| 00:00.00 | Jumpie | so its technically not near the fxo |
| 00:00.02 | Jumpie | er fxs mod |
| 00:00.09 | Jumpie | damn...i cant believe i mi ssed that |
| 00:00.34 | Jumpie | i knew that it saw it..but wasnt working somehow |
| 00:01.02 | jaytee | I had a friend build his own i7 system and couldn't get it to fully boot, it would power up and then shut off immediately. it helps to have the 12v supply to the CPU plugged in :-) |
| 00:02.17 | ChannelZ | oops |
| 00:02.54 | chazzam | huh, I thought someone said asterisk 1.6.2.0 was included in the asterisknow repo, but I just see it jumping from 1.6.0.26 to 1.6.2.6 |
| 00:03.09 | ChannelZ | My server died last week, an old Athlon MP board. It has a rare 'ATX GES' power supply.. the ports on the board look like normal ATX 24+8, but it's not! |
| 00:04.30 | *** join/#asterisk [OpenSys] (~vasco@fw.vslinux.net) |
| 00:04.51 | *** join/#asterisk Yudaisrael1984 (~Yuda@80.179.161.1.static.012.net.il) |
| 00:05.10 | Yudaisrael1984 | hey guys im having a major issue with my asterisk all of a sudden i got a segmentation fault |
| 00:05.33 | Yudaisrael1984 | and i keep getting a asterisk ended with exit status 139 |
| 00:05.40 | Yudaisrael1984 | what is going on with my asterisk? |
| 00:07.49 | Jumpie | yay!! they are all operational |
| 00:07.54 | Jumpie | chanks jaytee, ChannelZ |
| 00:08.02 | jaytee | yw |
| 00:08.33 | Yudaisrael1984 | anyonoe know why im getting that error |
| 00:08.39 | Yudaisrael1984 | its important my server is crashing |
| 00:09.15 | jaytee | Yudaisrael1984, is this a new install or has this been in production for awhile and just started happening? |
| 00:09.28 | Yudaisrael1984 | no its been in production for a while |
| 00:09.35 | Yudaisrael1984 | and it just starting happening |
| 00:09.43 | jaytee | and changes or updates applied to the server recently? |
| 00:09.44 | chazzam | did you update anything or change any configs? |
| 00:09.52 | Yudaisrael1984 | nope |
| 00:09.58 | Yudaisrael1984 | no changes were made to the server |
| 00:10.44 | jaytee | have you tried a coldboot? |
| 00:10.46 | *** join/#asterisk RypPn (~TuMbL@rosscom.co.uk) |
| 00:10.57 | Yudaisrael1984 | yes |
| 00:11.02 | Yudaisrael1984 | i rebooted the entire system |
| 00:11.06 | Yudaisrael1984 | twice |
| 00:11.51 | chazzam | isn't a cold boot shutting it off, then turning it back on? |
| 00:12.07 | jaytee | yes, that's a coldboot |
| 00:12.56 | chazzam | ok, thats what I thought |
| 00:13.04 | Yudaisrael1984 | ok so i will paste in pastebin part of the info i got in the cli |
| 00:13.26 | jaytee | if it still happens after a coldboot and there have been absolutely no changes to configs or any updates applied then the other potential causes are software corruption or failing memory. |
| 00:14.08 | *** join/#asterisk dinesh___ (~dinesh@84-73-120-175.dclient.hispeed.ch) |
| 00:14.43 | Yudaisrael1984 | how can i check which one it is? |
| 00:15.02 | dinesh___ | hi folks, I have an incoming number, but when I dial it with my mobile phone, nothing goes from my SIP -> mobile, however mobile -> SIP is all fine |
| 00:15.24 | dinesh___ | in the asterisk console, with verbose set to level 4, i don't get any warning/error message |
| 00:15.29 | Jumpie | hey Fairman |
| 00:15.38 | dinesh___ | any idea what could cause sip to be working only one way ? |
| 00:15.40 | Jumpie | how do you know which port is the fxs port on the back of the card? |
| 00:15.43 | Jumpie | there seems to be no indicator |
| 00:15.49 | Jumpie | is it all done logically? |
| 00:15.54 | *** join/#asterisk coppice (~chatzilla@191.193.17.210.dyn.pacific.net.hk) |
| 00:17.06 | jaytee | port 1 is furthest from the motherboard |
| 00:17.44 | jaytee | I believe the channel numbers are stamped on the plate |
| 00:18.01 | Jumpie | but do i get to decide which are what? |
| 00:18.07 | Jumpie | like, when i setup an fxs extension |
| 00:18.08 | jaytee | no |
| 00:18.11 | Jumpie | i wanna be sure its plugged into the right port |
| 00:18.27 | jaytee | according to your scan the fxs is port 1 |
| 00:18.29 | Jumpie | ok so port 1 is at the top.... |
| 00:18.31 | Jumpie | yeah |
| 00:18.34 | Jumpie | nice |
| 00:18.46 | Jumpie | hmm.so are the channels still like g0, g1, etc? |
| 00:18.50 | Jumpie | in regards to incoming |
| 00:18.57 | jaytee | you could physically move the modules around if you wanted to |
| 00:19.05 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 00:19.37 | jaytee | groups are logical, channels are physical |
| 00:20.08 | jaytee | you could assign g4 to channel => 1 in chan_dahdi.conf if you wanted to |
| 00:21.19 | Jumpie | so this has nothign to do with /etc/dahdi/system.conf |
| 00:21.22 | Jumpie | its in the chan dahdi? |
| 00:23.00 | Jumpie | ok..when i check dahdi-channels.conf i see all the fxo ports are 'group 0' |
| 00:23.13 | Jumpie | but with channels 2, 3, 4 respectively |
| 00:23.21 | Jumpie | the fxs is group 5, with channel 1 |
| 00:23.30 | Jumpie | so i want to be sure i reference it correctly |
| 00:25.23 | *** join/#asterisk kn1ghtus (~chatzilla@c-71-197-68-128.hsd1.ca.comcast.net) |
| 00:25.57 | Yudaisrael1984 | can someone please please explain this to me http://pastebin.com/D8g3yBDr |
| 00:26.16 | kn1ghtus | Hello ya'lls |
| 00:29.08 | kn1ghtus | makes crickett noises. |
| 00:29.19 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 00:29.39 | kn1ghtus | hello? |
| 00:29.43 | coppice | ah, the sound of ball on willow bat |
| 00:29.46 | jaytee | Jumpie, this is my system.conf with 1 fxs in port 1 and 1 fxo in port 4 http://pastebin.com/zdDEisXK |
| 00:29.47 | kn1ghtus | *echo hello? helllo hello hello |
| 00:30.45 | *** join/#asterisk grandpapadot (~nonya@99-175-248-81.lightspeed.brhmal.sbcglobal.net) |
| 00:30.45 | jaytee | Jumpie, although it may seem backwards an fxs port uses fxoks signalling and an fxo uses fxsks signalling |
| 00:31.22 | Jumpie | yea..i get it..based on remote end |
| 00:31.25 | Jumpie | well my problem is trixbox |
| 00:31.28 | Jumpie | (ya laugh) |
| 00:31.37 | Jumpie | refers to it as "zap identifier(trunk name)" |
| 00:31.47 | kn1ghtus | has anyone had the problem of LOCAL (so over sip) DTMF tones not sending? example you can't hit ## to transfer? This is happening at random times on random handsets... No rrymthe or reason that we can find yet. |
| 00:32.30 | Jumpie | kn1ghtus always on the same phone? |
| 00:32.53 | kn1ghtus | Jumpie, thanks for you question but no on random handsets. |
| 00:32.58 | kn1ghtus | or softphones |
| 00:33.18 | kn1ghtus | it's really quite bazzar |
| 00:33.26 | kn1ghtus | can't for the life of us pin it down! |
| 00:33.38 | Jumpie | ok..and dialtone is... |
| 00:33.40 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 00:33.45 | Jumpie | is this bri/pri/fxo, sip only? |
| 00:34.03 | Jumpie | i know ther are some tweaks to dtmf, as well as some rx/tx gain options |
| 00:34.54 | kn1ghtus | i honestly don't know |
| 00:35.36 | kn1ghtus | it's SIP only |
| 00:36.20 | kn1ghtus | but because it is happening locally it wouldn't be dtmf tones |
| 00:36.29 | kn1ghtus | because it is sent over sip 100% |
| 00:38.28 | kn1ghtus | dose that sound correct in my assumption? |
| 00:39.29 | boodu | I search a kernel with module to openvz and zaptel (ztdummy) , on lenny. I don't found :( |
| 00:41.34 | *** join/#asterisk SSJGotenks (~SSJGotenk@h65-255-231-227.gtcust.grouptelecom.net) |
| 00:41.40 | *** join/#asterisk riddlebox (~riddlebox@75-132-225-75.dhcp.stls.mo.charter.com) |
| 00:49.40 | Jumpie | so....if you have 7 extensions to ring in a ringgroup... |
| 00:49.46 | Jumpie | do you have to setup voicemail for the ringgroup extension? |
| 00:49.57 | Jumpie | or...can i have it leave a copy of left voicemail on every extension in the group? |
| 00:50.01 | *** join/#asterisk ReDNeQ (~ReDNeQ@70.114.229.58) |
| 00:50.11 | Jumpie | i'd want perspective customer to see he has a voicemail from any extension he's lookin at...its a house setup |
| 00:57.51 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 01:05.25 | manxpower | Jumpie: Yes. |
| 01:05.52 | manxpower | Almost every question you ask scream "I'm a FreePBX/Trixbox user!" |
| 01:05.56 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 01:06.14 | *** join/#asterisk tiav (~tiav@ram94-3-82-225-11-215.fbx.proxad.net) |
| 01:06.45 | *** join/#asterisk Kumbang (~kumbang@rusnas.paume.itb.ac.id) |
| 01:06.55 | *** join/#asterisk Jed__ (~Jed@ool-4574c525.dyn.optonline.net) |
| 01:08.21 | manxpower | I really need to switch to X-Chat so I have a /ignore feature |
| 01:14.36 | Jumpie | because i sometimes use trixbox it warrants and ignore? |
| 01:14.54 | TJNII | ~trixbox |
| 01:14.55 | infobot | trixbox is, like, SH1TB0X. Basically a CRAPPY, closed-source distro. STAY AWAY! |
| 01:15.01 | TJNII | #trixbox |
| 01:15.02 | Jumpie | ffs..shut up |
| 01:15.14 | TJNII | I forget that ~trixbox isn't very helpfull. |
| 01:15.23 | TJNII | ~freepbx |
| 01:15.24 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 01:15.26 | Jumpie | out of the box it may have issues, but you can make it an intuitive version and work just fine, if you know what your doing |
| 01:16.46 | TJNII | Yea, but this is #asterisk. Not #asterisk_and_every_goddamned_thing_slapped_on_asterisk. You should seek support in the proper channel. |
| 01:17.02 | Jumpie | you should shrink your head |
| 01:17.15 | Jumpie | my questions were related to general questions, NOT how they pertain to trixbox |
| 01:17.44 | Jumpie | if you hopped off your anti trixbox soapbox and werent always ready to attack people in a prickish attitude that merely mentions trixbox, maybe you'd not be such a bitter person |
| 01:18.13 | TJNII | I'm not bitter. You can't apply anything we tell you. You should seek support in the proper channel. |
| 01:18.25 | Jumpie | ok iwhatever man |
| 01:18.27 | TJNII | But I digress... |
| 01:18.37 | Jumpie | so far the help i've gotetn HAS applied |
| 01:18.53 | *** part/#asterisk Jumpie (~lah@c-76-111-80-110.hsd1.va.comcast.net) |
| 01:18.58 | TJNII | RAGE QUIT! |
| 01:21.56 | TJNII | The ~trixbox bot command should really be more like ~freepbx. Might cut down on the butthurt. |
| 01:32.38 | *** join/#asterisk sawgood (~sawgood@173-13-158-25-sfba.hfc.comcastbusiness.net) |
| 01:33.54 | *** join/#asterisk k-man (~jason@unaffiliated/k-man) |
| 01:33.58 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 01:33.58 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 01:34.31 | k-man | for some reason i am suddenly getting registration timeout for sip: [Apr 28 11:32:14] NOTICE[25308]: chan_sip.c:11502 sip_reg_timeout: -- Registration for '0xxxxxxxx@sip.myfone.com.au@sip.mynetfone.com.au' timed out, trying again (Attempt #48) |
| 01:34.38 | k-man | how can i debug why? |
| 01:35.21 | sawgood | k-man: is the client in a NAT? |
| 01:36.15 | k-man | sawgood: yes - but its been working for months |
| 01:36.23 | k-man | it just stopped working today |
| 01:36.30 | sawgood | so, the client (sip end point) right (a phone) |
| 01:36.35 | p3nguin | But now it's not working, so don't take the offense just yet. |
| 01:37.02 | k-man | sawgood: oh, sorry, misread what you said, no client and asterisk are on the same network |
| 01:37.33 | sawgood | So, you have a phone on the same subnet as your Asterisk box, and your phone stops registering? |
| 01:38.33 | k-man | sawgood: no, phone restirs correctly, asterisk fails to register with my vsp |
| 01:38.44 | k-man | s/restirs/registers |
| 01:38.55 | p3nguin | No phone that I know of would want to send "'0xxxxxxxx@sip.myfone.com.au@sip.mynetfone.com.au'" in registration. |
| 01:40.06 | sawgood | oh ... so the Asterisk box stops registering to your ITSP after a short while |
| 01:40.10 | k-man | p3nguin: maybe i'm missing something here, but the phone is connecting to asterisk ok, its asterisk that's not connecting to my vsp afaict |
| 01:40.17 | sawgood | So, is the Asterisk box also the router, or is that another device? |
| 01:40.43 | k-man | asterisk box is not the router. asterisk box is behind NAT, on same subnet as client |
| 01:42.09 | *** join/#asterisk riddlebox (~riddlebox@75-132-225-75.dhcp.stls.mo.charter.com) |
| 01:42.23 | k-man | fwiw, i can ping the sip.myfone... address from the asterisk box |
| 01:43.41 | sawgood | k-man: excellent answer ... so what type of device is your router/firewall/nat? |
| 01:44.41 | k-man | sawgood: linksys wrt54gl running openwrt |
| 01:45.10 | sawgood | ok .. how long has this device been working for you? |
| 01:45.20 | k-man | 2 months or so |
| 01:45.34 | k-man | asterisk was also working fine last time i made a call a few days ago |
| 01:45.47 | k-man | and I have not changed any configuration in the last few weeks |
| 01:46.54 | sawgood | it is possible the 'router' is not keeping the NAT dynamic port open long enough for a reply to return, so it closes the dyanmic port ... only later here comes the NAT return packet ... and it cannot be processed by your router because the port was closed |
| 01:47.11 | k-man | ok, i can try rebooting the router |
| 01:47.37 | *** join/#asterisk y2kdemon (~ASK@h224.68.219.67.cable.vncn.cablerocket.net) |
| 01:48.34 | k-man | i noticed that mynetfone suggests to use sip01.mynetfone.com.au in their current asterisk documentation |
| 01:48.44 | k-man | i'm using the host sip00 |
| 01:51.29 | k-man | how do i see the sip satatus? |
| 01:51.40 | k-man | hmm... changing it to sip01 still produces the time out |
| 01:53.19 | k-man | ok, rebooting router now |
| 02:05.53 | ChannelZ | wonders what the new Skype BCP will cost |
| 02:06.06 | *** join/#asterisk btsteve (~tstevens@24-196-234-39.dhcp.gwnt.ga.charter.com) |
| 02:10.30 | *** join/#asterisk Kumbang (~kumbang@rusnas.paume.itb.ac.id) |
| 02:12.46 | Slugs_ | ~waiting |
| 02:12.47 | infobot | extra, extra, read all about it, waiting is what mwilson does while fucking his mum |
