| 00:06.33 | *** join/#asterisk rajiv (~rajiv@gentoo/developer/rajiv) |
| 00:07.24 | *** join/#asterisk KingDavidNYC (~Chris1232@pool-71-191-150-2.washdc.fios.verizon.net) |
| 00:13.51 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 00:25.51 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 00:30.41 | *** join/#asterisk niekie (~niek@CAcert/Assurer/niekie) |
| 00:42.31 | *** join/#asterisk sjobeck (~sjobeck@137.118.149.195) |
| 00:42.52 | *** part/#asterisk sjobeck (~sjobeck@137.118.149.195) |
| 00:56.35 | *** part/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 00:56.53 | *** join/#asterisk ickmund (~ickmund@2001:470:c864:2:219:e3ff:fe02:b4c7) |
| 01:17.15 | roxdragon | wdoekes2, |
| 01:21.07 | *** join/#asterisk sjobeck (~sjobeck@137.118.149.195) |
| 01:22.20 | roxdragon | help :) |
| 01:22.24 | roxdragon | :( |
| 01:25.35 | *** join/#asterisk FinboySlick (~shark@74.117.40.9) |
| 01:26.59 | *** part/#asterisk sjobeck (~sjobeck@137.118.149.195) |
| 01:29.12 | *** join/#asterisk gray_ (~Gray@unaffiliated/remnant13) |
| 01:30.15 | FinboySlick | I'm trying to have an 1.6.2.17 asterisk talk with an old, decrepit 1.4.22 box through iax. It all works but I get massive stuttering only in the direction from the new box to the old box. I'm assuming it's some sort of jitter buffer issue but could probably use a few hints. (Yeah, I know the big hint is to upgrade the remote box, it's just not mine). |
| 01:32.03 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 01:41.09 | raden | elb, ? |
| 01:41.22 | raden | does asterisk actually no when a call is totally missed ? |
| 01:41.42 | elb | raden: if asterisk receives it, yes |
| 01:41.54 | elb | if it doesn't make it *to* asterisk, then of course no |
| 01:42.02 | raden | well im saying in asterisk |
| 01:42.05 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 01:42.11 | raden | where does it say if it was totally missed ? |
| 01:43.12 | elb | well, that depends on your dialplan |
| 01:43.14 | elb | you should know, right? |
| 01:43.23 | elb | you know what Dial() returns |
| 01:43.33 | elb | if you don't do something *else* with it after Dial ... it was missed |
| 01:48.24 | *** part/#asterisk shortcircuit (~shortcirc@rosettacode.org) |
| 01:49.18 | raden | hmm never paid close enough attention I guess |
| 01:49.26 | *** join/#asterisk tstorm (~tstorm@c-98-207-90-194.hsd1.ca.comcast.net) |
| 01:52.37 | *** join/#asterisk Iamnacho (~Iamnacho@ip174-70-138-161.ks.ks.cox.net) |
| 01:52.40 | FinboySlick | Hmmm, during a call, tcpdump shows the 1.6.2 box sending few, larger (336 and 500 bytes) packets to the 1.4.22 box, while the 1.4.22 box always sends 164 byte packets. Maybe this has something to do with it. |
| 01:53.45 | raden | FinboySlick, I would start there |
| 01:54.36 | FinboySlick | raden: Codec discrepancy? I think they're both ulaw. |
| 01:55.12 | raden | you using IAX ? |
| 01:55.26 | FinboySlick | raden: Yes. |
| 01:56.20 | *** join/#asterisk DrDigi (~mmurphy@108-69-211-16.lightspeed.frokca.sbcglobal.net) |
| 01:57.06 | raden | trying to find something one moment |
| 01:59.38 | raden | u might have a codex issue |
| 01:59.55 | *** join/#asterisk remnant13 (~Gray@unaffiliated/remnant13) |
| 02:00.15 | FinboySlick | Yeah, just forced gsm and it's all smooth and nice. |
| 02:00.25 | roxdragon | help SIP/2.0 401 Unauthorized |
| 02:15.52 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 02:19.58 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 02:21.34 | roxdragon | how to configure on asterisk: bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) Cos'è sto bind address? |
| 02:21.49 | roxdragon | how to configure? |
| 02:23.49 | *** join/#asterisk fnordus (~dnall@S01060023693bfad4.va.shawcable.net) |
| 02:36.17 | *** join/#asterisk coppice (~chatzilla@9.160.232.220.dyn.pacific.net.hk) |
| 02:36.57 | *** join/#asterisk DrDigi (~mmurphy@108-69-211-16.lightspeed.frokca.sbcglobal.net) |
| 02:45.41 | *** part/#asterisk root52 (~root52@ip70-191-116-76.cl.ri.cox.net) |
| 02:47.13 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 02:52.05 | *** join/#asterisk roxdragon (~roxdragon@unaffiliated/roxdragon) |
| 02:57.57 | roxdragon | Hi |
| 02:59.34 | Freeaqingme | Can anybody explain me this line? Google isn't very helpful: Difference is 3936, ms is 512 |
| 03:04.21 | roxdragon | used your nokia with asterisk? |
| 03:04.51 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 03:09.09 | Freeaqingme | roxdragon, nope |
| 03:14.19 | *** join/#asterisk Aut0Exec (~Joe@65.75.65.130) |
| 03:14.27 | Aut0Exec | anyone get asterisk going on their router? |
| 03:17.14 | *** join/#asterisk DrDigi (~mmurphy@108-69-211-16.lightspeed.frokca.sbcglobal.net) |
| 03:22.57 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 03:26.59 | roxdragon | hi |
| 03:27.08 | roxdragon | MOH don't work |
| 03:27.12 | roxdragon | why? |
| 03:30.42 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 03:45.13 | *** join/#asterisk DrDigi (~mmurphy@108-69-211-16.lightspeed.frokca.sbcglobal.net) |
| 03:49.15 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 03:52.19 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 03:56.48 | *** join/#asterisk DrDigi (~mmurphy@108-69-211-16.lightspeed.frokca.sbcglobal.net) |
| 03:57.07 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 04:00.21 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 04:02.12 | *** join/#asterisk DrDigi (~mmurphy@108-69-211-16.lightspeed.frokca.sbcglobal.net) |
| 04:05.39 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 04:09.25 | *** join/#asterisk asteriskmonkey (~darlek@74-51-38-204.telnetcommunications.com) |
| 04:09.51 | asteriskmonkey | im having an issue with realtime and i think its an asterisk draw back |
| 04:10.15 | asteriskmonkey | im creating contexts in my realtime extensions and there not reachable.. i know i tried this 5 years ago with no luck as it wasnt designed that ay |
| 04:10.24 | asteriskmonkey | is it still this way? |
| 04:13.13 | *** join/#asterisk w3rt (~w3rt@66-227-209-168.dhcp.trcy.mi.charter.com) |
| 04:13.18 | asteriskmonkey | is there no way have having contexts usable in realtime without having to first define them and use the switch=> Realtime in the flat file? |
| 04:18.42 | *** join/#asterisk WWGD (~WWGD@208.79.14.130) |
| 04:18.48 | *** part/#asterisk WWGD (~WWGD@208.79.14.130) |
| 04:22.08 | *** join/#asterisk moy (~moy@bas5-toronto47-1088894188.dsl.bell.ca) |
| 04:42.17 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 04:44.31 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 04:51.04 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 04:59.21 | *** join/#asterisk radic (~radic@dslb-178-002-208-162.pools.arcor-ip.net) |
| 05:04.56 | *** join/#asterisk psilikon (~psilikon@147-112.96-97.tampabay.res.rr.com) |
| 05:05.06 | *** join/#asterisk sourcode (~code@ppp-58-8-119-250.revip2.asianet.co.th) |
| 05:07.16 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 05:26.15 | *** join/#asterisk drcode (~user1@bzq-84-111-89-77.red.bezeqint.net) |
| 05:26.16 | drcode | hi all |
| 05:26.59 | drcode | I want to use db or to write into db for iax users |
| 06:15.26 | *** join/#asterisk Iamnacho (~Iamnacho@ip174-70-138-161.ks.ks.cox.net) |
| 06:20.21 | *** join/#asterisk tstorm_ (~tstorm@c-98-207-90-194.hsd1.ca.comcast.net) |
| 06:35.36 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 06:45.59 | *** join/#asterisk bmg505 (~leon@196-209-120-116.dynamic.isadsl.co.za) |
