
| 23:30.04 | *** join/#asterisk Supari (~Supari@c-68-38-232-185.hsd1.nj.comcast.net) |
| 23:30.47 | *** join/#asterisk niekie (~niek@CAcert/Assurer/niekie) |
| 23:31.04 | Supari | is there anyway to put a backup sip server for 1 trunk ? incase 1 fails itcan swap over to diff server ip |
| 23:34.42 | ManxPower | Supari, no, use the dialplan for failover |
| 23:35.07 | *** join/#asterisk micols (~mio@rlogin.dk) |
| 23:38.18 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 23:41.37 | *** join/#asterisk weta (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 23:45.31 | *** join/#asterisk jeremy_g (~jeremey_g@c-98a0e755.022-186-73746f3.cust.bredbandsbolaget.se) |
| 23:45.56 | jeremy_g | phew |
| 23:58.29 | *** join/#asterisk neurosys (~neurosys@c-65-34-190-58.hsd1.fl.comcast.net) |