| 02:13.35 | *** join/#asterisk spenguin[work] (~penguin@59.162.86.164) |
| 02:14.12 | spenguin[work] | test |
| 02:14.20 | ChannelZ | FAIL |
| 02:14.28 | spenguin[work] | ACK |
| 02:14.34 | *** join/#asterisk diegomad (mad@186.81.138.29) |
| 02:14.35 | ChannelZ | NAK |
| 02:14.36 | Fairman | infobot: help |
| 02:14.59 | *** join/#asterisk synch (~d2d46c84@gateway/web/freenode/x-gelhkfcnivmdmsnq) |
| 02:18.00 | synch | hay! anyone could help me with asterisk installation ! |
| 02:18.54 | TJNII | Maybe, if you ask a specific question |
| 02:19.15 | *** join/#asterisk OrNix (~ornix@178.49.0.149) |
| 02:19.19 | synch | yup sure ! |
| 02:19.29 | ChannelZ | There, I fixed it! |
| 02:20.03 | spenguin[work] | can I haz help, asterisk broken for me |
| 02:20.25 | ChannelZ | ctrl-alt-delete, k yr welcom |
| 02:21.00 | spenguin[work] | k |
| 02:22.56 | synch | actually installing asterisk for 1st time . i need to know do i need to install zaptel , libpri libraries also with asterisk. |
| 02:23.18 | spenguin[work] | yep all of it in proper order |
| 02:23.25 | TJNII | zaptel has been renamed to dahdi |
| 02:23.34 | TJNII | That is probably most of your problem. |
| 02:23.45 | spenguin[work] | dahdi, libpri, asterisk, spandsp |
| 02:24.32 | spenguin[work] | synch: do you have a pri card |
| 02:24.41 | spenguin[work] | only then youd need dahdi |
| 02:24.57 | ChannelZ | who yo dahdi |
| 02:25.15 | spenguin[work] | dundi |
| 02:25.50 | spenguin[work] | dahdi in an indian language means beard |
| 02:26.05 | spenguin[work] | hindi* |
| 02:26.32 | synch | pri card . i didnt get u? |
| 02:27.15 | spenguin[work] | synch: are you hooking a PSTN line to the box? |
| 02:27.52 | synch | yep |
| 02:28.51 | spenguin[work] | how |
| 02:29.25 | *** join/#asterisk sourcode (~code@ppp-58-8-236-169.revip2.asianet.co.th) |
| 02:29.35 | *** join/#asterisk tiav (~tiav@ram94-3-82-225-11-215.fbx.proxad.net) |
| 02:30.24 | synch | basically m at intial stage , just learning the installation , and further working., wasnt in much deep |
| 02:31.11 | synch | i using ebook for guidance , Asterisk , orelliy |
| 02:35.55 | Slugs_ | you should be in good hands ;) |
| 02:44.31 | *** part/#asterisk QbY (~kelvin@c-24-126-145-123.hsd1.ga.comcast.net) |
| 02:46.54 | *** join/#asterisk dotnetted (~dotnetted@75.138.79.27) |
| 03:08.38 | p3nguin | When calling a phone, after there's no answer, voicemail takes the call... press the * key to be able to check your voicemail... rather than asking for the password for that mailbox, it asks for a mailbox number first, then asks for the password. Is this normal behavior? Can it be changed to use the mailbox that you've just called rather than having to enter the mailbox number again? |
| 03:11.46 | *** join/#asterisk joobie (~joobie@mx01.anric.com.au) |
| 03:13.42 | *** join/#asterisk Deeewayne (~dwayne@206.61.30.131) |
| 03:13.42 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 03:16.34 | joobie | guys anyone know any really modern looking voip phones? been looking at the openpeak range but they seem to be all "demo" versions, not really production ready.. |
| 03:20.02 | *** join/#asterisk endemic (~endemic@76.73.14.123) |
| 03:25.39 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 03:25.57 | manxpower | ~phones |
| 03:25.58 | infobot | While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Linksys SPA-9XX, Snom, Cisco 79XX, everything else. Do not consider Grandstream phones. Ever. |
| 03:26.37 | manxpower | p3nguin: "core show application voicemail" was not helpful? |
| 03:27.20 | p3nguin | Oh, it is dialplan-configurable? Perfect! |
| 03:32.02 | *** join/#asterisk c0dyhi11 (~CodyHill@ip70-190-105-213.ph.ph.cox.net) |
| 03:38.42 | c0dyhi11 | hello. What would be the best place to ask for assistance to setup dahdi 2.3.0, Asterisk 1.6, & FreePBX 2.7.0.2? |
| 03:43.55 | manxpower | ~freepbx |
| 03:43.56 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 03:45.23 | joobie | manxpower, that's more so based on "phone quality" rather than the wow factor |
| 03:45.51 | manxpower | joobie: I wish you the BEST of luck. |
| 03:46.44 | c0dyhi11 | thanks |
| 03:46.45 | p3nguin | If I call a DID (888 321 1234) which has a Dial(SIP/400) attached to the exten which matches that DID, and SIP/400 has mailbox=400 associated with it, how can I figure out the phone device's number for use in dialplan variable to pass on to another exten? |
| 03:47.16 | p3nguin | Using Voicemail() and the 'a' extension seems hard to figure out the mailbox id I was in when I pressed the * key. |
| 03:47.19 | joobie | manxpower, those bums at openpeak dont answer their phone to see if they will sell me one |
| 03:47.31 | joobie | manxpower, but apart from taht, havent come across any that look cool |
| 03:48.52 | p3nguin | If I call that DID and the phone is not answered, I would end up in voicemail mailbox id 400, but it would be variable depending on the DID that was used to reach the SIP phones. |
| 03:49.12 | p3nguin | Pressing * doesn't leave any clue that I was in mailbox 400. |
| 03:49.37 | *** join/#asterisk riddlebox (~riddlebox@75-132-225-75.dhcp.stls.mo.charter.com) |
| 03:51.05 | p3nguin | To add more complexity, the phone could be connected via IAX2, so I can't rely on SIP channel info. |
| 03:53.19 | *** part/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 03:53.21 | p3nguin | I suppose I can create my own variable in each extension to set the mailbox id, then carry that along to the next context/extension... but I was hoping something already existed that I could parse out. |
| 03:55.45 | ChannelZ | Well you could look up the mailbox ID for SIP/400 |
| 03:57.03 | p3nguin | through an automated process? |
| 03:57.27 | ChannelZ | yeah you can get it through one of the SIPPEER functions.. trying to remember this just came up the other day |
| 03:58.01 | ChannelZ | SIPPEER(peername,mailbox) |
| 03:58.09 | *** part/#asterisk y2kdemon (~ASK@h224.68.219.67.cable.vncn.cablerocket.net) |
| 03:58.12 | p3nguin | Yes, but the phones could also be connected via IAX2, so I'm wanting to stay away from SIP-only functions. |
| 03:58.47 | p3nguin | I think this is going to be one of those things that just plain needs a variable set in each exten. |
| 03:58.56 | ChannelZ | then you probably need to keep track of it yourself |
| 03:59.01 | p3nguin | yep |
| 03:59.16 | boodu | re, i need help i can't use meetme on my vpz (openVZ), zaptel with ztdummy and zaptel module is loaded in the hd and i have made device link with vzctl set 101 --devnodes zap/pseudo:rw --save |
| 03:59.33 | boodu | for pseudo ctl timer and channel |
| 03:59.39 | ChannelZ | If it's just SIP and IAX, there is an IAXPEER function so you could do some logic to figure it out |
| 03:59.51 | boodu | but asterisk doesn't work on the vpz |
| 03:59.54 | ChannelZ | But it might be just as easy to set a var and do it yourself |
| 04:00.15 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 04:00.42 | boodu | my is message is Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) after an #asterisk -rvvv |
| 04:01.04 | boodu | I tried all the day to run correctly meetme |
| 04:01.06 | ChannelZ | then it's probably not running |
| 04:01.09 | boodu | but i loose |
| 04:02.06 | boodu | /etc/init.d/asterisk restart said : Unable to install capabilities. |
| 04:02.25 | boodu | and zaptel : /etc/init.d/zaptel: line 104: modprobe: command not found |
| 04:03.33 | p3nguin | In a macro format... phones would end up using something like Voicemail(${EXTEN}@default) anyway, so I would just add a priority right before that to take the EXTEN and set some variable such as Set(mailboxID=${EXTEN}), then Voicemail(${mailboxID}@default). For hard-coded extensions, I'll just apply the same theory. |
| 04:04.15 | boodu | the test of : head -c 0 /dev/zap/pseudo failed |
| 04:04.23 | boodu | why |
| 04:04.39 | boodu | I don't understand the problem |
| 04:05.43 | boodu | that it's after reboot the vpz 101 |
| 04:06.01 | boodu | if I make a vzctl set 101 --devnodes zap/pseudo:rw --save |
| 04:06.24 | boodu | and no reboot 101 i can do /etc/init.d/zaptel restart |
| 04:06.38 | p3nguin | ~standard extensions |
| 04:06.54 | p3nguin | ~standard-extensions |
| 04:07.02 | spenguin[work] | heh |
| 04:07.07 | p3nguin | well... |
| 04:07.15 | p3nguin | I don't know what it is. |
| 04:07.28 | p3nguin | ~standard-extens |
| 04:07.33 | p3nguin | ~standard extens |
| 04:08.06 | boodu | nobody can help me please |
| 04:08.14 | boodu | ? |
| 04:11.00 | *** join/#asterisk Torrieri (~Torrieri@nelug/crew/torrieri) |
| 04:11.05 | manxpower | boodu: if modprobe is not found then you have major operating system issues. not many people use Asterisk in VMs |
| 04:11.22 | manxpower | heck, I've never even heard of vzctl |
| 04:12.23 | boodu | ok, vzctl is the tool of openVZ |
| 04:14.18 | manxpower | have you tried the openVZ channel? (is there an openVZ channel?) |
| 04:14.20 | spenguin[work] | what do we have to do with openvz |
| 04:16.55 | boodu | how tried it ? |
| 04:17.07 | boodu | i try to use meetme |
| 04:17.15 | boodu | in the vps |
| 04:17.17 | boodu | z |
| 04:17.54 | manxpower | boodu: no! You are trying to install zaptel |
| 04:18.50 | boodu | manxpower, both i use meetme in my extension.conf |
| 04:19.00 | manxpower | *sigh* |
| 04:19.07 | k-man | rebooting the router seems to have fixed the issue i was having with asterisk connecting to my sip provider |
| 04:20.01 | boodu | zaptel is installed on my hd |
| 04:20.08 | *** part/#asterisk manxpower (~ewieling@216.186.151.147) |
| 04:23.00 | p3nguin | Now look what you did. |
| 04:25.30 | *** join/#asterisk pinoyskull (~pinoyskul@122.55.80.205) |
| 04:29.54 | boodu | bye |
| 04:30.26 | spenguin[work] | Im guessing it wasnt his native language |
| 04:30.34 | spenguin[work] | poor boodu |
| 04:32.52 | ChannelZ | I send you this file in order to have your advice |
| 04:34.14 | spenguin[work] | wut file |
| 04:45.27 | *** join/#asterisk mr_ian (~mr_ian@S0106001b63f49383.du.shawcable.net) |
| 04:48.24 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-qmgpeauimxsdexcc) |
| 04:59.17 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 05:07.15 | ChannelZ | heh.. old trojan email |
| 05:07.49 | ChannelZ | Sircam! |
| 05:14.33 | *** join/#asterisk endemic (~endemic@lynx.ipv6.onvox.net) |
| 05:15.15 | spenguin[work] | sends ChannelZ kewltetris.exe |
| 05:17.56 | *** join/#asterisk fnordus (~dnall@70.70.0.215) |
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| 05:51.59 | *** join/#asterisk nickw (~nickw@c-76-111-107-117.hsd1.md.comcast.net) |
| 05:56.41 | *** join/#asterisk sawgood (~sawgood@173-13-158-25-sfba.hfc.comcastbusiness.net) |
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| 06:00.10 | sawgood | Where can I learn more about making Asterisk into a multi tenant processs (so three business can share one PBX)? |
| 06:01.05 | carrar | research about contexts |
| 06:02.21 | sawgood | so it shouldn't be a super hard thing then |
| 06:02.42 | sawgood | I was thinking maybe each instance of Asterisk had to be run in a process ... |
| 06:03.31 | florz | no, you can run multiple asterisk processes in one process .o(?!) |
| 06:04.21 | sawgood | cool ... so If I had one PC with Asterisk on it for a small office ... all three business could have their own stand alone version of Asterisk and not interfer with the other parties |
| 06:04.26 | sawgood | I would separate out the phones via a VLAN process on the switch ... |
| 06:04.31 | sawgood | that part is easy for me ... |
| 06:04.46 | *** join/#asterisk nickw (~nickw@c-76-111-107-117.hsd1.md.comcast.net) |
| 06:05.12 | sawgood | I have a customer who talked to the other partners in the building and all three now want to pitch in and have a PBX, but have it be separate processes |
| 06:06.15 | sawgood | I was thinking I had to deal with a special program like "Thirdlane" to get this done |
| 06:06.19 | sawgood | or put in 3 PCs |
| 06:06.39 | florz | sawgood: you do have some clue what a process is? |
| 06:07.46 | *** join/#asterisk AndyF (~a@212.248.238.62) |
| 06:07.54 | *** join/#asterisk pinoyskull (~pinoyskul@124.6.182.55) |
| 06:08.30 | sawgood | I do ... I do ... I'm using the wrong terms to describe what I want to do ... |
| 06:08.32 | sawgood | I'm sorry |
| 06:09.15 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 06:20.34 | spenguin[work] | is it possible to have asterisk playback() .mp3 files? |
| 06:20.44 | spenguin[work] | or do they all need to be WAV/gsm |
| 06:20.56 | Corydon76-dig | It is, by using -addons |
| 06:21.08 | Corydon76-dig | format_mp3 is what you're looking for |
| 06:21.12 | spenguin[work] | hrm |
| 06:21.34 | ChannelZ | "but why?" |
| 06:23.17 | spenguin[work] | for whatever reason, I get |
| 06:23.18 | carrar | sawgood, you can have 1 Asterisk process serve all 3 companies |
| 06:23.18 | spenguin[work] | [Apr 28 11:55:31] WARNING[7569]: format_wav.c:95 check_header: Does not begin with RIFF |
| 06:23.22 | spenguin[work] | on wav files |
| 06:23.30 | spenguin[work] | that means I need -addons? |
| 06:24.19 | ChannelZ | no that means your wave file is probably not a wave file |
| 06:24.33 | ChannelZ | or has some extra crud on it perhaps |
| 06:24.35 | spenguin[work] | twas converted from mp3 using lame |
| 06:24.40 | ChannelZ | what does 'file' say about it in a shell? |
| 06:25.19 | spenguin[work] | 040510_MakeMySite_Message1.wav: MPEG ADTS, layer III, v1, 64 kBits, 44.1 kHz, Monaural |
| 06:25.22 | Corydon76-dig | Most sound files need to be in 8000Hz format |
| 06:25.22 | spenguin[work] | hrm |
| 06:25.50 | spenguin[work] | Corydon76-dig: asterisk wont work with em if they arent in 8hz? |
| 06:25.56 | Corydon76-dig | That's MPEG format, not WAV format |
| 06:26.02 | Corydon76-dig | spenguin[work]: correct |
| 06:26.11 | spenguin[work] | ok |
| 06:26.12 | Corydon76-dig | 8kHz |
| 06:26.21 | spenguin[work] | yok |