| 06:53.23 | *** join/#asterisk cashback (~mac@ip68-2-140-46.ph.ph.cox.net) |
| 07:26.39 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 07:47.00 | *** join/#asterisk kaushal (~kaushal@triband-mum-120.61.5.4.mtnl.net.in) |
| 07:47.05 | kaushal | Hi |
| 07:47.51 | kaushal | whats the Card specification for setting up E1 PRI 30 Channels |
| 07:47.59 | kaushal | for asterisk software |
| 07:50.09 | kaushal | Also is there a Automated dialer facility available in Asterisk ? |
| 08:07.18 | *** part/#asterisk kaushal (~kaushal@triband-mum-120.61.5.4.mtnl.net.in) |
| 08:13.29 | *** join/#asterisk kaushal (~kaushal@triband-mum-120.61.5.4.mtnl.net.in) |
| 08:13.44 | kaushal | checking in again for the query ? |
| 08:14.28 | wdoekes2 | there is a project that uses asterisk |
| 08:14.39 | wdoekes2 | I don't remember the name.. some kind of callcenter app |
| 08:15.06 | wdoekes2 | otherwise, the facility can be made.. asterisk provides all the building blocks |
| 08:15.35 | kaushal | wdoekes2: Are you referring to me ? |
| 08:15.43 | wdoekes2 | to your second question, yes |
| 08:16.07 | kaushal | ok |
| 08:16.26 | kaushal | wdoekes2: which card do you recommend for E1 PRI line |
| 08:17.56 | wdoekes2 | kaushal: not a clue, I use SIP over ethernet only |
| 08:18.06 | kaushal | and also which Linux Distribution OS would be suitable to set up Asterisk |
| 08:18.13 | kaushal | ok |
| 08:18.17 | wdoekes2 | debian or ubuntu |
| 08:18.24 | kaushal | ok |
| 08:18.31 | wdoekes2 | not because they're better suited for asterisk, just because they're better ;) |
| 08:18.42 | kaushal | so debian squeeze ? |
| 08:18.47 | kaushal | or ubuntu lucid ? |
| 08:18.59 | wdoekes2 | if you're unsure, go with ubuntu |
| 08:19.08 | kaushal | ok |
| 08:19.33 | kaushal | Also does asterisk support Voice XML ? |
| 08:19.43 | wdoekes2 | never heard of |
| 08:20.37 | kaushal | wdoekes2: ok |
| 08:21.16 | kaushal | Anyone else can guide me about Voice XML and the PRI card |
| 08:23.01 | kaushal | also what would be the recommended Hardware box for setting up Asterisk ? |
| 08:23.08 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 08:23.08 | wdoekes2 | http://lmgtfy.com/?q=asterisk+voicexml |
| 08:23.10 | kaushal | is it quad core or dual core .... ? |
| 08:23.27 | kaushal | and with Memory requirements |
| 08:23.35 | wdoekes2 | hardware depends entirely on the use |
| 08:23.50 | kaushal | is it better to go with the Intel Xeon or AMD ? |
| 08:24.01 | wdoekes2 | my my.. you have a lot of questions, don't you? |
| 08:24.12 | kaushal | yes i do have |
| 08:24.18 | kaushal | is that a problem ? |
| 08:24.35 | wdoekes2 | afaik, it really doesn't matter which processor brand you use |
| 08:24.52 | wdoekes2 | as for memory and speed/cores.. you'll need to find out by hand |
| 08:25.13 | kaushal | is it listed out in www.asterisk.org ? |
| 08:25.23 | wdoekes2 | until you're running several hundred calls, or are doing complicated scripting/recording, a low-end machine will do just fine |
| 08:25.44 | kaushal | ok |
| 08:25.51 | wdoekes2 | (thinking 2ghz/2gb ram) |
| 08:25.59 | kaushal | I would be dialling out 10000 calls |
| 08:26.08 | wdoekes2 | simultaneously? |
| 08:26.33 | kaushal | 240 calls simulatenaously |
| 08:26.47 | kaushal | I mean Outbound Calls |
| 08:27.20 | kaushal | and also what would be the Disk space requirement |
| 08:27.42 | kaushal | Does it have a HA mode available too ? |
| 08:27.49 | wdoekes2 | nothing significant.. unless you plan to do special things (like store recordings) |
| 08:28.15 | wdoekes2 | it's not designed to be HA.. but there are ways you can minimize downtime |
| 08:28.23 | wdoekes2 | like with any other app |
| 08:28.35 | kaushal | and also what would be the Disk space requirement ? |
| 08:28.39 | wdoekes2 | 09:27 < wdoekes2> nothing significant.. unless you plan to do special things (like store recordings) |
| 08:30.08 | kaushal | so ideally it would be a quad core / 8 Gigs RAM / 500 GB Hard Disk ? |
| 08:30.22 | kaushal | is that a overkill ? |
| 08:30.47 | kaushal | since i believe its processor intensive |
| 08:31.30 | wdoekes2 | unsure.. SIP/RTP is not that cpu intensive when you're not transcoding |
| 08:31.52 | wdoekes2 | but I know nothing about other channel drivers |
| 08:32.26 | wdoekes2 | the specs look fine.. |
| 08:34.25 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 08:44.01 | *** join/#asterisk Sipster (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 08:46.27 | *** join/#asterisk SiNGLer (~singler@78-60-139-121.static.zebra.lt) |
| 08:54.00 | *** join/#asterisk Olivier_54 (~Olivier_5@vaillant.famille-fontes.net) |
| 09:04.23 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
| 09:04.23 | *** mode/#asterisk [+o file] by ChanServ |
| 09:32.02 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 09:37.19 | *** join/#asterisk cashback (~mac@ip68-2-140-46.ph.ph.cox.net) |
| 09:50.37 | *** part/#asterisk adeel (~adeel@c-67-174-60-55.hsd1.ca.comcast.net) |
| 09:51.33 | *** join/#asterisk hehol (~Adium@ip-78-94-0-76.unitymediagroup.de) |
| 10:04.06 | *** join/#asterisk Sipster (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 10:41.03 | *** join/#asterisk niekie (~niek@CAcert/Assurer/niekie) |
| 10:41.06 | *** join/#asterisk luke-jr (~luke-jr@ishibashi.dashjr.org) |
| 10:48.28 | *** join/#asterisk luke-jr (~luke-jr@ishibashi.dashjr.org) |
| 10:49.49 | *** join/#asterisk niekie (~niek@CAcert/Assurer/niekie) |
| 10:58.55 | *** join/#asterisk cnu (cnu@the.ultimate.lamer.la) |
| 11:06.04 | *** join/#asterisk proute (~proute@ARouen-153-1-49-195.w90-17.abo.wanadoo.fr) |
| 11:06.12 | proute | Hello, |
| 11:06.43 | tzafrir_laptop | hi |
| 11:06.57 | proute | someone can tell me where I can find on asterisk.org or digium website, a technical docs about concurent call limit please? |
| 11:11.04 | tzafrir_laptop | proute, theoretically there's no limit. In practice there are quite a few bottle-necks |
| 11:11.42 | tzafrir_laptop | First and foremost - how many calls can your CPU(s) process? |
| 11:11.55 | tzafrir_laptop | There are a number of other bottlenecks |
| 11:12.24 | tzafrir_laptop | The actual numbers vary greatly and highly depend on the exact usage pattern |
| 11:13.12 | tzafrir_laptop | Generally - take a system, start bombarding it with calls (That represent your calls profile realistcly enough |
| 11:13.17 | proute | tzafrir_laptop: some people say me that asterisk can support about 250 simultaneous calls |
| 11:13.26 | tzafrir_laptop | And make sure calls running on the system still sound good enough |
| 11:13.55 | tzafrir_laptop | I'm not sure where such an exact number comes from |
| 11:14.13 | tzafrir_laptop | But it can easily top that. Given the right hardware |
| 11:14.35 | proute | tzafrir_laptop: for you, the call limit (incoming, outboun and internal) depend of cpu, ram and system? |
| 11:14.47 | tzafrir_laptop | (And natually, if you're only fter benchmarketting: make sure calls involve no transcoding and such ;-) ) |
| 11:15.08 | proute | tzafrir_laptop: I will use can reinvite |
| 11:15.24 | tzafrir_laptop | RAM is actually much less of an issue. Unless you record calls. In which case I/O matters |
| 11:15.44 | proute | tzafrir_laptop: I would like use asterisk like an opensips |
| 11:16.04 | tzafrir_laptop | If you don't send media through Asterisk, you avoid many of the nicer abilities. |