| 06:27.10 | Corydon76-dig | If they were in 8Hz format, the highest sound they could produce would be 4Hz, which is well below human hearing |
| 06:27.32 | spenguin[work] | heh k |
| 06:27.38 | ChannelZ | But your bling would be ROCKIN IT |
| 06:28.03 | spenguin[work] | http://blogs.terrorware.com/geoff/2006/05/26/converting-mp3-to-wav-using-lame/ |
| 06:28.07 | spenguin[work] | that is wrong then |
| 06:28.20 | spenguin[work] | cause thats what I refered to to convert these files |
| 06:29.28 | *** join/#asterisk sprite-- (~sprite@c-98-251-108-29.hsd1.ga.comcast.net) |
| 06:29.57 | sprite-- | Does anyone have experience setting up a 900 number with asterisk? |
| 06:30.28 | carrar | exten => _900NXXXXXX,1,Noop |
| 06:30.36 | carrar | setup away! |
| 06:30.40 | ChannelZ | Yeah, but Festival is not really sexy for dirty talk. |
| 06:30.50 | *** join/#asterisk coppice (~chatzilla@m121-202-68-246.smartone-vodafone.com) |
| 06:31.05 | sprite-- | I meant more if anyone have any experience with companies that assign 900 and handles billing and support for example SIP so I can set it up with Asterisk. |
| 06:34.27 | carrar | Why would SIP be any different for a 900 number then any other number? |
| 06:35.15 | carrar | Your call durations are going to be in the CDR |
| 06:38.24 | *** join/#asterisk ktwilight[m] (~ktwilight@91.179.115.17) |
| 06:38.52 | spenguin[work] | do I need to rebuilt asterisk if Im installing asterisk-addons? |
| 06:38.57 | spenguin[work] | or just a restart should be fine? |
| 06:39.06 | carrar | Just load in the module |
| 06:42.36 | spenguin[work] | hrm, I have the mp3 module loaded |
| 06:42.46 | spenguin[work] | but still cant playback |
| 06:42.58 | spenguin[work] | [Apr 28 12:14:37] WARNING[9113]: app_playback.c:471 playback_exec: ast_streamfile failed on DAHDI/6-1 for en/sterco/MP3file |
| 06:43.28 | carrar | sure it's a mp3? |
| 06:44.13 | carrar | try restart |
| 06:44.31 | spenguin[work] | Mp3file.mp3: Audio file with ID3 version 23.0 tag, MP3 encoding |
| 06:44.33 | spenguin[work] | did |
| 06:45.01 | sprite-- | carrar: I'm trying to find a reputable company that will do 900 number with sip termination. |
| 06:45.19 | spenguin[work] | rebuild asterisk? |
| 06:45.40 | carrar | shouldn't need too |
| 06:46.58 | spenguin[work] | ok what do I do now :s |
| 06:48.09 | spenguin[work] | guess I need to find out how to correctly convert this mp3 to WAV |
| 06:50.15 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 07:06.28 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 07:06.44 | spenguin[work] | is there any way to figure out if a bri channel is bonded |
| 07:06.55 | spenguin[work] | it would be at 128kbits right? |
| 07:08.13 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 07:09.09 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 07:13.10 | spenguin[work] | YAY |
| 07:13.19 | spenguin[work] | we have WAV |
| 07:14.36 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 07:15.22 | Preatorian | Party \\o o// \o/ |
| 07:15.42 | spenguin[work] | \0| |
| 07:15.49 | spenguin[work] | hrm ignore |
| 07:17.15 | spenguin[work] | format_wav.c:131 check_header: Not in mono 2 |
| 07:17.18 | carrar | spenguin, correct single isdn channels are 64k |
| 07:17.19 | spenguin[work] | hrmpf |
| 07:17.50 | spenguin[work] | carrar: is there anyway i can figure out from my end if the channels are bonded |
| 07:18.02 | spenguin[work] | cause my telco says 1bri line would just give us 1 channel |
| 07:18.07 | carrar | I've not worked with a BRI Card |
| 07:18.09 | spenguin[work] | is that non standard |
| 07:18.15 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 07:18.31 | carrar | Using the channels for data or voice? |
| 07:18.44 | spenguin[work] | voice |
| 07:18.54 | carrar | shouldn't need to be "bonded" then |
| 07:19.03 | carrar | just make sure they roll over to the next channel is all |
| 07:19.12 | spenguin[work] | roll over? |
| 07:19.21 | carrar | your cals if 1st channel is busy |
| 07:19.23 | *** join/#asterisk pinoyskull_ (~pinoyskul@122.55.80.205) |
| 07:19.27 | carrar | calls |
| 07:19.29 | spenguin[work] | ah |
| 07:19.38 | spenguin[work] | well as of now we can just make 1 call per bri line |
| 07:19.47 | carrar | correct |
| 07:19.51 | spenguin[work] | ? |
| 07:19.57 | spenguin[work] | I thought it was two channels |
| 07:20.07 | carrar | if thats what you purchased |
| 07:20.12 | carrar | dual channel ISDN |
| 07:20.22 | carrar | from the carrier |
| 07:20.32 | carrar | but ISDN can be single or dual channel |
| 07:21.03 | carrar | You show both channels up? |
| 07:26.39 | spenguin[work] | carrar: |
| 07:26.46 | carrar | spenguin[work]: |
| 07:27.02 | spenguin[work] | eh ill pastebin |
| 07:27.14 | spenguin[work] | http://pastebin.com/nSiTqf4h |
| 07:27.19 | spenguin[work] | that looks cool right? |
| 07:28.01 | carrar | What does the dahdi tools say? |
| 07:28.20 | carrar | I've nto used pri so I don't know what it should look like |
| 07:28.23 | carrar | err BRI |
| 07:28.26 | carrar | I use PRI's |
| 07:29.05 | spenguin[work] | RED B4XXP (PCI) Card 0 Span 1 |
| 07:29.10 | spenguin[work] | RED B4XXP (PCI) Card 0 Span 2 |
| 07:29.14 | spenguin[work] | OK B4XXP (PCI) Card 0 Span 3 |
| 07:29.22 | spenguin[work] | OK B4XXP (PCI) Card 0 Span 4 |
| 07:29.26 | spenguin[work] | its a 2port BRI card |
| 07:29.37 | carrar | I'm gonna assume they are dual channels then |
| 07:29.43 | spenguin[work] | yeap |
| 07:29.45 | carrar | can you send calls out each channel? |
| 07:29.58 | carrar | up to you to roll over to the next open channel (group) |
| 07:30.06 | spenguin[work] | dude |
| 07:30.14 | spenguin[work] | but RED indicates |
| 07:30.16 | carrar | and up to your carrier ot make sure they roll over incoming calls to the next unbusy channel |
| 07:30.16 | spenguin[work] | its down right? |
| 07:30.25 | carrar | RED is down |
| 07:30.35 | carrar | I personally would use span 1 & 2 |
| 07:30.41 | ChannelZ | If you swap cables do they switch? |
| 07:30.48 | spenguin[work] | yeah ill check |
| 07:32.15 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 07:33.21 | spenguin[work] | why does dahdi show status |
| 07:33.28 | spenguin[work] | show the channel as "inservice" |
| 07:33.36 | spenguin[work] | when its RED |
| 07:33.44 | spenguin[work] | isnt that incorrect |
| 07:34.20 | spenguin[work] | http://pastebin.com/nSiTqf4h |
| 07:36.05 | carrar | cause you have it configured |
| 07:36.17 | *** join/#asterisk TimeRider (steve@5ac318c1.bb.sky.com) |
| 07:36.17 | spenguin[work] | ok |
| 07:36.26 | spenguin[work] | in service but down |
| 07:36.39 | spenguin[work] | that doesnt sound good :p |
| 07:37.37 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
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| 07:37.38 | *** join/#asterisk btsteve (~tstevens@24-196-234-39.dhcp.gwnt.ga.charter.com) |
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| 07:42.13 | creativx | god damnit astmanproxy is doing suicide le grande |
| 07:45.57 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 07:49.51 | spenguin[work] | carrar: |
| 07:49.58 | spenguin[work] | ChannelZ, carrar turns out |
| 07:50.04 | spenguin[work] | 1 channel per bri is down |
| 07:50.37 | carrar | possible that you ordered 2, 1 channel ISDN lines |
| 07:50.47 | carrar | instead of 2, 2 channel ISDN lines |
| 07:51.19 | spenguin[work] | so asterisk could show a bri line with 2 channels |
| 07:51.20 | spenguin[work] | ? |
| 07:51.31 | spenguin[work] | even though I ordered it with 1 channel? |
| 07:51.41 | carrar | don't know |
| 07:51.55 | carrar | But I would think there would be a tool that would tell you that |
| 07:52.40 | spenguin[work] | hrm |
| 07:55.23 | carrar | bbl |
| 07:55.31 | ChannelZ | You should check with your telco and find out what the hell they did |
| 07:55.39 | carrar | or what was ordered |
| 07:55.46 | carrar | BOTH |
| 07:56.06 | carrar | Ask them "Just who do you think you are"!! |
| 07:56.07 | ChannelZ | If you had 2 lines with 1 channel each I would expect to not see them grouped the way they were, but maybe they screwed up and activated 1 line with 2 channels |
| 07:56.18 | ChannelZ | Easy way to test, unplug one, try to call out each channel.. |
| 07:56.41 | spenguin[work] | ChannelZ: thing is this is remotely located, corresponding with someone |
| 07:57.27 | spenguin[work] | ChannelZ: at the momemnt - we can just have two simulataneous calls even thought there are 2 bri lines |
| 07:57.48 | ChannelZ | Well only going by your pastes I would say you have 1 line with 2 channels on it |
| 07:57.59 | spenguin[work] | yes |
| 07:58.14 | spenguin[work] | did you check the Alarms paste? |
| 07:58.43 | ChannelZ | that spans 1-2 are red, yes |
| 07:59.07 | spenguin[work] | yeah so when I got 1 of the lines unplugged - one OK turned red |
| 07:59.15 | spenguin[work] | so it was RED RED RED OK |
| 07:59.28 | spenguin[work] | that would mean, just 1 channel is up on each bri |
| 07:59.31 | ChannelZ | ok so maybe you do have a single channel on each then |
| 07:59.37 | spenguin[work] | yeah |
| 07:59.48 | ChannelZ | But I haven't seen your dahdi config as to why they would be grouped as they were earlier |
| 08:00.02 | spenguin[work] | hrm ill check the config |
| 08:00.24 | ChannelZ | plus I don't deal with ISDN and as such have never used that card, so maybe that's just the way it is., |
| 08:00.31 | *** join/#asterisk war9407 (war@liquidswords.org) |
| 08:01.26 | spenguin[work] | ChannelZ: http://pastebin.com/1LkXr7YB |
| 08:01.59 | spenguin[work] | could config affect how many lines show OK? |
| 08:02.04 | spenguin[work] | s/lines/channels |
| 08:04.35 | ChannelZ | sure |
| 08:05.53 | ChannelZ | do you have other hardware in this box? |
| 08:06.09 | spenguin[work] | hardware like? |
| 08:06.54 | ChannelZ | other telephony hardware |
| 08:07.02 | ChannelZ | analog cards or something |
| 08:07.08 | spenguin[work] | no |
| 08:07.09 | spenguin[work] | why |
| 08:07.18 | spenguin[work] | probably, Id need to check |
| 08:07.23 | *** join/#asterisk ReDNeQ (~ReDNeQ@70.114.229.58) |
| 08:07.23 | ChannelZ | just wondering why your spans show up as 3 and 4 |
| 08:08.35 | ChannelZ | again, may be normal |
| 08:08.42 | ChannelZ | what does 'dahdi_scan' show you |
| 08:08.45 | spenguin[work] | hrm |
| 08:08.52 | spenguin[work] | well the card is a 2 port BRI card |
| 08:12.14 | spenguin[work] | ChannelZ: http://pastebin.com/9CzubbF4 |
| 08:12.47 | spenguin[work] | man this stuff is interesting |
| 08:14.15 | ChannelZ | uhm |
| 08:14.46 | ChannelZ | this isn't a 4-port card? The B410P as it's saying? |
| 08:15.13 | spenguin[work] | hrm |
| 08:15.17 | spenguin[work] | it sure does say |
| 08:15.18 | spenguin[work] | 06:01.0 ISDN controller: Digium, Inc. Wildcard B410 quad-BRI card (rev 01) |
| 08:15.21 | *** join/#asterisk sulex (~sulex@dynamic-adsl-78-14-170-150.clienti.tiscali.it) |
| 08:16.06 | ChannelZ | You have a 4-port card with 2 lines connected to the second two ports (spans 3 and 4) which makes sense why the other two have alarms.. those weren't individual channels, but the whole span |
| 08:16.37 | spenguin[work] | lol |
| 08:16.37 | *** join/#asterisk Da-Geek (~Da-Geek@nat/redhat/x-jflasmpzqhdcaekg) |
| 08:16.49 | ChannelZ | now whether or not you have dual channel through your provider on each line, I dunno but again that's easy to tell, make a call out each one |
| 08:16.50 | spenguin[work] | <dude>: so damn sorry |
| 08:16.58 | spenguin[work] | <dude>: its 4 port |
| 08:17.00 | spenguin[work] | :S |
| 08:17.29 | spenguin[work] | hrm |
| 08:19.57 | ChannelZ | hrm what? |
| 08:20.00 | spenguin[work] | man I should have notived |
| 08:20.06 | spenguin[work] | 06:01.0 ISDN controller: Digium, Inc. Wildcard B410 quad-BRI card (rev 01) |
| 08:20.12 | spenguin[work] | noticed* |
| 08:20.25 | spenguin[work] | thanks ChannelZ |
| 08:20.45 | ChannelZ | So can you Dial(DAHDI/7/xxxxx) and Dial(DAHDI/8/xxxxx) ? |
| 08:20.53 | *** join/#asterisk aidinb (~Aidin@71-94-148-218.static.mtpk.ca.charter.com) |
| 08:22.02 | spenguin[work] | not yet, will |
| 08:26.05 | *** join/#asterisk sourcode (~code@ppp-61-90-16-60.revip.asianet.co.th) |
| 08:26.39 | ChannelZ | bed time |
| 08:36.25 | *** join/#asterisk e-jones (~jkastner@nat/redhat/x-hybmzlghfevtiysk) |
| 08:41.11 | carrar | w00t |
| 08:41.12 | carrar | back |
| 08:41.37 | *** join/#asterisk sourcode_ (~code@ppp-61-90-16-60.revip.asianet.co.th) |
| 08:43.07 | spenguin[work] | man |
| 08:43.14 | Preatorian | woman! |
| 08:43.15 | spenguin[work] | getting the right wav format is a pita |
| 08:43.49 | spenguin[work] | mpg123 -b 80000 -s file.mp3 | sox -t raw -r 80000 -s -w -c1 - file.wav |
| 08:43.54 | spenguin[work] | any comments? |
| 08:45.29 | kaldemar | spenguin[work]: http://www.voip-info.org/tiki-index.php?page=Convert+WAV+audio+files+for+use+in+Asterisk |
| 08:46.03 | spenguin[work] | kaldemar: kewl, looking at it |
| 08:46.53 | coppice | -r 80000 would be very hi-fi, but almost certainly not what you want |
| 08:47.22 | Preatorian | LOL true |
| 08:47.33 | Preatorian | i'm sure you mean "8000" |
| 08:48.19 | tuxx- | eugh, friggin linksys :P |
| 08:48.28 | tuxx- | 406 not acceptable, AARGH |
| 08:48.28 | tuxx- | :D |
| 08:49.20 | Preatorian | \o/ |
| 08:49.52 | *** join/#asterisk Z_God (~julius@wlan228042.mobiel.utwente.nl) |
| 08:50.06 | tuxx- | anyone know why this request fails: http://www.pastie.org/938832 |
| 08:50.10 | tuxx- | i cant seem to find it :-( |
| 08:50.16 | spenguin[work] | WE HAVE WAV |
| 08:50.24 | spenguin[work] | thanks kaldemar ! |
| 08:50.47 | spenguin[work] | I was thinking to myself asterisk should have some tools for conversion |
| 08:51.00 | Preatorian | Sox is all you need |
| 08:51.02 | spenguin[work] | coppice: ah yeah and my bad |
| 08:51.26 | spenguin[work] | format_wav.c:131 check_header: Not in mono 2 |
| 08:51.30 | spenguin[work] | thats what I was getting |
| 08:51.40 | spenguin[work] | but it works now |
| 08:51.56 | spenguin[work] | rasterisk -x "file convert file.mp3 file.wav" |
| 08:52.02 | spenguin[work] | awesomez |
| 08:52.26 | tuxx- | lol nice |
| 08:52.26 | tuxx- | ;x |
| 08:53.35 | tuxx- | anyone know why this request fails: http://www.pastie.org/938832 , been looking for 2+ hours now cant seem to find the cause of the error ;-\ |
| 08:54.37 | tuxx- | i also removed the Alert-Info: info=<Bellcore-dr4>^M header for the sip request, but still the same error ;p |
| 08:55.42 | spenguin[work] | Preatorian: this sox was not built with mp3 support |
| 08:55.46 | spenguin[work] | for whatever reason |
| 08:57.14 | Preatorian | ahh yeah |
| 09:00.39 | *** join/#asterisk sourcode (~code@ppp-61-90-16-60.revip.asianet.co.th) |
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| 09:17.10 | spenguin[work] | hrm |
| 09:17.17 | spenguin[work] | where are the music on hold music? |
| 09:17.30 | spenguin[work] | under /usr/lib/asterisk/sounds |
| 09:17.33 | *** join/#asterisk The-Bat (~The-Bat@59.162.86.164) |
| 09:17.57 | carrar | where ever you tell asterisk in the confi file |
| 09:18.00 | tuxx- | /var/lib/asterisk/moh/ |
| 09:18.01 | tuxx- | i think |
| 09:18.01 | tuxx- | ;p |
| 09:18.27 | carrar | grep directory /etc/asterisk/musiconhold.conf |
| 09:18.33 | spenguin[work] | carrar: I meant default ones |
| 09:18.40 | spenguin[work] | that come with asterisk |
| 09:18.56 | carrar | probably /var/lib/asterisk/mohmp3/ |
| 09:19.05 | carrar | or /var/lib/asterisk/moh/ |
| 09:19.12 | carrar | if you did a proper install |
| 09:20.02 | *** join/#asterisk TimeRider (~steve@109.224.131.68) |
| 09:22.38 | tuxx- | anyone know why this request fails: http://www.pastie.org/938832 , been looking for 2+ hours now cant seem to find the cause of the error ;-\ |
| 09:25.22 | *** join/#asterisk joobie (~joobie@CPE-124-180-8-218.lns1.lon.bigpond.net.au) |
| 09:26.21 | spenguin[work] | yep its under the moh director |
| 09:28.29 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
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| 09:28.51 | spenguin[work] | how exactly do I use moh with Dial() |
| 09:29.08 | spenguin[work] | it says something like Dial(..,m(class)) |
| 09:29.13 | spenguin[work] | what does that maen |
| 09:29.15 | spenguin[work] | mean* |
| 09:29.27 | spenguin[work] | m: Provide Music on Hold to the calling party until the called channel answers. This is mutually exclusive with option 'r', obviously. Use m(class) to specify a class for the music on hold. |
| 09:29.44 | spenguin[work] | or nvm |
| 09:30.32 | spenguin[work] | I think I get it |
| 09:31.55 | spenguin[work] | why does moh need a application |
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| 09:53.04 | kruemeltee | hello everybody :-) |
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| 09:53.47 | carrar | HARRO |
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| 10:13.18 | spenguin[work] | carrar: HERRO* |
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| 10:45.50 | FreezeS | hey guys |
| 10:46.12 | FreezeS | I've got a problem with a Digium TE121E card |
| 10:46.21 | FreezeS | every 2 minutes I get things like this: RI got event: HDLC Abort (6) on Primary D-channel of span 1 |
| 10:46.40 | FreezeS | and the line disconnects after a random interval |
| 10:47.33 | FreezeS | I've disabled the serial ports and audio to free some interrupts |
| 10:48.00 | FreezeS | I've read that setting some hdd parameters can fix this, but this doesn't work on that server |
| 10:48.18 | FreezeS | any hints on what else to check ? |
| 10:49.09 | *** join/#asterisk AsteriskNoob (~AsteriskN@host86-128-145-94.range86-128.btcentralplus.com) |
| 10:49.19 | AsteriskNoob | Hello all. |
| 10:49.22 | *** join/#asterisk pawz (~pawz@ppp118-208-8-27.lns20.bne1.internode.on.net) |
| 10:49.37 | AsteriskNoob | Can anyone tell me how to get two sip accounts from the same provider working? |
| 10:50.11 | AsteriskNoob | They register, but calls show as coming from the second account only. |
| 10:50.44 | pawz | how do i create the astdb ? |
| 10:54.31 | kaldemar | pawz: you'll have it upon asterisk installation. are you looking for usage instructions? |
| 10:55.14 | kaldemar | AsteriskNoob: be more specific, it depends on the provider. show a SIP trace of calls and your sip configuration. |
| 10:55.58 | AsteriskNoob | kaldemar: I don't have a trace to show right now. The provider is SipGate.co.uk |
| 10:56.23 | AsteriskNoob | kaldemar: I'm working on a production box right now so I can't upset it yet. |
| 10:58.12 | pawz | kaldemar, my bad. i didn't chown /var/lib/asterisk |
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| 11:03.51 | *** join/#asterisk dwjv (~IceChat7@88-96-95-134.dsl.zen.co.uk) |
| 11:05.05 | dwjv | hi, does anybody have any experience with UK NTL/Virgin media ISDN30e1 and asterisk? |
| 11:07.59 | spenguin[work] | FreezeS: heh Im having the same issue |
| 11:08.07 | spenguin[work] | calls seem to be ok though |
| 11:08.09 | spenguin[work] | as of now |
| 11:08.27 | FreezeS | I could make calls 10 min long |
| 11:08.28 | spenguin[work] | sometimes need to call multiple times before I get the number |
| 11:08.38 | FreezeS | but there are random times when the line cuts |
| 11:08.48 | spenguin[work] | hrm, havent tried long calls |
| 11:08.51 | spenguin[work] | yet |
| 11:08.55 | FreezeS | I've had the exact same problem with an openvox pci card |
| 11:09.03 | FreezeS | and changed it with a digium pci express card |
| 11:09.26 | FreezeS | and it's exactly the same |
| 11:11.02 | spenguin[work] | im on a digium card |
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| 11:32.05 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 11:44.48 | sun28 | moin |
| 11:46.44 | *** join/#asterisk coppice (~chatzilla@m121-202-91-103.smartone-vodafone.com) |
| 11:47.57 | *** join/#asterisk hodendc (~hodendc@dhcp-077-250-093-072.chello.nl) |
| 11:48.02 | hodendc | how do I see ongoing calls in CLI? |
| 11:48.11 | gelo | core show channels |
| 11:48.34 | *** join/#asterisk sun28 (~light@78.108.73.46) |
| 11:50.13 | hodendc | thanks! |
| 11:51.08 | hodendc | My colleagues here no voice when they call me. |
| 11:51.12 | hodendc | But line does go over. |
| 11:51.16 | hodendc | If I call them, it's fine |
| 11:51.48 | beek | ~NAT |
| 11:51.49 | infobot | i heard nat is Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 11:52.10 | hodendc | It worked fine and the config was not touched. |
| 11:52.22 | hodendc | But there have been issues a lot of times lately and we lack Asterisk knowledge, mine is limited. |
| 11:52.32 | hodendc | But they have internet with 700ms ping |
| 11:52.35 | hodendc | but it used to work fine |
| 11:53.20 | gelo | if it was working and you did not touch anything... |
| 11:53.29 | gelo | not your fault |
| 11:53.34 | gelo | :) |
| 11:53.49 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-fjwbvznewzzdqtrh) |
| 11:53.50 | gelo | maybe your isp have some issues? |
| 11:54.17 | hodendc | I have called them, they say it's fine. |
| 11:54.42 | hodendc | I have called the ISP, they say they have no issues. |
| 11:54.47 | hodendc | I have also called my colleagues and that does work. |
| 11:54.50 | hodendc | But they cannot call me. |
| 11:54.54 | hodendc | or anybody else |
| 11:55.00 | hodendc | they can receive incoming calls, not place outgoing |
| 11:55.49 | gelo | take a look at nat again then |
| 11:55.50 | *** join/#asterisk grEvenX (~even@apb9hb.ip.ssc.net) |
| 11:55.53 | Yudaisrael1984 | it might be a codec incompatibilty |
| 11:56.18 | hodendc | The VOIP manager says it was not related to their VOIP Server. |
| 11:56.25 | hodendc | and I have no idea if I can trust that :( |
| 11:56.35 | hodendc | codec, hmm, |
| 11:57.17 | Yudaisrael1984 | can they call anyone else or is it only to u that they cant call? |
| 12:00.11 | *** join/#asterisk coppice (~chatzilla@m121-202-31-18.smartone-vodafone.com) |
| 12:00.36 | *** join/#asterisk rttrey (~trey@andc-office-fw.atlantic.net) |
| 12:00.53 | gelo | I'm trying to integrate kamailio and asterisk |
| 12:01.17 | gelo | but when i make a transfer in a call established between kamailio and asterisk |
| 12:01.37 | gelo | it does not work |
| 12:01.53 | gelo | and the call gets stuck in asterisk and the transfer-to |
| 12:02.00 | gelo | has anyone tried this? |
| 12:06.57 | *** part/#asterisk rttrey (~trey@andc-office-fw.atlantic.net) |
| 12:07.17 | dwjv | Has anybody used asterisk with an NTL/Virgin connection before? |
| 12:08.46 | dwjv | I can't seem to get any signal on the digium card |
| 12:09.05 | hodendc | Yudaisrael1984: They cannot call anybody |
| 12:09.14 | hodendc | They cannot send outgoing calls, but can receive incoming calls |
| 12:13.42 | hodendc | What could be the reason my colleagues can receive incoming calls, but not outgoing calls? |
| 12:15.03 | Yudaisrael1984 | try using g711 as the preffered codec and make sure ilbc and g711a is working |
| 12:15.08 | Yudaisrael1984 | and available |
| 12:15.29 | Yudaisrael1984 | is there a different channel for a2billing? |
| 12:15.32 | hodendc | How can I see in sip debug, what happens wrong? |
| 12:15.46 | hodendc | Where do I have to look for? |
| 12:16.01 | Yudaisrael1984 | run the cli and do a sip debug on that peer |
| 12:16.08 | Yudaisrael1984 | and have them try making a call |
| 12:16.12 | Yudaisrael1984 | then paste the output |
| 12:16.33 | hodendc | I have problems with publically posting asterisk logs :( |
| 12:16.46 | Yudaisrael1984 | pastebin |
| 12:16.57 | Yudaisrael1984 | then no one here can help you |
| 12:17.12 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 12:17.46 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 12:18.05 | hodendc | I guess that's true. |
| 12:18.07 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 12:18.10 | hodendc | welcome [TK]D-Fender |
| 12:18.18 | ariel_ | Morning folks |
| 12:19.53 | hodendc | Restricting registration for peer 'bu' to 60 seconds (requested 30) < unimportant right? |
| 12:20.04 | hodendc | hi ariel_; how's the wheather there? |
| 12:20.40 | dwjv | Is anybody able to assist me with a ISDN30 issue? |
| 12:20.41 | ariel_ | it's picture perfect, one of those days that sucks being inside |
| 12:20.55 | ariel_ | what is your isdn30 issue? |
| 12:21.27 | dwjv | we have an asterisk system intercepting a pbx and isdn line but i'm not getting any signal from either side |
| 12:21.46 | dwjv | I've confirmed with the telco thats it's isdn30e1 |
| 12:22.26 | dwjv | but im not seeing any data, pri intense debug reveals absolutly no data |
| 12:22.29 | ariel_ | where are you located? |
| 12:22.32 | dwjv | uk |
| 12:22.46 | ariel_ | your setting should be euroISDN |
| 12:22.50 | dwjv | it is |
| 12:23.14 | dwjv | i had a feeling it might be a setting in zaptel.conf |
| 12:23.15 | ariel_ | can you post your system.conf and your chan_dahdi.conf on pastebin.com or any other pastebin |
| 12:23.17 | dwjv | but it looks fine |
| 12:23.26 | ariel_ | your running zaptel? |
| 12:23.27 | dwjv | it's zaptel |
| 12:23.29 | dwjv | yeah |
| 12:23.37 | ariel_ | ~pb |
| 12:23.38 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.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. |
| 12:23.54 | ariel_ | please post your settings and if you have any errors when you make the calls |
| 12:24.00 | ariel_ | pri show spans |
| 12:24.51 | *** join/#asterisk pinoyskull (~pinoyskul@202.138.177.34) |
| 12:27.45 | dwjv | PRI span 1/0: Provisioned, In Alarm, Down, Active |
| 12:27.45 | dwjv | PRI span 2/0: Provisioned, In Alarm, Down, Active |
| 12:29.27 | ariel_ | dwjv: you should not pm this info as others can see it and could help |
| 12:29.44 | ariel_ | most UK setups that I know of don't use crc4 |
| 12:29.46 | dwjv | sorry, http://pastebin.com/06fidwWF |
| 12:29.57 | dwjv | most of the ones we do, do |
| 12:30.37 | dwjv | but i've tried that anyway |
| 12:30.45 | ariel_ | what about your other file? |
| 12:31.00 | hodendc | Yudaisrael1984: Apr 28 09:28:03 WARNING[2512] channel.c: Failed to write data to channel monitor read stream |
| 12:31.01 | hodendc | Apr 28 09:28:03 WARNING[2512] translate.c: No translator path from alaw to g723 |
| 12:31.01 | hodendc | Apr 28 09:28:03 WARNING[2512] file.c: Unable to translate to format wav49, source format g729 |
| 12:31.01 | dwjv | sorry, just trying to figure out how to capy all on vi :) |
| 12:31.03 | hodendc | could that be the reason? |
| 12:31.47 | ariel_ | hodendc: seems you don't have the codec for your system |
| 12:31.49 | hodendc | They can receive incoming calls, but outgoing calls have no voice. |
| 12:31.54 | hodendc | It used to work fine for a long time. |
| 12:31.59 | hodendc | But stopped working since 2 days ago. |
| 12:32.08 | ariel_ | something has changed |
| 12:32.14 | Yudaisrael1984 | as u can see its a codec issue |
| 12:32.39 | dwjv | http://pastebin.com/ztg0h5k3 |
| 12:32.48 | dwjv | thats zapata.conf |
| 12:34.00 | ariel_ | so one is net and the other is cpe |
| 12:34.19 | dwjv | yeah |
| 12:34.49 | ariel_ | any reason your still on zatel? |
| 12:35.14 | dwjv | we had problems when dahdi was first realeased so we rolled back |
| 12:35.24 | dwjv | have got a tested version using dadhi yet |
| 12:35.24 | *** join/#asterisk beefpastry (~tmr@74-129-198-56.dhcp.insightbb.com) |
| 12:35.33 | dwjv | haven't* |
| 12:38.25 | hodendc | Yudaisrael1984, ariel_: Thanks, VOIP provider takes blame now. |
| 12:40.04 | Yudaisrael1984 | no problem |
| 12:40.18 | Yudaisrael1984 | happy that i can help u i just wish people would help me on my issues lol |
| 12:40.29 | Yudaisrael1984 | yours is the easy stuff mine is the mind breaking ones |
| 12:42.01 | Preatorian | Which one is that Yudaisrael1984 ? |
| 12:42.20 | Yudaisrael1984 | what do u mean by which is is that? |
| 12:42.37 | Preatorian | 14:40 < Yudaisrael1984> yours is the easy stuff mine is the mind breaking ones |