| 11:16.39 | tzafrir_laptop | Asetrisk is not like opensips. I hope you're well aware of the differences |
| 11:17.20 | proute | tzafrir_laptop: Yes, I know difference. for me when I say "like opensisp" it's an example |
| 11:17.49 | proute | Do you thinks that asterisk is able to work with about 2000 simultaneous calls? |
| 11:18.40 | proute | of course, with a good CPU and hardware |
| 11:28.24 | tzafrir_laptop | 2000 is quite a lot. In such a case, try to have more than one server |
| 11:28.41 | tzafrir_laptop | In any case, you'll need to spend some time optimizing things |
| 11:31.04 | proute | tzafrir_laptop: thanks. To resume asterisk is able to make more 250 simultaneous call? |
| 11:39.46 | tzafrir_laptop | proute, yes |
| 11:47.09 | *** join/#asterisk tris (~tristan@173-164-188-122-SFBA.hfc.comcastbusiness.net) |
| 11:53.27 | *** join/#asterisk ariel_ (~chatzilla@216.7.159.250) |
| 12:00.51 | *** join/#asterisk bmg505 (~leon@196-209-120-116.dynamic.isadsl.co.za) |
| 12:01.36 | *** join/#asterisk Sipster (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 12:02.17 | *** join/#asterisk resno (~bryan@unaffiliated/resno) |
| 12:02.28 | resno | what ports do i need to open up to allow access to asterisk? |
| 12:05.55 | ariel_ | ssh = 22, Sip = 5060 rtp =10,000 to 20,000 |
| 12:06.10 | resno | ariel_: ") |
| 12:06.18 | ariel_ | If your running a gui then 443 = https and 80 = http |
| 12:06.23 | ariel_ | resno: morning |
| 12:06.37 | resno | going out of town and hoping to use elastix to call to locals |
| 12:07.48 | ariel_ | nice |
| 12:09.30 | tzafrir_laptop | sip: UDP 5060. RTP: you can change that range of ports. Feel free to (and it's UDP) |
| 12:09.35 | *** join/#asterisk pressureman (~daniel@pik.hrz.tu-chemnitz.de) |
| 12:09.41 | tzafrir_laptop | ssh, HTTP, HTTPS: those are TCP |
| 12:09.56 | *** join/#asterisk Denial (Denial@drgi.co.uk) |
| 12:09.58 | pressureman | anyone here managed to configured ipv6 on yealink or tiptel phones? |
| 12:11.50 | resno | tzafrir_laptop: yea, ssh, and http(s) arent needed so much... just access |
| 12:12.18 | resno | wheres the specfication for port in elastix? |
| 12:14.56 | *** join/#asterisk mykhyggz (~col@evolone.org) |
| 12:44.23 | *** join/#asterisk Sipster (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 12:48.00 | *** join/#asterisk IridiumScaffold (~Will_Roue@124.169.157.159) |
| 12:48.28 | IridiumScaffold | anyone know anything about fax detection in asterisk 1.8? |
| 12:55.51 | *** join/#asterisk roxdragon (~roxdragon@unaffiliated/roxdragon) |
| 12:55.56 | roxdragon | /j #debian-it |
| 12:56.00 | roxdragon | XD |
| 13:02.24 | *** join/#asterisk imox1234 (~imox1234@p4FC5C07D.dip0.t-ipconnect.de) |
| 13:04.46 | roxdragon | hi all |
| 13:18.13 | *** join/#asterisk markit (~marco@88-149-177-66.staticnet.ngi.it) |
| 13:21.27 | *** join/#asterisk niekie (~niek@CAcert/Assurer/niekie) |
| 13:23.00 | *** join/#asterisk imox1234 (~imox1234@p4FC5C675.dip0.t-ipconnect.de) |
| 13:29.39 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 13:33.32 | *** join/#asterisk imox1234 (~imox1234@p4FC5C675.dip0.t-ipconnect.de) |
| 13:34.18 | *** join/#asterisk timahvo1 (~rogue@41.223.57.78) |
| 13:49.28 | *** join/#asterisk Mhaddog (~Mhaddog@z65-50-116-17.ips.direcpath.com) |
| 13:49.37 | *** join/#asterisk Denial (Denial@drgi.co.uk) |
| 14:02.40 | benngard | IridiumScaffold: no but i am looking for info to |
| 14:06.10 | benngard | IridiumScaffold: u mean like jumping to fax extension? |
| 14:08.01 | *** join/#asterisk ajkaanbal (~ajkaanbal@190.29.130.194) |
| 14:09.29 | *** join/#asterisk cyphorious (~cyphoriou@212-183-70-214.adsl.highway.telekom.at) |
| 14:10.53 | IridiumScaffold | yes. |
| 14:11.11 | IridiumScaffold | I have a voicetronix card |
| 14:11.19 | IridiumScaffold | chan_vbp, but no one seems to use it for that. |
| 14:11.37 | roxdragon | how do I install asterisk-addons-1.6.2.3? |
| 14:11.40 | IridiumScaffold | yet earlier our problem was it kept intercepting the faxes |
| 14:12.09 | IridiumScaffold | Now I can't seem to get it to throw the right events up. |
| 14:12.17 | roxdragon | only in the form to mp3. MySQL are not interested in installing |
| 14:12.37 | roxdragon | kaldemar, ping |
| 14:14.32 | *** join/#asterisk Sertys (~sertys@89.252.247.42) |
| 14:24.21 | benngard | IridiumScaffold: let me do some test, i will enable faxdetect on an extension and send a fax to it |
| 14:24.32 | *** join/#asterisk psilikon (~psilikon@147-112.96-97.tampabay.res.rr.com) |
| 14:24.48 | roxdragon | benngard, |
| 14:30.55 | benngard | IridiumScaffold: it didnt jump to fax extension :( |
| 14:31.13 | benngard | and i am 100% sure it was a t38 call |
| 14:31.18 | benngard | roxdragon: yes? |
| 14:31.33 | psilikon | I am trying to figure out why i am getting poor performance on a Debain 6.0 box running * 1.8.2.4. When the system is playing moh and I am ssh'd to the box and do something as simple as a 'find' or large dir 'ls' I get really choppy audio. Atom d510 (1.66Ghz dual core w/ HT) 2GB ram. This doesn't happen on my p3 800mhz system with 512MB ram. Any suggestions? |
| 14:31.47 | roxdragon | benngard, http://paste.ubuntu.com/582946/ |
| 14:32.00 | roxdragon | help plrase |
| 14:32.21 | psilikon | The p3 is running * 1.4.39 on Arch Linux |
| 14:33.36 | benngard | roxdragon: what kind of linux are u runnig? |
| 14:34.11 | roxdragon | yes debian |
| 14:34.51 | benngard | i am running debian 2.6.26-2-686 and Asterisk 1.8.4-rc2 |
| 14:36.27 | roxdragon | benngard, how to solve? |
| 14:36.29 | roxdragon | chan_ooh323.h:53:26: error: asterisk/rtp.h: No such file or directory |
| 14:36.42 | benngard | roxdragon: sec, i am thinking |
| 14:37.41 | benngard | roxdragon: u aint afraid of compiling asterisk from scratch? |
| 14:39.33 | roxdragon | asterisk compiled from 0 |
| 14:39.44 | roxdragon | i try installing addons |
| 14:40.00 | roxdragon | but rtp.h no such file |
| 14:40.01 | benngard | roxdragon: u dont need asterisk-addons, just enable ooh323 in make menuselect, its already there |
| 14:40.45 | roxdragon | benngard, i need activate Music on hold with support mp3 |
| 14:40.46 | benngard | roxdragon: give me a minute a will send u a very simple "build" file |
| 14:40.58 | roxdragon | ok |
| 14:42.18 | tzafrir_laptop | psilikon, what disk do you use? |
| 14:43.23 | tzafrir_laptop | Also: do you get the same choppy sound the second time you run that find command on the same directory? |
| 14:44.32 | psilikon | tzafrir_laptop: a 5400 rpm 160gb sata. |
| 14:44.51 | psilikon | my p3 system is using a old OLD 5400 40gb drive. |
| 14:45.34 | tzafrir_laptop | psilikon, also: Debian 6.0 with a Lenny kernel? Haven't booted yet after the upgrade? |
| 14:45.43 | *** join/#asterisk jeffik (~chatzilla@69-165-175-200.dsl.teksavvy.com) |
| 14:46.10 | psilikon | tzafrir_laptop: 6.0 kernel. Fresh install from 6.0. |
| 14:46.30 | benngard | roxdragon: http://pastebin.com/mTKj3nbg <- that will build with ooh323 support, u can ofc change to whatever tag/branch u want |
| 14:47.32 | psilikon | tzafrir_laptop: ppl in the debian channel are telling me to upgrade to the non-free realtek firmware for the eth adapter. |
| 14:47.37 | benngard | roxdragon: if a where u, i go for that version, alot of new oo323 stuff in rhere, will be even more when may has comitted this weakend work |
| 14:48.33 | roxdragon | benngard, i try compile asterisk addons not asterik 1.8 |
| 14:49.50 | benngard | roxdragon: but addons are in asterisk |
| 14:51.23 | *** join/#asterisk ajkaanbal (~ajkaanbal@190.29.130.194) |
| 14:51.52 | benngard | atleast ooh323 and mp3 support |
| 14:51.59 | tzafrir_laptop | psilikon, oops. Sorry, I confused you with someone else on the channel... |
| 14:52.11 | roxdragon | benngard, i have download asterisk Asterisk 1.8.3.2 from http://www.asterisk.org/downloads |
| 14:52.29 | roxdragon | and addons... Add-Ons 1.6.2.3 |
| 14:52.42 | roxdragon | from http://www.asterisk.org/downloads |
| 14:53.06 | benngard | u dont need addons for ooh323 and mp3 support |
| 14:53.13 | benngard | did u run make menuselect? |
| 14:55.08 | roxdragon | benngard, http://paste.ubuntu.com/582957/ |
| 14:55.14 | *** join/#asterisk datacompboy (~datacompb@l49-200-67.cn.ru) |
| 14:55.21 | roxdragon | choose? |
| 14:55.52 | datacompboy | Hi! :) Sorry for very-very dumb question, but what *prefix variables in chan_dahdi.conf stands for? |
| 14:56.34 | datacompboy | i have got incoming call with "TON: National Number" type of caller id, i have nationalprefix set to 0049, but CALLERID(num) is still without 0049 prefix. |
| 14:57.33 | WIMPy | IIRC the prefix stuff is only for outgoing calls. |
| 14:57.57 | WIMPy | For incomming you have to do it in your dialplan. |
| 14:59.32 | roxdragon | benngard, ???? |
| 14:59.57 | WIMPy | OTOH the whole TON stuff in dahdi confuses me every time :-( |
| 15:00.05 | benngard | roxdragon: i am running back and forth, if u gonna install from scratch, go for 1.8 |
| 15:00.23 | benngard | roxdragon: try the small script i sent u |
| 15:00.42 | benngard | it will take a while to build it but it will work |
| 15:01.30 | roxdragon | ok |
| 15:03.21 | *** join/#asterisk wesphillips (~wphillips@137-237-233-124.harris.com) |
| 15:04.07 | datacompboy | WIMPy: pffr... outgoing numbers i have always normalized... fine. where to get class of incoming number? i see no that in CALLERID() |
| 15:05.02 | datacompboy | WIMPy: aggh... callerid(ton). funny, 1.6+. |
| 15:06.19 | WIMPy | One of the reasons why I still prefer chan_lcr over chan_dahdi. |
| 15:06.43 | WIMPy | You have full screening and pattern replacement in both directions with LCR. |
| 15:07.26 | roxdragon | benngard, don't work |
| 15:07.34 | roxdragon | on make |
| 15:08.19 | datacompboy | WIMPy: i need only receive and send calls over zaptel's E1 card, nothing more. chan_lcr are too overbloated for that :) |
| 15:08.54 | WIMPy | It does eat resources, yes. |
| 15:08.57 | datacompboy | well, then, need to make additional per-server config for incoming number decode. i was think that chan_dahdi per-server should be ok for that. |
| 15:10.40 | WIMPy | And it depends on your hardware, off course. |
| 15:10.41 | datacompboy | WIMPy: Thanks for help! |
| 15:10.46 | *** part/#asterisk datacompboy (~datacompb@l49-200-67.cn.ru) |
| 15:10.53 | *** join/#asterisk jkprg_ (~jarda@ip-62-245-93-150.net.upcbroadband.cz) |
| 15:11.26 | *** join/#asterisk wonderworld (~ww@port-92-201-63-72.dynamic.qsc.de) |
| 15:16.52 | *** join/#asterisk datacompboy (~datacompb@l49-200-67.cn.ru) |
| 15:17.11 | benngard | roxdragon: what+ what kind of error du u get? |
| 15:17.19 | datacompboy | WIMPy: ${CALLERID(ton)} return unstead of 1 for international... |
| 15:17.31 | *** join/#asterisk nathan7 (nathan@unaffiliated/nathan7) |
| 15:17.47 | datacompboy | WIMPy: ${CALLERID(ton)} returns 17 instead of 1 for international... |
| 15:18.32 | roxdragon | benngard, i am executing your command, make menuselectd, > save > make |
| 15:18.38 | WIMPy | 17? Sounds like a bug. Maybe it includes the SI. |
| 15:18.45 | WIMPy | What version of Asterisk? |
| 15:18.48 | roxdragon | make: asterisk /rtp.h no such file o directory |
| 15:18.59 | datacompboy | WIMPy: Asterisk 1.6.2.9-2 |
| 15:19.41 | WIMPy | I'd go for a more recent version of libpri and Asterisk. |
| 15:20.31 | datacompboy | libpri is patched for a bit, but only in case of replace zero byte in number to letter "B" |
| 15:20.50 | datacompboy | to make sure receive number with letters over DSS |
| 15:20.58 | roxdragon | :( |
| 15:21.02 | datacompboy | not think that touch that option |
| 15:21.29 | WIMPy | Zero byte? |
| 15:22.31 | datacompboy | WIMPy: yes, when nuber "55555A3333" routed to asterisk, asterisk receive packet with "35 35 35 35 35 00 33 33 33 33". It just cuts off at zero byte, since it string-terminator. |
| 15:23.04 | datacompboy | I always patch libpri to replace zero byte with letter "A", and that makes me happy. |
| 15:23.38 | datacompboy | if (!strncasecmp("ton", data, 3)) { snprintf(buf, len, "%d", chan->cid.cid_ton); } |
| 15:23.39 | *** join/#asterisk anny__ (~chatzilla@80.79.159.205) |
| 15:23.44 | datacompboy | looks like there should be no bug... |
| 15:23.49 | anny__ | hey all |
| 15:25.18 | datacompboy | WIMPy: Fuuny :) found: pri->pvts[chanpos]->cid_ton = e->ring.callingplan; /* this is the callingplan (TON/NPI), e->ring.callingplan>>4 would be the TON */ |
| 15:25.46 | datacompboy | i need to divide that number by 16. |
| 15:26.40 | roxdragon | benngard, |
| 15:27.00 | datacompboy | WIMPy: can you fix wiki page at http://www.voip-info.org/wiki/view/Asterisk+func+callerid ? to note that it can also require shift |
| 15:30.12 | pressureman | when i run freeswitch on my PC Engines ALIX board (500 MHz CPU, 256MB RAM), my kernel compilation runs a bit slower than usual, but my audio sounds fine. any suggestions? |
| 15:30.23 | WIMPy | It should only contain te TON, not the whole byte. |
| 15:30.45 | WIMPy | And I have only seen valid vslues for TON so far. |
| 15:31.50 | *** join/#asterisk imox1234 (~imox1234@p4FC5C675.dip0.t-ipconnect.de) |
| 15:31.54 | datacompboy | WIMPy: http://google.com/codesearch?as_q=e-%3Ering.callingplan&btnG=Search+Code&hl=ru&as_package=asterisk&as_lang=&as_filename=chan_dahdi&as_class=&as_function=&as_license=&as_case= |
| 15:33.39 | roxdragon | benngard, http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/figs/web/ast2_0302.png |
| 15:33.40 | datacompboy | WIMPy: yes, looks like in 1.8 that get fixed |
| 15:34.45 | datacompboy | WIMPy: if, of course, p->cid_ton = caller->id.number.plan; == id.number.plan is only plan part :) |
| 15:35.52 | WIMPy | I don't think Asterisk has any concept of screening, so that makes sense. |
| 15:36.51 | WIMPy | Which is something that needs some work. The translation between screening and CALLERID types and back again. |
| 15:36.53 | datacompboy | Pffr... "*/% Return the results of multiplication, integer division, or remainder of integer-valued arguments." it do FLOAT division. $[17/16]=1.0625, not 1 :( |
| 15:37.54 | WIMPy | You already fond the correct spot in the source to correct it there. |
| 15:38.14 | *** part/#asterisk pressureman (~daniel@pik.hrz.tu-chemnitz.de) |
| 15:39.18 | WIMPy | If oyu fix it in the dialplan it will fail if you upgrade to a version where the bug was fixed. |
| 15:39.21 | datacompboy | WIMPy: i have to manage ~10 servers, where asterisk used as E1<=>SIP router. for me easier to make extensions.conf that can be shipped to any system-distributed asterisk version, that maintain patched/source built asterisk on every |
| 15:39.45 | datacompboy | WIMPy: if > 16; then /=16; else eat_as_is |
| 15:39.59 | WIMPy | That will wok, yes. |
| 15:40.44 | WIMPy | But I still wonder what kind of funny caller IDs you're patching around. |
| 15:41.16 | *** join/#asterisk jeffik (~chatzilla@69.165.175.200) |