| 12:42.48 | Yudaisrael1984 | i wrote that to hodendc |
| 12:42.58 | hodendc | How can I help you Yudaisrael1984? |
| 12:43.13 | hodendc | I know 0,342% of Asterisk. |
| 12:43.17 | hodendc | so maybe we're lucky. |
| 12:43.18 | Yudaisrael1984 | lol do u understand a2billing? |
| 12:43.20 | Yudaisrael1984 | hehe |
| 12:44.08 | Yudaisrael1984 | i have it set up again after not touching it in 2 years and i run a agi debug and it gets half way and just exits with a return code of 0 |
| 12:44.34 | Yudaisrael1984 | no explanation what went wrong or anything |
| 12:44.58 | *** join/#asterisk hodendc (~hodendc@dhcp-077-250-093-072.chello.nl) |
| 12:45.07 | hodendc | Yudaisrael1984: Sorry mate, hehe. :p |
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| 13:17.40 | pigpen | Has anyone seen where, on 1.6.1.x, of course running DAHDI, that "ZAP" shows up in verbose output? Please note, my friend grep confirms the word ZAP (lower/upper/combo) does not exist anywhere under /etc/ |
| 13:18.16 | pigpen | Yes, this "oddity" was affecting handling of the FXO/FXS ports. I had to reload chan_dahdi to resolve. |
| 13:18.35 | pigpen | wonders if he still has the output..... |
| 13:19.50 | pigpen | Specifically I am running 1.6.1.18 on 64bit Gentoo. |
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| 13:21.17 | *** join/#asterisk rgsteele (~rgsteele@207.106.239.81) |
| 13:23.44 | pabelanger | pigpen: If you can get a debug log showing it, we could track it down. |
| 13:24.13 | pigpen | heh....I am hoping I have the standard output at this point. ;-) |
| 13:24.20 | pigpen | but if it happens again, yeah. |
| 13:24.26 | pigpen | it was "odd" |
| 13:24.55 | dwjv | ariel_ you couldn't think of anything for my euroISDN problem? I have a feeling it's some kind of signalling problem but we've never seen it before... |
| 13:24.59 | pabelanger | Not all references to ZAP have been removed, I believe there are still some in documentation. |
| 13:25.28 | pigpen | hmm... Well, in short, my wife (who is my tester) says, the phones don't work. |
| 13:25.36 | pigpen | sure enough, they don't. |
| 13:25.54 | ariel_ | your going to have to do some research into this but your correct dwjv , Most of the ones I setup over 300 now are all using EuroISDN and no crc4 |
| 13:26.04 | pigpen | Looking into the standard output, "zap" was being called. |
| 13:26.14 | pigpen | I'll go see if I at least have the output. |
| 13:26.43 | *** join/#asterisk muiro (~muiro@unaffiliated/muiro) |
| 13:26.45 | pigpen | may be worth opening a bug as a place holder, but I for one hope to never see it again. |
| 13:27.17 | pigpen | It should be in the log now that I think about it. |
| 13:27.26 | *** part/#asterisk muiro (~muiro@unaffiliated/muiro) |
| 13:27.46 | dwjv | ok, thanks for your help anyway :) |
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| 13:30.30 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 13:30.58 | tzafrir | pigpen, you use 'ZAP' instead of DAHDI somewhere in your dialplan? |
| 13:31.10 | tzafrir | What's the actual Dial() you use? |
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| 13:32.05 | pigpen | I actually tried forcing it with inserting DAHDI instead of a variable like ${TRUNK} |
| 13:32.09 | pigpen | but it still borked. |
| 13:32.25 | [TK]D-Fender | pigpen: Do you have something to actually show us? |
| 13:32.32 | pigpen | heh...working on it. |
| 13:32.34 | pigpen | :) |
| 13:32.40 | *** join/#asterisk superbeef (~lanej@74.84.194.4) |
| 13:32.50 | pigpen | but when I do, I want to be sure, rather than wasting everyone's time. |
| 13:33.01 | pigpen | hopefully that is appreciated. |
| 13:33.04 | [TK]D-Fender | pigpen: ... you already have... |
| 13:33.16 | pigpen | Dam you sound like my business partner. |
| 13:33.17 | [TK]D-Fender | pigpen: even thinking about it |
| 13:33.29 | pigpen | yeah, wasting those cycles.... |
| 13:33.43 | *** join/#asterisk nitram (foo@superblob.com) |
| 13:34.38 | pigpen | as I do "try shit" on this box, I am trying to make sure I don't have anything goofy going on. |
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| 13:39.57 | Phurl_ | hi all, is is possible to have an extension that is the user name, not numeric? |
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| 13:42.55 | nitram | hi guys, how would i do dial(${foo1}&${foo2},10); dial(${foo1}&${foo2}&${foo3}) so that after 10 seconds extension foo3 rings in addition to 1 and 2 |
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| 13:47.10 | leifmadsen | nitram: Local channel |
| 13:47.22 | ctooley | Is there a way to set the OLI on a call I am dialing? |
| 13:47.35 | nitram | leifmadsen: local channel? |
| 13:47.39 | leifmadsen | Dial(Local/${foo3}@delayed_dial&${foo1}&${foo2}) |
| 13:47.42 | leifmadsen | [delayed_dial] |
| 13:47.52 | leifmadsen | exten => _XXX.,1,Wait(10) |
| 13:48.13 | leifmadsen | exten => exten => _XXX.,n,Dial(SIP/${EXTEN}) |
| 13:48.16 | *** join/#asterisk manxpower (~ewieling@216.186.151.147) |
| 13:48.17 | nitram | ah |
| 13:48.19 | leifmadsen | errr... ignore the double exten => |
| 13:48.20 | nitram | interesting |
| 13:48.22 | leifmadsen | but you get the idea |
| 13:48.22 | manxpower | ~mailinglist |
| 13:48.23 | infobot | [~mailinglist] The Asterisk mailing lists can be found at http://lists.digium.com , http://www.asteriskguru.com/archives Search the archives by adding "site:lists.digium.com" to your Google search. |
| 13:48.24 | manxpower | ~answers |
| 13:48.25 | infobot | [~answers] Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & Docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: /path/to/src/asterisk/UPGRADE*.txt |
| 13:48.28 | leifmadsen | wtf |
| 13:48.41 | leifmadsen | that's your default join action? |
| 13:48.51 | manxpower | leifmadsen: no. |
| 13:49.04 | vik_ | i have installed asterisk from its source code. but when i run it by 'asterisk -vvvc' , it shows errors as " No 'modules.conf' found " and "Can't find indications config file indications.conf". |
| 13:49.04 | *** join/#asterisk synch (~d2d46c84@gateway/web/freenode/x-nlqxgoztrrtptbis) |
| 13:49.16 | manxpower | ~toolkit |
| 13:49.17 | infobot | i guess toolkit is Remember Asterisk isn't really a PBX. Asterisk is a TOOLKIT that helps you build a PBX from scratch, much like libraries help you build an application from scratch. |
| 13:49.26 | leifmadsen | vik_: you're missing those files obviously |
| 13:49.27 | nitram | leifmadsen: do you see any possibility to playback an announcement at the same time when foo3 gets dialed? |
| 13:49.28 | manxpower | vik_: did you configure Asterisk and build the config files? |
| 13:49.29 | leifmadsen | vik_: make samples |
| 13:49.41 | ctooley | vik_, make sure your configuration files are in /etc/asterisk (assuming you didn't modify the paths when you ran configure). |
| 13:49.49 | leifmadsen | nitram: probably not |
| 13:49.59 | leifmadsen | there is probably a way, but I can't think of it |
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| 13:50.07 | leifmadsen | maybe something clever using music on hold |
| 13:50.07 | nitram | leifmadsen: that was my impression aswell, when i was looking at playback and background aswell |
| 13:50.41 | leifmadsen | playing back something from background or playback will answer the channel |
| 13:50.58 | vik_ | ahh...make samples was necessary |
| 13:51.03 | vik_ | it works now |
| 13:51.26 | manxpower | vik_: remember the sample config files are not designed to be a working system |
| 13:51.31 | nitram | leifmadsen: exactly. and not continue to ring some extension |
| 13:51.45 | nitram | leifmadsen: i was also thinking about using queues |
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| 13:53.09 | manxpower | Does anyone know if a way to make "service asterisk status" work when running as non-root? |
| 13:54.00 | vik_ | manxpower : i did configure asterisk. wat does it mean that sample config files are not designed to be a working system ? |
| 13:54.41 | leifmadsen | vik_: it means they are samples, not production ready files |
| 13:55.06 | leifmadsen | be careful about what files you use. You could also have just copied the files from /usr/src/asterisk/configs/indications.conf.sample and modules.conf.sample |
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| 13:57.37 | manxpower | vik_: what did you do to "configure asterisk" |
| 13:58.08 | manxpower | Configuring Asterisk involves you editing the config files and adding in your sip devices and writing your dialplan as well as many other things |
| 13:58.31 | *** part/#asterisk mpd (~chatzilla@70.28.49.95) |
| 13:58.46 | vik_ | manxpower: i Just installed the asterisk through normal procedure of instaling from source code. |
| 13:58.58 | vik_ | i am very naive on this. |
| 13:59.04 | manxpower | vik_: that is not configuring Asterisk. |
| 13:59.11 | manxpower | You should read the Asterisk book |
| 13:59.12 | manxpower | ~book |
| 13:59.13 | 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 |
| 14:00.02 | vik_ | manxpower: thanx |
| 14:00.24 | vik_ | currently i just need to call the asterisk server from zoiper soft phone. |
| 14:00.46 | manxpower | *sigh* |
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| 14:11.16 | jaytee | morning everyone |
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| 14:12.58 | devmod | morning, I know it is not possible to play anything else other than asterisk rrecorded video with Playback, but if I wanted to implement some sort of transcoder to convert video into asterisk video how would I go about it ? any pointers? |
| 14:13.31 | ManxPower-work | jaytee, run for your sanity! |
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| 14:22.36 | pigpen | Well, seems I didn't grep out the /etc/ directory for "Zap", as I found a reference to the variable TRUNK=Zap/1 in the extensions.ael. Now why the variable in the extensions.conf was being used instead for a week (been on 1.6.1.18 for a week, 1.6.x since released) I don't know |
| 14:23.48 | pigpen | The other oddity is when I did set the dial command to DAHDI instead of a variable it still borked. Maybe I pissed it off with the second variable in the default ael file. Dunno |
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| 14:26.07 | pigpen | so it doesn't seem to be a ghost in the code, just a missed sample file. |
| 14:26.33 | jaytee | ManxPower-work, sanity "left the building" decades ago :-) |
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| 14:30.17 | grandpapadot | devmod: best to build a SIP client and just dial it ... |
| 14:31.13 | Naikrovek | devmod: find out what format asterisk uses for video (h.323, probably) and use ffmpeg to convert to that |
| 14:31.41 | devmod | I dont think its as simple as that Naikrovek, the video is just an rtp dump i believe with some special hints added to it |
| 14:31.43 | Naikrovek | make sure you match height and width definitions on the video, as well as bitrate |
| 14:31.59 | Naikrovek | asterisk converts the video file to RTP, same as audio |
| 14:32.23 | Naikrovek | you can't store a file as RTP anyway |
| 14:32.58 | Naikrovek | just like you don't store SIP or IAX. you store a file (audio or video) and asterisk streams it when requested |
| 14:33.42 | Naikrovek | but that's me and i could be wrong |
| 14:33.44 | Naikrovek | best of luck |
| 14:33.50 | devmod | what? |
| 14:34.05 | devmod | there is no conversion going on rtp from the video stream |
| 14:34.13 | Naikrovek | RTP is transport |
| 14:34.19 | Naikrovek | it's not a codec or a container |
| 14:34.32 | Naikrovek | just like RTP is not a codec or a container for audio files that asterisk plays |
| 14:34.38 | devmod | right i know |
| 14:34.50 | devmod | but just simply putting a h263 encoded video wont do either |
| 14:35.02 | Naikrovek | you put a .wav of the proper format on your asterisk server, when asterisk uses Playback() to serve that, asterisk stuffs it into RTP packets |
| 14:35.41 | devmod | with video its different tho, its not just about the format |
| 14:35.46 | Naikrovek | fine |
| 14:35.50 | Naikrovek | you ask us then you tell me i'm wrong |
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| 14:35.56 | pigpen | Naikrovek, is "stuffs" a technical term? ;) |
| 14:35.58 | Naikrovek | BEST OF LUCK |
| 14:36.36 | devmod | Naikrovek, http://www.voip-info.org/wiki/view/Asterisk+video |
| 14:36.47 | devmod | "Currently, Asterisk dumps the content of RTP packets including some timing information in .h263, ph263p, .h264 files" |
| 14:36.49 | gelo | hi, when i make a call from asterisk to a phone registered in a kamailio and try a transfer, asterisks 202s the refer, but after that sends a notify with 481 Call leg/transaction does not exist |
| 14:36.55 | gelo | is this the expected behaviour? |
| 14:37.03 | gelo | because the call gets stuck... |
| 14:37.31 | jaytee | I know that it's not uncommon for some people to use #FILEINCLUDE in both extensions.conf and sip.conf but has anyone here that uses #FILEINCLUDE ever used it in voicemail.conf? |
| 14:38.00 | Takapa | is it possible to abort an extension when the user press 1, * etc? |
| 14:38.04 | *** join/#asterisk AsteriskNoob (~AsteriskN@host86-150-212-70.range86-150.btcentralplus.com) |
| 14:38.19 | Takapa | playback() and waitexten works, but i'd like a playbackwhilewaitexten()... :) |
| 14:38.51 | Takapa | (uh meant "abort playback" above, of course) |
| 14:39.04 | pigpen | jaytee, I think you can. I am trying to remember if I have or not. |
| 14:39.13 | pigpen | give it a shot. Trial and error. |
| 14:40.25 | ManxPower-work | jaytee, as I understand it #include filename is part of the asterisk config parser, so it should work for any config file. |
| 14:41.01 | ManxPower-work | Takapa, "core show application background" and "core show application read" and "core show applications" |