| 15:42.22 | datacompboy | WIMPy: i can receive Local, National or International number, depending from place of caller. But system understand only 00XXXX normalized numbers. So i need to convert any kind of number to 00-prefixed international number |
| 15:42.50 | WIMPy | Sure. I mean that zero byte thing. |
| 15:43.39 | datacompboy | WIMPy: non-dialable by normal users numbers. paid 0900 lines routed to number with letter in it. noone can call that number directly |
| 15:44.27 | WIMPy | So libpri does not accept anything that's outside the range 0-9? |
| 15:44.43 | *** join/#asterisk sizzers (~k_killah@c-68-41-174-94.hsd1.mi.comcast.net) |
| 15:44.45 | *** part/#asterisk sizzers (~k_killah@c-68-41-174-94.hsd1.mi.comcast.net) |
| 15:45.06 | WIMPy | Or even terminates there? |
| 15:45.14 | datacompboy | WIMPy: Accept. But number that sent over PRI line have zero byte in it. If it send zero byte to asterisk as-is, asterisk cuts number at that point, and instead "5555\03333" asterisk see only "5555" |
| 15:46.07 | WIMPy | So you receive it as 0x00 already? Or is it sent to you as letter and then gets lost somewhere? |
| 15:46.25 | datacompboy | May be it can be useful to add option to handle that byte on asterisk side, but it took too much time to discuss at bugzilla, to describe why that really useful etc. i have negative experience in past with that :) so now have just easier to have patched version locally. |
| 15:46.53 | datacompboy | According to "pri set debug span 1" it receive it as 00 already |
| 15:47.36 | WIMPy | Hmm. That wouldn't be too friendly from your telco. |
| 15:48.06 | datacompboy | WIMPy: i have that on several telco in several countries, so i think that just realty, where i should live |
| 15:48.13 | WIMPy | something like 1234B5678 as you gave as an example would be more usual. |
| 15:48.45 | datacompboy | WIMPy: telco technick said that letters a-f not working using DSS signalling, and only can be used within SS7 |
| 15:49.06 | datacompboy | 1 < [70 10 a1 31 31 31 31 30 38 33 38 36 00 30 30 30 30 31] |
| 15:49.07 | datacompboy | 1 < Called Number (len=18) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '111108386B00001' ] |
| 15:50.24 | WIMPy | It's perfectly possible to even dial an URL using DSS1. |
| 15:50.44 | WIMPy | But it might be hard to find something that will understand it :-) |
| 15:51.16 | WIMPy | I have already seen numbers containing semicolon for exactly your purpose. |
| 15:51.35 | WIMPy | But a zero byte seems a little strange. |
| 15:51.46 | datacompboy | WIMPy: that can be problem only to their hardware :) yes, i understand that there no problem with sending 41 instead of 00 |
| 15:51.57 | datacompboy | But i get 00 from several telco's |
| 15:52.10 | datacompboy | Colon? E.163 doesn't allow colon... |
| 15:52.30 | WIMPy | semicolon |
| 15:52.48 | datacompboy | Afair, there was only 0-9 and A-F |
| 15:52.54 | datacompboy | sorry, A-D |
| 15:52.58 | datacompboy | E/F reserved |
| 15:53.20 | WIMPy | Well, as you explaind yourself, the idea is that it cannot be dialed by any user. |
| 15:53.44 | datacompboy | yes, and zero byte can't be dialed too :) |
| 15:53.59 | WIMPy | Yes, but that's usually hard to match as well. |
| 15:54.14 | datacompboy | but just possibility to have B,C,D just adds me 30000 more numbers to route. |
| 15:54.23 | WIMPy | So that doesn't seem to be an obvious choice. |
| 15:55.01 | datacompboy | Telco said they route number as-is. We have tried A/B/C/D -- and all of them reach telco as letter, and reach us as 00 |
| 15:55.56 | WIMPy | What kind of equipment do they expect on your end that would be able to match numbers that contain null bytes? |
| 15:56.18 | WIMPy | I think I'd take that as a bug on their side. |
| 15:56.42 | WIMPy | Or at the place hosting the numbers. |
| 15:57.08 | datacompboy | WIMPy: may be. you gave me good point, i'll try to eat brain of technics from all available telcos in case of what hardware they do use, and why letter get eaten |
| 15:57.32 | WIMPy | :-) |
| 15:57.54 | WIMPy | That doesn't look intentional to me. |
| 15:58.24 | datacompboy | WIMPy: yes, but if it can work better, imho, it should work better. :) |
| 15:59.27 | *** join/#asterisk Andrew__M (62712e05@gateway/web/freenode/ip.98.113.46.5) |
| 15:59.31 | WIMPy | Indeed. |
| 16:02.12 | Andrew__M | Q: What's the best way of making a list of 100 or so numbers that are allowed to call me? |
| 16:03.45 | Andrew__M | I would then send all others to voice mail. |
| 16:05.13 | roxdragon | benngard, where install redame-addons.txt? |
| 16:05.17 | elb | I dunno about best, but for a small list like that I'd use astdb |
| 16:06.06 | elb | if you don't mind entering the numbers at the console (perhaps via a script), that's trivial ... writing a dialplan to add/remove numbers is not super difficult, but would take more time |
| 16:06.29 | elb | do something like |
| 16:07.09 | Andrew__M | elb: I am listening... |
| 16:08.06 | *** part/#asterisk datacompboy (~datacompb@l49-200-67.cn.ru) |
| 16:08.13 | elb | exten => s,n,GotoIf($[${DB(whitelist/CALLERID(num))} = "yes"]?ring) |
| 16:08.23 | elb | exten => s,n,Voicemail(...) |
| 16:08.30 | elb | exten => s,n,Hangup() |
| 16:08.35 | elb | exten => s,n,Dial(...) |
| 16:08.40 | elb | erm, that last one should be |
| 16:08.51 | elb | exten => s,n(ring),Dial(...) |
| 16:08.55 | elb | or whatever, you get the picture |
| 16:09.33 | elb | then you can add entries to the whitelist at the console with 'database put whitelist 15155551212 yes' |
| 16:10.15 | Andrew__M | elb: Yes, thank you, I understand. But how do I keep track of whose number is in whitelist? To avoid duplicates. Currently the numbers are in Google Contacts. |
| 16:11.19 | elb | if you use a db put like that, you can't have duplicates |
| 16:11.27 | elb | putting the same number twice is idempotent |
| 16:11.36 | elb | I mean, don't get me wrong, that's a hack of a solution |
| 16:11.42 | elb | it's just super easy :-) |
| 16:12.08 | benngard | roxdragon: sip:/usr/src/asterisk-1.8.4# ls README-addons.txt |
| 16:12.08 | benngard | README-addons.txt |
| 16:12.10 | benngard | :) |
| 16:12.47 | Andrew__M | elb: I originally thought of having an external file with 2124354657 ;Mom etc. with several entries under each other. But could not figure out a way to make Asterisk read and use a file like that. |
| 16:13.07 | elb | sure, you could do that, and there are interfaces which would make that rasonable |
| 16:13.16 | elb | but the db is so easy to use that I use it for many such tasks |
| 16:13.43 | elb | I even use it for a dialin authentication so that users can dial as their own extension from remote, etc. |
| 16:13.53 | elb | password management in the dialplan and everything ;-) |
| 16:14.00 | Andrew__M | elb: could I store and retrieve associated names? |
| 16:14.11 | elb | absolutely |
| 16:14.18 | elb | you might want to look at CUT() |
| 16:15.19 | elb | I store entries like exten;jid@jabberserver;defaultcontext;etc |
| 16:15.43 | elb | then use DB to retrieve them based on whatever key, and CUT() to extract the field I want |
| 16:16.08 | Andrew__M | elb: OK, thanks. I need to read The Asterisk book section about the AstDB. |
| 16:16.24 | elb | yeah, it's worht reading |
| 16:16.47 | elb | I believe there are also modules to let you use external databases in a similar fashion |
| 16:16.53 | elb | which, for a large install, you might want to look into |
| 16:17.12 | elb | for me, my server has only myself and maybe six other people on it... so I do a lot with DB() |
| 16:17.55 | Andrew__M | elb: Yes, I love the book, but did not get to AstDB yet. This server is a home system, so a small DB should be more than enough. |
| 16:18.17 | elb | cool |
| 16:18.19 | elb | good luck :-) |
| 16:18.37 | Andrew__M | elb: Thanks again! :-) |
| 16:19.33 | elb | by the way, check my syntax up there on the DB/CUT/etc. |
| 16:19.41 | elb | that was off the top of my head |
| 16:19.49 | elb | there may be some bracket/brace/paren problems |
| 16:20.40 | Andrew__M | elb: I will check, but your head top is pretty good. I could never memorize so much code. |
| 16:21.35 | elb | I just happen to have worked on this quite recently |
| 16:21.38 | elb | otherwise, yeah |
| 16:21.43 | elb | dialplan syntax is a bit grotty ;-) |
| 16:22.36 | roxdragon | benngard, asterisk moh don't work with mp3.... how to solve? |
| 16:25.11 | *** join/#asterisk KingDavidNYC (~Chris1232@pool-71-191-150-2.washdc.fios.verizon.net) |
| 16:25.46 | KingDavidNYC | Hello, I have a question on how to setup asterisk with opensips, can anyone helpe me? |
| 16:27.21 | *** join/#asterisk eject_ck (~eject_ck@83-218-247-220.dynamic.vega-ua.net) |
| 16:27.24 | eject_ck | Hi all |
| 16:28.12 | Andrew__M | KingDavidNYC: Do you mean how to connect them together? |
| 16:29.01 | eject_ck | I want to send ~ 100 faxes. I have 10 licenses fax for *. I'm creating 100 .call files and putting all of them to /outgoing folder. It works for me with 1 .call file. How it will work if I will put all 100 files ? |
| 16:29.32 | KingDavidNYC | Andrew_M: I think I got some concepts now, but the way I am getting it, is that all registration and user information will be handled now by opensips, is that correct? |
| 16:29.46 | eject_ck | what's best practice ? |
| 16:30.10 | Andrew__M | KingDavidNYC: What are you trying to achieve? |
| 16:31.06 | eject_ck | how asterisk's queue deal with channel licenses ? |
| 16:31.15 | eject_ck | I can guess no interaction :)) |
| 16:31.23 | eject_ck | I'm wrong ? |
| 16:32.14 | KingDavidNYC | Andrew_M: In asterisk I was used to put the list use users and devices in sip.conf... it looks like I dont need to do that anymore since opensips now handles the database of the users correct?... I am trying to use opensips as a proxy and asterisk as a the pbx |
| 16:32.50 | KingDavidNYC | Andrew_M: but I want to do voip hosting with this combination, |
| 16:32.52 | Andrew__M | eject_ck: Be a little patient. Sometimes you get an answer minutes and even hours later from users that are away, but monitoring the chat. |
| 16:33.10 | eject_ck | Andrew__M: sure I will :0 |
| 16:33.12 | eject_ck | Andrew__M: sure I will :) |
| 16:35.22 | Andrew__M | KingDavidNYC: I know this can be done, because we have a vendor (in Manhattan) who proposed exactly this. I personally don't yet know how this is done. |
| 16:36.02 | *** part/#asterisk IridiumScaffold (~Will_Roue@124.169.157.159) |
| 16:36.14 | eject_ck | found part of answer in faq |
| 16:36.17 | eject_ck | In the event that all of your available fax channel licenses are in use, perhaps two calls are in progress and two licenses are available, and another call, the third, requests access to the fax modems, then the third call will fail. Asterisk does not queue up calls pending the release of a fax license from an active session because remote fax machines do not sit idle waiting to begin transmitting |
| 16:36.40 | eject_ck | a fax, they respond better if the call is hung up most fax machines include automated redial capabilities. Best practice then is to always maintain one more channel of Fax For Asterisk licenses then you think that you will need. |
| 16:37.02 | KingDavidNYC | Andrew_M: in an asterisk+opensips combination, where does the list of users go? in the asterisk db, or in the opensips db? |
| 16:41.41 | *** join/#asterisk Andrew__M (62712e05@gateway/web/freenode/ip.98.113.46.5) |
| 16:42.13 | Andrew__M | KingDavidNYC: I am back. |
| 16:43.06 | KingDavidNYC | Andrew_M: it is just that I find it hard to believe that opensips will handle the database of users..I need confirmation |
| 16:44.13 | Andrew__M | KingDavidNYC: I believe it acts as a heavy-duty switch, freeing Asterisk. |
| 16:45.04 | KingDavidNYC | Andrew_M: are you familiar with the db_mysql.so module of opensips? |
| 16:45.32 | Andrew__M | KingDavidNYC: not familiar, sorry. |
| 16:45.42 | KingDavidNYC | Andrew_M: ok thanks |
| 16:46.58 | Andrew__M | KingDavidNYC: If you need a vendor in Manhattan, I can refer you. They are experts in this. |
| 16:47.46 | KingDavidNYC | Andrew_M: that will be good, thank you |
| 16:50.59 | Andrew__M | KingDavidNYC: EUS Networks on 5th Ave and 30th st (or so) 212-624-5943. Jeronimo Romero is one of the owners and he is extremely knowleagable. |
| 16:58.06 | *** join/#asterisk SiNGLer (~singler@78-60-139-121.static.zebra.lt) |
| 16:58.30 | n1x0n | if I user register => in sip.conf to forward incomming call to some extension - does this have to be in [genera] ? I would prefer to have [general] with default context and then [inbound] with context=inbound but it doesn't quite work - any suggestions appreciated |
| 16:58.47 | *** join/#asterisk codefreeze-lap (~Steve_Mur@69.169.177.221.provo.static.broadweavenetworks.net) |
| 16:59.06 | elb | n1x0n: I'm not sure register does what you think it does |
| 16:59.15 | elb | n1x0n: the register statement does nothing but notify the SIP peer that you exist |
| 16:59.26 | elb | it does not control how inbound calls from that SIP peer are handled |
| 17:00.00 | elb | if that peer matches the [inbound] stanza according to the matching rules in the documentation, then incoming calls will be handled by [inbound] |
| 17:00.07 | n1x0n | but that's the only place I tell it to forward incoming calls to particular extensions - that's why I assumed it is for inbound |
| 17:00.21 | elb | define 'that' |
| 17:00.41 | n1x0n | register => |
| 17:01.00 | elb | the register doesn't tell it anything about forwarding |
| 17:01.03 | n1x0n | I do user:pas@example.com/extension |
| 17:02.33 | n1x0n | elb: ok so... I must be going mad as only register => section has the extension number that is called when I get incomming call. What in this case decides which extension to call ifthere is incoming call from particular provider ? |
| 17:02.50 | elb | oh |
| 17:02.58 | elb | you're telling the remote provider to call a specific extension? |
| 17:03.06 | elb | if you don't want them to do that, don't do that |
| 17:04.12 | elb | the information you put in the register statement is sent ot the other server, it doesn't affect your local dialplan |
| 17:04.27 | n1x0n | mmmmm ok I start seeing what I'm doing wrong |
| 17:04.31 | elb | if you send /context to the remote side, they'll try to dial in on the context you send -- but you can redirect or deny that if you want |
| 17:05.07 | KingDavidNYC | Andrew_M: thank you Andrew |
| 17:05.24 | elb | n1x0n: see p. 103ish of the Asterisk book |
| 17:05.31 | n1x0n | elb: ok thanks, makes more sense I think... i.e. I totally missunderstood register :-/ |
| 17:05.46 | n1x0n | elb: will do now , back to documentation then :) |
| 17:07.10 | n1x0n | btw I started breaking my config as caller id was not passed to phones - but I can see From: in tcpdump o_O but that later... thx again |
| 17:12.18 | *** join/#asterisk Sipster (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 17:28.27 | *** join/#asterisk Jasnejac (kvirc@81.91.107.236) |