| 14:42.07 | Naikrovek | devmod: that's how asterisk records files. into h263 files. how is that different from what i said? timing is included in all video files for things like a/v sync. |
| 14:42.11 | Naikrovek | nevermind |
| 14:42.24 | devmod | Naikrovek, not really |
| 14:42.31 | *** join/#asterisk cjk (~cjk@80.90.44.29) |
| 14:42.38 | Naikrovek | fine, you figure it out yourself then |
| 14:42.44 | devmod | Naikrovek, anyway not here to get into an argument, just looking for some hints on where to start |
| 14:42.45 | kaldemar | Takapa: core show application Background |
| 14:43.21 | kaldemar | oops, you got that already. |
| 14:43.58 | pigpen | devmod, I think that article is stating that it dumps the timing into files. |
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| 14:44.17 | pigpen | I think it is just not worded very well. |
| 14:44.39 | gelo | is asterisk supposed to be able to transfer a call if it doesn't have the two legs of the transfer, like in sip transfer acting as ua? |
| 14:45.08 | cjk | hi, maybe someone has solved a similar problem already. all my phones are behind on multiple sites behind NAT and I would access their webinterface/api from a central point. any ideas or thought are welcome on how to achieve this |
| 14:45.44 | pigpen | devmod, I can see how it could be read that way however. Sure, it could convert/transcode a video into a different file. But video on asterisk is a far out thing anyway from what I know. |
| 14:46.41 | pigpen | cjk, vpn. |
| 14:46.50 | kaldemar | devmod: the format source files (format_h263.c and format_h264.c) mention "raw, headerless data". |
| 14:47.16 | devmod | kaldemar, oh ok thanks gonna take a look there |
| 14:47.34 | ManxPower-work | cjk, I suggest you ask on a networking channel |
| 14:47.56 | ManxPower-work | ah, a "central point" |
| 14:48.35 | cjk | pigpen, i hope to find a different way |
| 14:49.40 | pigpen | in short, you could do a central deploy (example polycom is better deployed this way and many others). |
| 14:50.03 | pigpen | But still, vpn's are very handy in many aspects. |
| 14:50.36 | pigpen | but as Manx said, this is a bit beyond the topic of this channel. |
| 14:51.11 | Naikrovek | cjk: vpn if possible, ftps over public internet if your phones support it and vpn isn't available. |
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| 15:18.25 | hwa | I'm looking for a way to dump PRI D channel data using a Digium T1 card. Perhaps something similar to OpenZap's 'oz q931_pcap'. Can anyone point me in the right direction?I'm looking for a way to dump PRI D channel data using a Digium T1 card. Perhaps something similar to OpenZap's 'oz q931_pcap'. Can anyone point me in the right direction? |
| 15:18.40 | hwa | Sorry about the double paste! |
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| 15:25.22 | raden_work | is there a easy way to fax over ethernet ? |
| 15:25.31 | raden_work | without having to scan and then use like email to fax ? |
| 15:25.36 | raden_work | Naikrovek, good morning |
| 15:26.08 | *** part/#asterisk gelo (~gelo@mx01.quobis.com) |
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| 15:27.23 | Naikrovek | raden_work: scanner + email. i'm told there are devices that do this |
| 15:28.38 | raden_work | i want something they can punch the fax number into the damn scanner LOL |
| 15:35.11 | chuckf | raden_work: isn't that called a fax machine? |
| 15:36.16 | *** join/#asterisk Brookss (~SSJGotenk@mail.sis.ca) |
| 15:36.18 | raden_work | chuckf, yea LOL but that works on VOIP |
| 15:37.26 | chuckf | good luck with that |
| 15:38.06 | raden_work | i want something that has like scan to email but can just send the email from the damn machine |
| 15:38.15 | chuckf | email is the best ethernet fax technology I've found so far |
| 15:39.06 | raden_work | http://www.sharpusa.com/ForBusiness/DocumentSystems/FaxMachines/FOIS125N.aspx |
| 15:39.17 | chuckf | raden_work: I remember some copiers I used ot work on in the 2005 era that had that built in |
| 15:39.27 | raden_work | that is cheap |
| 15:39.30 | raden_work | $125 :) |
| 15:39.34 | raden_work | looks like it will work |
| 15:39.40 | *** join/#asterisk cusco (~trilili@213.63.137.210) |
| 15:39.46 | raden_work | im the only one here that can ever scan anything so im the office B*&^% |
| 15:40.18 | *** join/#asterisk digiv (~mlhess@141.214.234.28) |
| 15:41.28 | chuckf | just remember you have to enter all the email addys in yourself more than likely |
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| 15:42.52 | beefpastry | We have a Ricoh 5060 document server that does both faxing and scan->email...I just eliminated PSTN faxing and am pushing everybody to email (but I have an email to fax through asterisk button on the Ricoh just in case). |
| 15:43.28 | beefpastry | It's a little over the $125 range, though. ;) |
| 15:45.10 | raden_work | beefpastry, we use email to fax service so i can do like faxnum@emailyourfax.com and itll be all good :) |
| 15:45.21 | raden_work | that sharp has alot of nice features was looking at manual |
| 15:45.25 | raden_work | and its laser :) |
| 15:46.57 | beefpastry | that works...seemed too expensive to outsource for me. |
| 15:47.06 | bmoraca_work | anyone at or going to Interop this week? |
| 15:47.07 | raden_work | is there a way i could set something up so it could just type the number and setup a email server to append @emailyourfax.com or something |
| 15:47.10 | raden_work | people here are lazy |
| 15:47.26 | raden_work | beefpastry, we pay $6 a month for 100 pages then 3 cents a page after that |
| 15:47.38 | raden_work | beefpastry, and our inbound just comes to our emails |
| 15:47.50 | raden_work | so saves alot of paper |
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| 15:48.14 | bmoraca_work | raden_work: you could set up Hylafax with a postfix/sendmail install to accomplish what you want |
| 15:48.36 | raden_work | bmoraca_work, thanks :) i will look into that cause i know that will be the next issue ! |
| 15:48.38 | raden_work | breb |
| 15:49.06 | bmoraca_work | raden_work: it won't be easy and will take a lot of external services and programming, but it is possible. |
| 15:49.13 | beefpastry | Yeah...when I looked up our fax output, I think we were around 200 pages a week or so. Figured a quick mail2fax script was in order. |
| 15:50.52 | *** join/#asterisk ReDNeQ (~ReDNeQ@70.114.229.58) |
| 15:50.59 | *** join/#asterisk jsjc (~chatzilla@219-90-182-151.ip.adam.com.au) |
| 15:51.20 | jsjc | why could be that after reinstalling asterisk no modules are loaded?? just res_fax and res_digium... |
| 15:51.22 | beefpastry | You'd have to read the manual, I guess...doubt the thing would do an auto-append, but you never know. If it auto-appended an @domainname, then it'd be pretty easy to relay. |
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| 15:58.56 | raden_work | beefpastry, it would be sweet LOL |
| 15:59.24 | raden_work | or even have it so a speed dial key would put the rest of the address in or something |
| 16:00.23 | beefpastry | good luck...don't think Sharp has any incentive to allow a softkey config on a $125 machine. |
| 16:00.24 | jsjc | i am tryung to troubleshoot why modules are not loading... anyone could give me a hand? |
| 16:02.30 | ManxPower-work | jsjc, "asterisk -cvvv" with asterisk not running. |
| 16:03.46 | jsjc | ManxPower-work: nothing at all.. except an asterisk without modules... |
| 16:04.04 | ManxPower-work | jsjc, then you have a config file format issue. |
| 16:04.28 | jsjc | but i still have ame config files that had form before... |
| 16:04.32 | jsjc | oohh maybe permisions on reading... |
| 16:04.34 | jsjc | let me check |
| 16:04.38 | ManxPower-work | jsjc, then what did yo change? |
| 16:04.45 | jsjc | reinstalled |
| 16:05.23 | ManxPower-work | How do you know no modules are loaded? |
| 16:05.55 | *** join/#asterisk ReDNeQ (~ReDNeQ@70.114.229.58) |
| 16:06.00 | jsjc | because i do in CLI module show and all its 0 except a few 1 |
| 16:06.59 | ManxPower-work | jsjc, that does not mean you have a modules problem |
| 16:07.17 | ManxPower-work | if the modules are NOT EVEN LISTED, then you have a modules problem |
| 16:07.28 | jsjc | they are listed |
| 16:08.24 | ManxPower-work | what verison of Asterisk |
| 16:09.02 | jsjc | 1.6.2.6 |
| 16:09.20 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.133.226.dsl.dyn.forthnet.gr) |
| 16:09.57 | jsjc | so fax and skype that are modules that i installed after they work smoothly |
| 16:10.09 | jsjc | but moduels compiled and installed with asterisk they do not |
| 16:10.55 | beefpastry | has anybody tried to get the sip warning header pop-up message on polycoms? |
| 16:11.08 | ManxPower-work | does "dialplan show" or maybe "core dialplan show" show you your dialplan? |
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| 16:16.17 | jsjc | ManxPower-work: yes it does show |
| 16:16.28 | jsjc | i actually can use SIP and skype no worries |
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| 16:16.34 | jsjc | it is module ssuch us dahdi not loaded |
| 16:17.00 | ManxPower-work | As I'm sure you know, we don't support Skype here. |
| 16:17.30 | jsjc | ye syes |
| 16:17.32 | jsjc | i know |
| 16:17.41 | jsjc | i dont wnat any support on it was just a comment |
| 16:17.56 | jsjc | so you know what is working to help troubleshooting |
| 16:18.22 | ManxPower-work | I must have missed you mentioning the DAHDI issue before. |
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| 16:18.49 | ManxPower-work | 'cause I've wasted my time trying to diagnose module issues |
| 16:19.05 | jsjc | isnt it?? |
| 16:19.23 | ManxPower-work | dahdi issues are dahdi issues. |
| 16:19.35 | ManxPower-work | does "service dahdi restart" work? |
| 16:19.42 | jsjc | sorry ManxPower-work I thought was module stuff because modules in module show is all 0 |
| 16:19.55 | ManxPower-work | jsjc, that means "NUMBER OF USE INSTANCES" |
| 16:19.56 | jsjc | exctep res_fax and chan_skype |
| 16:19.58 | ManxPower-work | or similar. |
| 16:20.28 | jsjc | service restart? |
| 16:20.32 | jsjc | its service a command? |
| 16:20.49 | ManxPower-work | it is a linux command. |
| 16:21.12 | ManxPower-work | it loads/unloads the required dahdi kernel hardware modules |
| 16:21.20 | jsjc | -bash: service: command not found |
| 16:21.28 | ManxPower-work | jsjc, what distro |
| 16:21.34 | jsjc | ubuntu |
| 16:21.46 | ManxPower-work | you'll have to ask someone what the equiv ubuntu command is |
| 16:22.26 | jsjc | let me check |
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| 16:30.51 | bmoraca_work | i've never liked Debian-based distros...too obscure |
| 16:31.35 | jaytee | jsjc, sudo /etc/init.d/dahdi restart |
| 16:31.50 | jaytee | much prefer RHEL or CentOS |
| 16:33.47 | jsjc | but dahdi not exist... |
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| 16:35.03 | ManxPower-work | then I guess you'd better install it |
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| 16:38.22 | ManxPower-work | jaytee, he's on Ubuntu i.e. a debian platform |
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| 16:39.24 | jaytee | ManxPower-work, I know. so he can't use service "whatever" restart, he has to use /etc/init.d/"whatever" restart and prefix it with sudo |
| 16:39.52 | ManxPower-work | jaytee, Ah. that makes sense |
| 16:40.11 | ManxPower-work | jaytee, maybe he should read The Book. |
| 16:40.13 | jaytee | which is why I prefer RHEL or CentOS, much cleaner |
| 16:40.25 | ManxPower-work | jaytee, I standardized on CentOS |
| 16:40.31 | jaytee | ManxPower-work, what? and actually have to think for himself? |
| 16:40.55 | jaytee | ManxPower-work, same here. I like RHEL but CentOS is the same code minus the cost of support contract. |
| 16:41.32 | jaytee | ManxPower-work, AFAIK if you let the service contract expire you can no longer get updates on RHEL |
| 16:43.23 | bmoraca_work | from what i understood, RHEL is actually free to use and download...it just has the added benefit of having official support AVAILABLE...and for an enterprise with significant costs of downtime, that''s generally a good thing to have |
| 16:43.53 | Qwell | bmoraca_work: nope. why do you think CentOS exists? |
| 16:44.20 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 16:45.18 | bmoraca_work | curious |
| 16:45.22 | bmoraca_work | wonder when that changed |
| 16:45.28 | Qwell | never? |
| 16:45.32 | jaytee | can't download RHEL without buying it and it has to register with RedHat when you install or run it the first time using a key |
| 16:45.44 | bmoraca_work | i seem to remember running RHEL and redhat back in the day for free |
| 16:45.57 | Qwell | RHEL != RH |
| 16:46.08 | bmoraca_work | notice the "and" |
| 16:46.17 | beefpastry | my re-entry into Linux after a 5 year hiatus began with FC6...I've never had a reason to look elsewhere. |
| 16:47.19 | bmoraca_work | either way, i've had good experiences with Centos and it seems to be stable and easy to work with |
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| 16:47.59 | beefpastry | CentOS experiences have not been as pleasant for me (seems to lag) |
| 16:48.12 | bmoraca_work | lag? |
| 16:48.28 | beefpastry | lag in updates/hw support, I mean |
| 16:48.30 | ManxPower-work | I don't want the latest and greatest stupid shiny thing on the internet. I want a stable distro |
| 16:48.48 | Qwell | of course those things are delayed. thats the point. |
| 16:48.58 | bmoraca_work | yeah...i'm OK with my box being a bit behind in favor of stability |
| 16:50.07 | jaytee | yeah, cuz running mission critical production applications on a bleeding edge OS is really cool. NOT!!!! |
| 16:50.27 | ManxPower-work | the worst seems to be Trixbox. "yum update" may or may not totally blow up your system. |
| 16:50.31 | bmoraca_work | just ask McAfee users :P |
| 16:50.42 | Qwell | ManxPower-work: that's because they're friggen idiots though |