| 17:33.50 | gentoo_fun2 | just got this error The PRI_CALL_HOLD installation appears to be missing or broken |
| 17:33.54 | gentoo_fun2 | when updating asterisk |
| 17:38.07 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 17:39.12 | *** join/#asterisk micols (~mio@rlogin.dk) |
| 17:43.00 | n1x0n | elb: so I have mu two providers with context="inout" but I can see in logs that it rejects "because extension not found in contest 'default' :-/ |
| 17:43.52 | elb | how are you matching them |
| 17:44.02 | elb | (secret, ip, name, etc. ?) |
| 17:44.46 | n1x0n | w8, I'll strip passwords and upload to pastie |
| 17:48.17 | *** join/#asterisk zxd (~zxd@95.211.21.37) |
| 17:48.18 | zxd | Hi |
| 17:48.23 | zxd | how do I turn off logging in asterisk |
| 17:49.38 | n1x0n | elb: extensions : http://pastie.org/private/4jqrovlmy4zarnbfs2zezw , sip : http://pastie.org/private/4bz1dikxvjhwd3iv3wrbea (stripped) |
| 17:50.36 | tzafrir_laptop | zxd, edit logger.conf |
| 17:50.36 | n1x0n | i.e. I don't get why incomming calls are in context 'default' (which I intentionally want to leave blank/empty for security reasons) |
| 17:50.43 | tzafrir_laptop | To apply changer: logger reload |
| 17:50.49 | tzafrir_laptop | (in the asterisk CLI) |
| 17:51.12 | tzafrir_laptop | n1x0n, set the context in the config file of the channel driver |
| 17:53.22 | n1x0n | tzafrir_laptop: ugh, ok it's not clear where it is but I'll do further googling about 'channel driver' , thanks! |
| 17:53.45 | tzafrir_laptop | n1x0n, what type of call? SIP? analog? |
| 17:53.50 | n1x0n | sip |
| 17:55.03 | elb | n1x0n: so, you have 'password' for all your password entries ... when, say, internetcalls.com SIPs into your box, do they register with a password? |
| 17:55.15 | *** join/#asterisk cyphorious (~cyphoriou@212-183-70-214.adsl.highway.telekom.at) |
| 17:55.20 | elb | and are you sure that the calls are coming from sip.internetcalls.com ? |
| 17:55.46 | tzafrir_laptop | n1x0n, 'context = ' in sip.conf |
| 17:55.52 | n1x0n | elb: I've changed these from real passwords, I assume they do |
| 17:56.02 | elb | tzafrir_laptop: he has that |
| 17:56.24 | n1x0n | elb: well I call the DID number from each operator |
| 17:56.27 | elb | n1x0n: if they're not registering with a username/password, you probably don't want that there ... many SIP providers do not |
| 17:56.29 | tzafrir_laptop | I guess that not in the right place |
| 17:56.31 | n1x0n | so they must be comming from them I guess |
| 17:56.53 | n1x0n | elb: I', pretty sure they do - I can see them in sip show peers |
| 17:56.57 | n1x0n | and sip show registry |
| 17:56.57 | elb | ok |
| 17:56.57 | tzafrir_laptop | n1x0n, set 'context = does-not-exist' in [general] |
| 17:57.08 | tzafrir_laptop | Apply it ('sip reload') |
| 17:57.12 | n1x0n | k w8 |
| 17:57.19 | tzafrir_laptop | Now: where do incoming calls go to? |
| 17:57.32 | elb | tzafrir_laptop: did you look at his config? |
| 17:57.36 | elb | (he pasted it above) |
| 17:58.33 | n1x0n | [Mar 20 17:58:13] NOTICE[31217]: chan_sip.c:20152 handle_request_invite: Call from '' to extension 's' rejected because extension not found in context 'does-not-exist'. |
| 17:58.49 | elb | yes, we know that the incoming stanzas are not being respected |
| 17:58.54 | n1x0n | so it matches the default on in [general] which I don't want as mentioned |
| 17:59.02 | elb | the question is why |
| 17:59.24 | tzafrir_laptop | n1x0n, so now we know that the 'default' came from 'context=' in [general] |
| 17:59.34 | tzafrir_laptop | So it was not overriden by anything else |
| 17:59.36 | n1x0n | yes |
| 18:00.18 | *** join/#asterisk [netman] (~netman@184.Red-83-41-7.dynamicIP.rima-tde.net) |
| 18:00.25 | tzafrir_laptop | Do you see 'nixon' as registered in 'sip show peers' (with the right IP address)? |
| 18:00.36 | elb | ok, too many cooks in the kitchen, I'm out |
| 18:00.38 | n1x0n | and IIRC if I remove register => from sip.conf the DID stopps working, that's why I initially thought I can do registry in [someection] and do context = blabla |
| 18:01.04 | n1x0n | tzafrir_laptop: yes, that's my phone, how is that related ? |
| 18:01.20 | tzafrir_laptop | did the call come from it? |
| 18:01.20 | n1x0n | elb: thanks for help :) |
| 18:01.26 | n1x0n | no |
| 18:01.35 | tzafrir_laptop | sorry, my mistake. |
| 18:01.39 | n1x0n | it shouldn't , I'm calling my DID from my mobile |
| 18:01.43 | elb | n1x0n: I'd turn on 'sip set debug on' if I were you, and look to see what the incoming SIP looks like |
| 18:01.52 | elb | and/or take a tcpdump of the incoming session |
| 18:01.53 | n1x0n | tries |
| 18:01.58 | tzafrir_laptop | hmm... is 'context' respected in a type=peer entry? |
| 18:02.00 | elb | see if you can figure out why it's not matching properly |
| 18:02.05 | n1x0n | yeah already tcpdumping but my sip knowlendge sucks |
| 18:02.37 | tzafrir_laptop | n1x0n, actually, if you want to go technical with a protocol you're not familiar with, wireshark is your friend |
| 18:02.45 | elb | oh, good point |
| 18:02.46 | tzafrir_laptop | But I suspect this is something simpler |
| 18:02.48 | elb | you want type=user |
| 18:02.49 | elb | not type=peer |
| 18:03.04 | tzafrir_laptop | anyway, I really need to go |
| 18:04.06 | n1x0n | tzafrir_laptop: thanks |
| 18:04.10 | n1x0n | elb: I'll try that |
| 18:04.20 | n1x0n | I have the sip extracted from sip debug on |
| 18:04.51 | n1x0n | and it says totally different thing that what I see in logs ! |
| 18:05.04 | n1x0n | [Mar 20 18:02:51] NOTICE[31217]: chan_sip.c:20152 handle_request_invite: Call from 'kroemeke' to extension 's' rejected because extension not found in context 'inout'. |
| 18:05.09 | n1x0n | which makes much more sense now |
| 18:05.12 | elb | you're probably ultiamately going to want friend |
| 18:05.17 | elb | because you want to make outgoing calls, as well |
| 18:05.22 | elb | or else both peer and friend entries |
| 18:05.25 | n1x0n | yeah I'll change that |
| 18:05.47 | elb | yeah, you need an s extension in inout |
| 18:05.48 | n1x0n | but I think I found the issue and I'll do some more digging, very helpful ! thx |
| 18:05.52 | n1x0n | yeah :) |
| 18:05.55 | elb | incoming calls start at s |
| 18:06.00 | elb | unless you tell the other end otherwise |
| 18:06.13 | n1x0n | but odd that it was showing different thing in log (with context default etc..) |
| 18:06.21 | elb | the two 100,1s are also going to flub you up |
| 18:06.22 | n1x0n | anyway - at least I have a starting point |
| 18:06.29 | n1x0n | oh |
| 18:06.30 | elb | you want s,1,Set() |
| 18:06.35 | elb | s,n,Dial() |
| 18:06.40 | elb | s,n,(Voicemail) |
| 18:06.41 | elb | erm |
| 18:06.47 | elb | s,n,Voicemail() |
| 18:07.27 | n1x0n | kk , will do, back to vim and reloading config :-D *fun* |
| 18:07.35 | elb | :-) |
| 18:10.22 | *** join/#asterisk mykhyggz (~col@evolone.org) |
| 18:15.13 | *** part/#asterisk wesphillips (~wphillips@137-237-233-124.harris.com) |
| 18:15.21 | *** join/#asterisk cyphorious (~cyphoriou@212-183-70-214.adsl.highway.telekom.at) |
| 18:17.39 | *** join/#asterisk mawhii (~mawhii@99.117.188.231) |
| 18:26.10 | *** join/#asterisk b14ck (~b14ck@cpe-72-129-70-245.socal.res.rr.com) |
| 18:31.58 | *** join/#asterisk jkroon (~jkroon@dsl-242-1-14.telkomadsl.co.za) |
| 18:36.52 | *** join/#asterisk [netman] (~netman@80.31.144.145) |
| 18:46.33 | *** join/#asterisk micols (~mio@rlogin.dk) |
| 18:53.37 | rajiv | ~itsplist-us |
| 18:53.37 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 18:57.27 | *** join/#asterisk Dovid (~Dovid@213.8.121.90) |
| 18:59.36 | *** join/#asterisk nighty^ (~nighty@tin51-1-82-226-147-104.fbx.proxad.net) |
| 19:01.08 | Maliuta | ~itsplist-au |
| 19:04.43 | benngard | ~itsplist-se |
| 19:04.47 | benngard | :) |
| 19:06.47 | *** join/#asterisk [netman] (~netman@75.Red-83-41-1.dynamicIP.rima-tde.net) |
| 19:13.50 | *** join/#asterisk jkroon (~jkroon@dsl-242-1-14.telkomadsl.co.za) |
| 19:14.49 | benngard | hmm, i cant figure out how the fax extension is working :( |
| 19:17.07 | benngard | if i send a call (fax) to an extension with faxdetect=yes, i thought it would detect t38 and jump to fax,1 |
| 19:18.22 | benngard | i send the same call to "ReceiveFAX" and i can se the t38 is working so the call is a t38 call |
| 19:19.43 | benngard | i do understand that i am missing some important stuff, but what? |
| 19:25.11 | *** join/#asterisk ajkaanbal (~ajkaanbal@190.146.207.178) |
| 19:29.34 | *** join/#asterisk Olivier_54 (~Olivier_5@vaillant.famille-fontes.net) |
| 19:36.55 | *** join/#asterisk Hanumaan (~Hanumaan@dslb-088-066-135-045.pools.arcor-ip.net) |
| 19:42.30 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 19:50.35 | *** join/#asterisk drcode (~chatzilla@bzq-84-111-89-77.red.bezeqint.net) |
| 19:50.37 | drcode | hi all |
| 19:50.49 | drcode | how can I use asterisk with mysql |
| 19:53.19 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
| 19:56.32 | roxdragon | how to parking call? |
| 20:00.09 | russellb | drcode: see the relational database integration chapter of this book |
| 20:00.13 | russellb | ~newbook |
| 20:00.13 | infobot | Asterisk: The Definitive Guide. A preview of the upcoming release is available at http://ofps.oreilly.com/titles/9780596517342/. The book is being released under a Creative Commons license: http://creativecommons.org/licenses/by-nc-nd/3.0/us/. Please consider supporting the authors generosity by purchasing a copy at http://oreilly.com/catalog/9780596517342. |
| 20:00.20 | russellb | roxdragon: the same book has a parking chapter. |
| 20:00.25 | russellb | "parking and paging" |
| 20:00.43 | roxdragon | russellb, i am italian.. not speak english... book is english |
| 20:01.07 | russellb | this channel is also english, though ... |
| 20:01.47 | drcode | thanx |
| 20:05.38 | roxdragon | asterisk-it is empty |
| 20:05.38 | *** join/#asterisk dailylinux (~test@188.148.70.118) |
| 20:26.07 | *** join/#asterisk weta (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 20:39.48 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-145.cablep.bezeqint.net) |
| 20:41.53 | *** join/#asterisk cyphorious (~cyphoriou@212-183-72-105.adsl.highway.telekom.at) |
| 20:42.51 | *** join/#asterisk b14ck (~b14ck@cpe-72-129-70-245.socal.res.rr.com) |
| 20:45.44 | *** join/#asterisk Olivier_54 (~Olivier_5@vaillant.famille-fontes.net) |
| 20:47.14 | *** join/#asterisk ajkaanbal (~ajkaanbal@190.146.207.178) |
| 20:48.54 | *** join/#asterisk cashback (~mac@ip68-2-140-46.ph.ph.cox.net) |
| 21:10.59 | *** join/#asterisk tomaw (tom@freenode/staff/tomaw) |
| 21:12.44 | *** join/#asterisk corretico (~luis@201.201.44.82) |
| 21:15.05 | *** join/#asterisk b14ck (~b14ck@cpe-72-129-70-245.socal.res.rr.com) |
| 21:24.56 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
| 21:26.11 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 21:26.57 | *** part/#asterisk Olivier_54 (~Olivier_5@vaillant.famille-fontes.net) |
| 21:27.33 | *** join/#asterisk codefreeze-lap (~Steve_Mur@69.169.177.221.provo.static.broadweavenetworks.net) |
| 21:41.46 | *** join/#asterisk juliocesarlhg (~jcesarg@190.234.250.59) |
| 21:41.58 | juliocesarlhg | hi |
| 21:42.00 | juliocesarlhg | from peru |
| 21:42.07 | *** join/#asterisk b14ck (~b14ck@cpe-72-129-70-245.socal.res.rr.com) |
| 21:42.15 | juliocesarlhg | could someone tell me if asterisk support mgcp |
| 21:42.37 | juliocesarlhg | i have a gateway gaoke with 4 fxs ports, i would like to connect it to my asterisk |
| 21:43.01 | juliocesarlhg | my gateway has mgcp protocol |
| 21:45.29 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 21:51.31 | tzafrir_laptop | juliocesarlhg, generally, asterisk does. |
| 21:51.31 | tzafrir_laptop | I don't know much more than that. |
| 21:51.34 | tzafrir_laptop | Give it a shot |
| 21:52.09 | juliocesarlhg | i dont know either |
| 21:52.13 | juliocesarlhg | who can help me? |
| 22:01.09 | juliocesarlhg | ??? |
| 22:01.29 | juliocesarlhg | who has experience with mgcp |
| 22:01.31 | juliocesarlhg | please help |
| 22:04.44 | tzafrir_laptop | ~ask |
| 22:04.44 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 22:05.03 | tzafrir_laptop | juliocesarlhg, have you actually tried it? |
| 22:05.18 | tzafrir_laptop | Generally, take a look at the sample mgcp.conf |
| 22:05.23 | *** join/#asterisk corretico (~luis@201.201.44.82) |
| 22:05.44 | *** join/#asterisk ajkaanbal (~ajkaanbal@190.146.207.178) |
| 22:05.48 | juliocesarlhg | i am trying to connect a mgcp gateway |
| 22:06.06 | tzafrir_laptop | and? |
| 22:07.24 | juliocesarlhg | i've look in into mgcp.conf |
| 22:07.31 | juliocesarlhg | i but can get it work |
| 22:07.52 | juliocesarlhg | my mgcp gateway has fxs ports |
| 22:08.11 | juliocesarlhg | there's no tone |
| 22:09.03 | juliocesarlhg | in the phone |
| 22:19.55 | *** join/#asterisk cnu (cnu@the.ultimate.lamer.la) |
| 22:21.22 | *** join/#asterisk codefreeze-lap (~Steve_Mur@69.169.177.221.provo.static.broadweavenetworks.net) |
| 22:40.42 | Freeaqingme | I have one AGI script where I want /all/ my calls to route to. Any idea how I should do that? |
| 22:43.27 | *** join/#asterisk cashback (~mac@ip68-2-140-46.ph.ph.cox.net) |
| 22:45.28 | *** join/#asterisk CoffeeIV (~rgr@rrcs-71-42-183-81.sw.biz.rr.com) |
| 22:52.57 | Freeaqingme | nvm, I'm an idiot. Got it. Tnx |
| 22:53.40 | *** join/#asterisk wonderworld (~ww@port-92-201-45-217.dynamic.qsc.de) |
| 22:53.44 | *** join/#asterisk a_m_y (~chatzilla@121.96.33.130) |
| 22:54.26 | a_m_y | ~ebook |
| 22:54.33 | a_m_y | ~book |
| 22:54.33 | 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, or http://ofps.oreilly.com/ |
| 23:04.56 | *** join/#asterisk cyphorious (~cyphoriou@212-183-73-241.adsl.highway.telekom.at) |
| 23:05.23 | CoffeeIV | Hi, I have asterisk 1.8 installed on CentOS. I am trying to get it to register with a televantage server, and I took the default configs (from when you yum install asterisk18-configs) and added some stuff to sip.conf, but asterisk does not seem to be attempting the registration. I have pastebined what I added to sip.conf here: http://pastebin.com/NXn4GKMC |
| 23:07.39 | *** join/#asterisk Olivier_54 (~Olivier_5@vaillant.famille-fontes.net) |
| 23:08.08 | *** part/#asterisk Olivier_54 (~Olivier_5@vaillant.famille-fontes.net) |
| 23:18.37 | *** join/#asterisk marienz (~marienz@freenode/staff/marienz) |
| 23:22.49 | *** join/#asterisk fnordus (~dnall@S01060023693bfad4.va.shawcable.net) |
| 23:28.07 | a_m_y | ~sipnat |
| 23:28.07 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, canreinvite, externhost or externip, and localnet. |
| 23:37.32 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
| 23:42.10 | *** join/#asterisk vinhdizzo (~vinh@pool-173-51-129-161.lsanca.fios.verizon.net) |
| 23:44.15 | *** join/#asterisk psilikon (~psilikon@147-112.96-97.tampabay.res.rr.com) |