| 16:50.50 | ManxPower-work | Qwell, Yes. |
| 16:50.53 | Qwell | and they're *still* months behind |
| 16:51.38 | ManxPower-work | I did a yum update on a trixbox server the other day. I'll be in therapy for years. |
| 16:51.46 | beefpastry | Linux hw support is light years ahead of where it used to be...amazing what has happened over the last 12 years. |
| 16:51.47 | Qwell | kmod troubles? :D |
| 16:52.03 | Qwell | ManxPower-work: if so - you're welcome. |
| 16:52.08 | atis_work | last 12 light years? |
| 16:52.20 | ManxPower-work | Qwell, system would no longer boot. kernel options. |
| 16:52.31 | Qwell | atis_work: that would be a measurement of distance |
| 16:52.36 | ManxPower-work | since it was not quite in production yet, I formatted the system and installed our own custom ISO |
| 16:52.46 | Qwell | depending on your frame of reference, I suppose. |
| 16:52.59 | atis_work | Qwell: well, i thought that was measurment of hw support :p |
| 16:53.18 | Qwell | atis_work: support is measured in km/s |
| 16:53.22 | Qwell | 800km/s |
| 16:53.59 | atis_work | so we can measure actual development speed |
| 16:55.45 | beefpastry | I always thought support was measured in m/s square |
| 16:56.07 | beefpastry | or at least avoiding that part of the equation. |
| 16:56.25 | bmoraca_work | depends where you're from. could be measured logarithmically in some places. |
| 16:56.38 | beefpastry | 9.8 to be exact |
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| 17:19.42 | Greek-Boy | lo |
| 17:19.44 | Greek-Boy | lol |
| 17:20.27 | raden_work | what are the boxes called that you can put a USB stick into and play the pics on a TV |
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| 17:23.37 | saisoma | hey guys, are there any pre-built utils within * that will do line access codes and call usage recording? i can write macro's and use mysql for storage, and will likely end up going that route, but was curious if there was anything easier to use |
| 17:24.14 | saisoma | i'm on 1.6.1.0 right this minute, but will be moving to 1.6.2.x shortly |
| 17:25.47 | spiceycurry | where could I find a good resource on configuring DID and a softphone? |
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| 17:31.38 | bmoraca_work | spiceycurry: ... |
| 17:31.40 | bmoraca_work | ~book |
| 17:31.41 | 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 |
| 17:31.55 | spiceycurry | ty |
| 17:32.00 | spiceycurry | :D |
| 17:32.04 | spiceycurry | ~book |
| 17:32.04 | 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 |
| 17:32.23 | bmoraca_work | read it twice |
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| 17:40.03 | spiceycurry | what does DID stand wor? |
| 17:40.04 | spiceycurry | *for |
| 17:40.34 | brandonf | direct inward dialing, i believe |
| 17:40.40 | spiceycurry | ah ok thanks |
| 17:41.24 | brandonf | I have a CID prefix set on an inbound route, using RGPREFIX to save the prefix when it hits a queue, and that works fine for most numbers.. but it seems like if the callerid info is read from the sip trunk, it works fine, but if the call is international.. it removes my prefix.. any ideas? |
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| 17:56.34 | dotnetted | Every time I reboot the server I have to type "sip reload" in cli or * will never register to the SIP gateway- why might this be? |
| 17:57.56 | *** join/#asterisk rare1980_ (~rare1980@12.25.228.67) |
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| 17:58.15 | *** join/#asterisk imox1234 (~imox1234@p4FC5C515.dip0.t-ipconnect.de) |
| 17:58.43 | imox1234 | hello, how can i manage prepaids account with asterisk? has anyone any idea ? |
| 17:59.16 | Naikrovek | imox1234: it's possible, people do it |
| 17:59.35 | imox1234 | very good :-) but how ;) ? |
| 17:59.56 | Naikrovek | imox1234: it's all in the dial plan. query some resource for info on account, if credits remain, allow call. if credits do not remain, play message saying so |
| 18:00.21 | bmoraca_work | imox1234: learn asterisk very well, learn relational database design very well, and just do it. |
| 18:00.34 | Naikrovek | telling you exactly how to do it will not help you. |
| 18:00.39 | Naikrovek | i agree with bmoraca_work |
| 18:01.03 | Naikrovek | start at the beginning, learn asterisk |
| 18:01.05 | ManxPower-work | you should also learn Linux, networking, and SIP as well |
| 18:01.08 | Naikrovek | it's not hard |
| 18:01.17 | Naikrovek | but there are a lot of details |
| 18:01.23 | ManxPower-work | imox1234, Hint: the L() option to Dial |
| 18:01.25 | Gido-E | and forget your social skills |
| 18:01.26 | imox1234 | ok thanks |
| 18:01.43 | Naikrovek | imox1234: when you have more specific questions, we'll be happy to help you all you need |
| 18:02.16 | Naikrovek | what you're asking is like "guys how can i drive to cincinnati, thx" |
| 18:02.45 | Naikrovek | you drive, that's how. same thing though, you gotta learn how to read road signs, you gotta learn how to drive a car and change a tire and all that |
| 18:03.05 | Naikrovek | once you know those things, driving to cincinnati will not be a mystery |
| 18:03.40 | Naikrovek | and once you learn asterisk, and linux, and sip, and rdbms, doing what you want to do will not be a mystery |
| 18:04.30 | imox1234 | Naikrovek: you speak german ? |
| 18:04.35 | Naikrovek | nein |
| 18:04.37 | Naikrovek | sorry |
| 18:04.44 | imox1234 | ok |
| 18:06.08 | imox1234 | but how it will work? i have tu use asterisk and mysql or ? |
| 18:07.48 | leifmadsen | you have to use asterisk (to handle the calls) and some other mechanism to control the calls |
| 18:08.01 | leifmadsen | that can be done in several ways, a few of which have been partially provided |
| 18:08.34 | imox1234 | but which mechanism ? :) |
| 18:08.54 | leifmadsen | ManxPower-work gave you a hint |
| 18:10.17 | *** join/#asterisk defslap (~andy@80.47.59.78) |
| 18:10.26 | rubbs | just got his 15 polycom 650's. It's going to be a good rest of the afternoon. |
| 18:10.45 | Naikrovek | nice! |
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| 18:11.19 | Naikrovek | setting up that provisioning system will be fun. if you like that kind of thing (I do) |
| 18:11.23 | leifmadsen | time to install (if not already done) your provisioning system! :) |
| 18:11.49 | leifmadsen | res_phoneprov interests me, and I played around with it for a bit. It pretty much worked except for one thing I reported. |
| 18:11.50 | beefpastry | A good afternoon of unpacking and setting up? |
| 18:12.20 | beefpastry | The provisioning is fun...not so sure about dealing with the boxes. |
| 18:12.57 | superbeef | Is it possible to tell if a polycom phone is forwarded from the asterisk CLI? |
| 18:13.23 | Get_The_Fish | superbeef, no I dont think so, but if you find a way do share! |
| 18:13.23 | Naikrovek | it is possible by the AMI, but not sure about CLI |
| 18:13.40 | Get_The_Fish | how do you do it from AMI? |
| 18:13.41 | *** join/#asterisk lanning (~lanning@208.87.235.224) |
| 18:15.37 | bmoraca_work | superbeef: forwarding is a feature of the phone, not of asterisk. the asterisk CLI isn't going to be able to tell you anything about it. |
| 18:15.39 | Naikrovek | I use an app called AST that can show me those things |
| 18:16.04 | *** join/#asterisk afink (~chatzilla@204.26.87.226) |
| 18:16.58 | iscario | hi, i just have a quesion : could you confirm that the use of IAX2 would allow me to secure voice exchanges among two clients ? |
| 18:17.38 | Get_The_Fish | uh, was that sarcasm? |
| 18:17.51 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 18:17.53 | Naikrovek | no, it does it via the AMI |
| 18:17.59 | Naikrovek | so that's how I know. |
| 18:18.13 | Get_The_Fish | huh, I can't find any details on it, care to share? |
| 18:18.33 | Get_The_Fish | I thought you were being sarcastic, AST as in ASTerisk :) |
| 18:19.29 | Naikrovek | no |
| 18:19.38 | Naikrovek | i was wrong, it's call ASTassistant |
| 18:19.49 | Naikrovek | .com |
| 18:20.13 | Naikrovek | and it only shows forwardings set up by the phones, not followme |
| 18:20.50 | Get_The_Fish | ah ok yeah I didnt know it could do that... I wonder how it works it magic ways |
| 18:21.00 | Naikrovek | no idea |
| 18:21.41 | leifmadsen | iscario: yes you can |
| 18:21.43 | superbeef | show manager commands is that showing me AMI commands or no |
| 18:22.30 | rubbs | sorry, had to deal with a problem. I haven't dealt with provisining yet... I'm still learning as I go mostly |
| 18:22.30 | [TK]D-Fender | iscario: Yes, IAx supports encryption. |
| 18:22.34 | rubbs | reading the book a lot. |
| 18:23.10 | brandonf | I set a CID prefix on my inbound route, works fine.. it seems if the number can be looked up (IE: Cell Phone VA instead of #######), the CID prefix is fine, but if the number cannot be looked up, it removes my prefix. I believe it's the CNAM data coming from my sip trunk, but anyone know of a way to keep the prefix either way? |
| 18:23.13 | iscario | ok leifmadsen , so in the end, it is possible to secure !completely! the transaction from the user agent to the server, right ? |
| 18:23.13 | Get_The_Fish | try the asterisk provisioning system first rubbs, it makes it pretty easy |
| 18:23.29 | leifmadsen | iscario: that is the point of encryption, yes |
| 18:23.29 | rubbs | ok, will do |
| 18:23.39 | leifmadsen | iscario: you will have to define "completely" |
| 18:23.47 | [TK]D-Fender | brandonf: You get what they give you. You don't lose anything. You either have it, or don't |
| 18:23.54 | *** join/#asterisk bipolar (~bipolar@offsitesysadmin.com) |
| 18:24.00 | bmoraca_work | brandonf: has to do with the fact that callerid presentation is set to off...so regardless of what you set it to, you're still telling the phone not to display anything |
| 18:24.02 | leifmadsen | iscario: regardless, if that is your concern, listening to me is an insecure method of obtaining that knowledge |
| 18:24.06 | *** join/#asterisk murdock_ut (~chatzilla@mail.kimballequipment.com) |
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| 18:24.11 | murdock_ut | Help! |
| 18:24.13 | murdock_ut | :) |
| 18:24.20 | leifmadsen | throws a life ring |
| 18:24.24 | murdock_ut | thanks! |
| 18:24.47 | brandonf | [TK]D-Fender: I don't have a problem recieving the looked up names. It removes the cid prefix if it can't receive the looked up info |
| 18:24.50 | Qwell | leifmadsen: I knew it! You created the book so you could give people invalid security advice, so you could use them all some day. |
| 18:24.58 | iscario | just wante to be sure leifmadsen : data and voice can be encrypted (is it with SIPS for the data ?) |
| 18:25.01 | leifmadsen | Qwell: shut up! |
| 18:25.04 | Qwell | :D |
| 18:25.10 | Get_The_Fish | personally I use an FTP server and some dhcp magic, I've only tested the asterisk provisioning system out of curiosity, but that's because I already have all that in place |
| 18:25.38 | bipolar | has anyone found a way to have a single phone number act as the voicemail destination for multiple cell phones? I can't find a way to determine what cell phone transfered the call. I'd rather not need to use a separate DID for each phone if possible. |
| 18:25.40 | leifmadsen | iscario: IAX2 != SIP |
| 18:25.49 | iscario | yes i know leifmadsen |
| 18:26.36 | iscario | leifmadsen : when i said completely, i talked about SIP and IAX |
| 18:26.50 | leifmadsen | I don't understand your statement |
| 18:26.50 | iscario | hu, not very clear what i am saying |
| 18:27.01 | [TK]D-Fender | brandonf: "IT"? What is "IT"? |
| 18:27.09 | leifmadsen | the encryption would be secure between IAX2 end points and Asterisk |
| 18:27.20 | murdock_ut | I'm moving from a 1.2 system to a 1.6 system. I a receptionist phone that has a sidecar that subscribes to a bunch of extensions. When on the new system the asterisk box goes nuts with tons of sip messages. I have posted them here: http://pastebin.com/0JU24RKH |
| 18:27.21 | leifmadsen | iscario: which has nothing to do with SIP |
| 18:27.52 | iscario | I do not know IAX, i ' ll have a look, and ask back if needed :D thx leifmadsen |
| 18:27.52 | murdock_ut | The phone gets so many of these that slows it down to a crawl. When I remove the subscriptions they go away. |
| 18:27.55 | Get_The_Fish | bipolar- yes, it's easy- route the DID that you want to use for voicemails into the Voicemail dialplan application without passing username or password variables. |
| 18:28.17 | leifmadsen | iscario: basically IAX2 is a separate protocol from SIP. It has encryption capabilities. SIP does not (currently) have such capabilities in Asterisk. |
| 18:28.22 | Get_The_Fish | bipolar- the only issue is that the caller will have to know the voicemail mailbox, which could be an issue. |
| 18:28.35 | leifmadsen | iscario: if you purpose is to do encryption with SIP end points, you will not be able to. |
| 18:28.43 | iscario | oh... that was my point in fact leifmadsen |
| 18:28.50 | leifmadsen | hence the IAX2 != SIP |
| 18:29.01 | iscario | yeah yeah i got that |
| 18:29.09 | leifmadsen | you can not do SIP encryption with Asterisk at this time |
| 18:29.28 | leifmadsen | M5413 |
| 18:29.40 | iscario | si it means that password are in clear on the network |
| 18:29.40 | MuffinMan | [assigned] [Asterisk] Channels/chan_sip/NewFeature 0005413: [patch] [branch] Secure RTP (SRTP) reported by mikma https://issues.asterisk.org/view.php?id=5413 |
| 18:30.05 | brandonf | [TK]D-Fender: I believe macro-user-callerid is the function in extensions_additional removing it |
| 18:30.07 | leifmadsen | iscario: no..... passwords are encrypted of course -- media is not |
| 18:30.19 | [TK]D-Fender | brandonf: FreePBX is NOT supported here. |
| 18:30.35 | [TK]D-Fender | brandonf: Stay in their channel with this |
| 18:30.49 | brandonf | aaahhh.. gotcha, freepbx macro then, thanks :) |
| 18:31.25 | bipolar | Get_The_Fish: yeah, that won't work for what I need. I basicaly want to replace the cell phone voicemail with asterisk. I can do it if I dedicate a DID to a users voicemail. |
| 18:32.13 | iscario | ok that's a point then leifmadsen. It means that i can read the data of the SIP except passwords is using tls (i'm juste talking abt sip here^^) |
| 18:32.26 | Get_The_Fish | bipolar- I'm not sure that I understand... if they call the cell phone, they will get the cell phone's voicemail, as it never hits the * platform. |
| 18:32.28 | iscario | if i'm listening the network |
| 18:32.38 | Get_The_Fish | bipolar- am I missing something? |
| 18:32.41 | leifmadsen | iscario: I think 1.6.2 (or Asterisk trunk) has some TLS support |
| 18:32.44 | bipolar | Get_The_Fish: GSM allows you to set the number voicemail forwards to. |
| 18:32.53 | bipolar | Get_The_Fish: http://www.arcx.com/sites/gsmfeatures.htm |
| 18:33.03 | Get_The_Fish | ahhhhh ok I see what you are doing... |
| 18:33.14 | iscario | ok i'll read some documentation, thx for your help |
| 18:33.18 | iscario | leifmadsen |
| 18:33.25 | bipolar | Get_The_Fish: yeah. it works great as long as you have a DID that goes directly to the users voicemail. |
| 18:34.35 | Get_The_Fish | bipolar- well, not necessarily- whats the ANI that asterisk receives when the call is forwarded? |
| 18:34.36 | bipolar | Get_The_Fish: I don't think there is a way to tell what phone the call was forwarded from. |
| 18:35.25 | Get_The_Fish | bipolar- if the ANI is unique, then you can do some dialplan logic that routes the call to the correct mailbox based on the ANI of the call |
| 18:35.31 | bipolar | Get_The_Fish: hmm... good question. |
| 18:35.55 | bipolar | Get_The_Fish: I know callerid shows the original caller.... let me check the logs. |
| 18:36.11 | *** join/#asterisk moy (~chatzilla@bas1-unionville55-1177733627.dsl.bell.ca) |
| 18:36.15 | Get_The_Fish | so, I would think that the ANI would be the cell phone number |
| 18:38.35 | bipolar | Get_The_Fish: hmm... the full asterisk log doesn't show any ANI number, just caller ID. |
| 18:39.11 | Get_The_Fish | add a verbose line to the dialplan to display it, then call that dialplan extension... |
| 18:39.50 | *** join/#asterisk moy (~chatzilla@bas1-unionville55-1177733627.dsl.bell.ca) |
| 18:40.11 | raden_work | if a company is reporting our company name wrong on a customers caller ID how can i correct that |
| 18:40.15 | Get_The_Fish | bipolar: use ${CALLERID(ANI)} |
| 18:40.29 | *** join/#asterisk jart (~jart@c-76-23-206-246.hsd1.ct.comcast.net) |
| 18:40.42 | Get_The_Fish | raden_work: contact your carrier |
| 18:41.02 | raden_work | vitelity |
| 18:41.07 | raden_work | but they dont control that do they ? |
| 18:41.19 | Get_The_Fish | raden_work: AFAIK yes |
| 18:46.05 | *** join/#asterisk ManxPower-work (~manxpower@216.186.151.147) |
| 18:51.03 | bipolar | Get_The_Fish: it's still showing the original number. I'm guessing I'm going to need a DID for each phone. Oh well. Thanks for your help! |
| 18:51.07 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 18:54.22 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 18:54.44 | *** join/#asterisk rare1980_ (~rare1980@12.25.228.67) |
| 18:55.45 | *** join/#asterisk staffmember (~singulari@static-71-183-79-24.nycmny.fios.verizon.net) |
| 19:01.42 | *** join/#asterisk iratik (~itariki@74.223.41.171.nw.nuvox.net) |
| 19:03.06 | iratik | I need some help. I've got an error message in cli "WARNING[2534]: channel.c:780 channel_find_locked: Avoided deadlock for '0x__x0' 9 retries. This is running on CentOS 5.5, on amd/opteron asterisk version 1.2.30.2 |
| 19:03.47 | iratik | Its flooding my console and causing asterisk to bring the system load too high due to the number of channels generating this message. I need to know how to either fix the root cause or supress the message... thoughts? |
| 19:05.39 | *** join/#asterisk UserReg_CL (~COB@pc-167-235-86-200.cm.vtr.net) |
| 19:06.02 | UserReg_CL | hi, know one softphone web free ? |
| 19:07.00 | pabelanger | iratik: 1.2 is pretty old. |
| 19:07.47 | Get_The_Fish | bipolar- well, the advantage of using a DID for each phone is that you could take advantage of followme and give your cell users the privacy options it offers... |
| 19:15.10 | iratik | You guys were quite helpful |
| 19:15.20 | leifmadsen | UserReg_CL: I know of no free web softphones |
| 19:15.45 | leifmadsen | UserReg_CL: any of them are likely to be built on ActiveX |
| 19:15.59 | leifmadsen | pabelanger: s/pretty/extremely/ :) |
| 19:17.09 | ManxPower-work | iratik, deadlocks on 1.2 are commonly triggered by chanspy or monitor/mixmonitor. they are bugs/design issues. upgrade |
| 19:17.33 | iratik | MacxPower-work: I love you |
| 19:17.40 | hardwire | we all kinda love him |
| 19:17.44 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 19:17.50 | iratik | No amount of googling was helping me |
| 19:18.33 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 19:18.37 | *** join/#asterisk murdock_ut (~chatzilla@mail.kimballequipment.com) |
| 19:19.00 | iratik | The framework I'm using has issues with 1.4 What is the highest version of 1.2 i can upgrade to? |
| 19:19.28 | ManxPower-work | iratik, no version of 1.2 will do much to solve your problem |
| 19:19.42 | Get_The_Fish | iratik, are you using vicidial? |
| 19:19.47 | iratik | yup |
| 19:20.06 | Get_The_Fish | yeah thats what I thought. Sorry, but there is little that you can do. |
| 19:20.25 | Get_The_Fish | vicidial CAN work on 1.4, but I'm not sure how well |
| 19:20.59 | iratik | I see so many things in the forums about 1.4 and issues |
| 19:23.09 | Get_The_Fish | well, honestly, 1.2 goes end of life later this year, and is security fixes only since '07 |
| 19:23.31 | Get_The_Fish | so any issues that you have, you have, unless you want to pay someone to fix them, which still might break vicidial |
| 19:24.35 | iratik | Is there a way to treat the symptoms? |
| 19:24.42 | iratik | Supress the error messages from the channel.c ? |
| 19:24.53 | iratik | the io from the error messages is what's killing the system |
| 19:25.29 | Get_The_Fish | you can change logger.conf to not show those on the cli, if 1.2 lets you do that (I dont remember), but you will lose out on all messages of that class when you do |
| 19:26.09 | leifmadsen | iratik: disable WARNING messages in logger.conf per Get_The_Fish |
| 19:26.48 | leifmadsen | ways of treating the symptoms is either through testing and determining where something lies, and then disabling it, or looking at the C code and making changes. |
| 19:26.59 | leifmadsen | beyond that, your question is a much too vague to offer any definitive answer |
| 19:27.22 | iratik | I can figure out how to modify channel.c to not present those messages. However, I wouldn't know how to compile in those changes |
| 19:27.37 | Get_The_Fish | however you will still be writing those to file, which is probably more IO intensive. And I wouldnt recommending disabling logging of the "WARNING" class completely |
| 19:27.51 | Get_The_Fish | bounty it |
| 19:32.02 | leifmadsen | iratik: logger.conf --> console => notice,error (notice no warning listed) |
| 19:32.06 | leifmadsen | iratik: *CLI> reload |
| 19:32.41 | leifmadsen | iratik: otherwise, find the offending ast_warning() function, comment it out or delete it, then just run 'make install' again |
| 19:32.44 | iratik | I did logger.conf --> console => error, notice ;messages => error, notice |
| 19:33.05 | iratik | what does that error messages even mean? |
| 19:33.25 | *** join/#asterisk ChrisWi (~admin@mx2.wwserver.net) |
| 19:33.42 | iratik | I killed the log messages.. but system load is still shooting up |
| 19:33.45 | leifmadsen | it means asterisk avoided a deadlock |
| 19:33.50 | iratik | from 18.. to 26.. to 38 .. to 53 |
| 19:34.02 | leifmadsen | then something else is causing the issue and it's not the warning message that is causing it |
| 19:34.22 | iratik | Garr |
| 19:35.39 | iratik | http://pastie.org/939745 ? |
| 19:36.03 | leifmadsen | yes, I'm sure we believe you're not lying about it ;) |
| 19:37.46 | ctooley | iratik, how many calls? |
| 19:44.14 | iratik | On about 8 agents |
| 19:44.31 | iratik | Number of agents in call at that time was 7 |
| 19:49.45 | iratik | lol. uptime at 79.80 42.95 and 25.91 |
| 19:50.54 | *** join/#asterisk vik_ (~b49531e9@gateway/web/freenode/x-iocqlunzrquujbiz) |
| 19:51.48 | Get_The_Fish | iratik: according to this you can install vicidial on 1.4: http://astguiclient.sourceforge.net/scratch_install.html |
| 19:52.55 | p3nguin | iratik: That looks more like a load average than an uptime. :/ |
| 19:52.56 | iratik | The error messages apparently were not the issue |
| 19:53.15 | pabelanger | Get_The_Fish: I tried in on 1.6.0 and was able to do some outbound dials. |
| 19:53.33 | p3nguin | Speaking of uptime, I'm not very pleased that my uptime rolled over to 0. |
| 19:54.25 | Get_The_Fish | pabelanger: are you referring to the patch for issue 0017254? |
| 19:54.49 | *** join/#asterisk Yudaisrael1984 (~Yuda@80.179.204.164.cable.012.net.il) |
| 19:54.57 | pabelanger | Get_The_Fish: your previous vicidial comment. |
| 19:55.34 | Yudaisrael1984 | hi ppl anyone here know a2billing at all or can direct me to a2billing channel? |
| 19:55.41 | Get_The_Fish | oh, ok... I am sbrady_gtfservices btw, sorry for the name confusion, we've been going back and forth on the bugtracker about that issue |
| 19:56.09 | Get_The_Fish | yeah, a buddy of mine tried it with 1.6.0.x and had some issues with latency on the AMI connection with the perl scripts. |
| 19:56.25 | Get_The_Fish | I cant speak too much about it myself, haven't tried it out that much |
| 19:56.30 | beek | infobot, show me thebook |
| 19:56.41 | beek | infobot, show me book |
| 19:56.42 | *** join/#asterisk spiceycurry (~mcurry@proxy.hostopia.com) |
| 19:56.48 | Get_The_Fish | ~book |
| 19:56.49 | 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 |
| 19:57.24 | beek | Get_The_Fish: Tnx. I was looking to get that sent to me in a PM. |
| 19:57.32 | Get_The_Fish | lol |
| 19:58.17 | pabelanger | Get_The_Fish: cool the patch in 17254 fixed your issue? |
| 19:58.24 | Get_The_Fish | absolutely |
| 19:58.48 | pabelanger | Get_The_Fish: nice! Be sure to thank Marquis42. He was the one who made the connection to a previous issue. |
| 19:59.50 | Get_The_Fish | I am thrilled about it, absolutely thrilled... thanks so very much for your help, and from Marquis42, as well as Leif a big thanks!! |
| 20:00.15 | Get_The_Fish | should I update the linked issue as well with my testing? |
| 20:02.27 | pabelanger | Get_The_Fish: should be fine at this point. We'll just update your issue to confirming the fix. Then we can start work on getting sigh off and a somebody to commit it. |
| 20:02.39 | Get_The_Fish | super sweet |
| 20:07.30 | *** join/#asterisk jhatch (~jhatch@208.53.57.55) |
| 20:10.12 | *** join/#asterisk ReDNeQ (~ReDNeQ@70.114.229.58) |
| 20:14.13 | murdock_ut | So I could use some pointers for my issue.... :) |
| 20:15.09 | *** join/#asterisk diegomad (~mad@190.146.200.120) |
| 20:16.08 | murdock_ut | I have a phone that subscribes to a bunch of extensions. When I do this I get flooded with sip messages. You can see them here: http://pastebin.com/hvXztHBd |
| 20:16.08 | Get_The_Fish | murdock_ut: whats that issue again? |
| 20:16.19 | murdock_ut | It is a big pastbin. |
| 20:17.23 | murdock_ut | I was on 1.2 not I'm on 1.6.1.18 but I also saw the problem on 1.6.2.6 |
| 20:17.38 | murdock_ut | opps. ..now I'm on.... |
| 20:18.01 | Get_The_Fish | are you using realtime? |
| 20:18.10 | paulc | Anyone had any experience/expertise with large scale conferencing? Like a handful of people talking, and hundreds of people in a listen-only mode? |
| 20:19.01 | Get_The_Fish | yes. I been in one of those before. :) |
| 20:19.13 | *** join/#asterisk Alagar (~Administr@122.164.32.153) |
| 20:21.02 | murdock_ut | Get_The_Fish: No. |
| 20:21.15 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 20:22.23 | Get_The_Fish | murdock_ut: I would think that it is expected behavior, honestly... I dont see the issue here |
| 20:22.38 | ManxPower-work | murdock_ut, did you read UPGRADE*.txt? |
| 20:23.38 | *** join/#asterisk Torrieri (~Torrieri@nelug/crew/torrieri) |
| 20:24.21 | murdock_ut | I had copied most of my dial plan from a working 1.6 install. |
| 20:24.33 | p3nguin | paulc: People in #vuc are familiar with that. You could ask there. |
| 20:24.58 | paulc | p3nguin: cheers (and what's #vuc?) |
| 20:25.22 | paulc | Ah.. http://www.voipusersconference.org/ ? |
| 20:25.51 | p3nguin | right |
| 20:26.16 | murdock_ut | The issue is that I'm getting a constant flow of those messages and it is bogging the phone down. |
| 20:26.33 | p3nguin | paulc: as in: /join #vuc |
| 20:27.05 | paulc | is that how you join channels? :) |
| 20:27.07 | paulc | cheers - I'm there now |
| 20:28.11 | p3nguin | oh |
| 20:28.17 | p3nguin | I didn't know you were already joined. |
| 20:29.14 | paulc | s'all good :) Let's see what the collective brain comes up with.. |
| 20:40.14 | *** join/#asterisk Jumpie (~lah@c-76-111-80-110.hsd1.va.comcast.net) |
| 20:40.54 | Yudaisrael1984 | p3nguin wassup how u doing? |
| 20:42.08 | ManxPower-work | murdock_ut, why don't you just turn off sip debug |
| 20:42.24 | murdock_ut | The phone is bogged down and is unusable. |
| 20:42.53 | Yudaisrael1984 | murdock where u from? |
| 20:43.10 | murdock_ut | Utah. |
| 20:43.25 | ManxPower-work | murdock_ut, try commenting out the mailbox=whatever lines in sip.conf. If that fixes it then you know where to look for a perm solution |
| 20:43.36 | ManxPower-work | don't forget your sip reload |
| 20:43.45 | Yudaisrael1984 | p3nguin u read pm? |
| 20:44.04 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 20:44.49 | murdock_ut | ManxPower-work: If that fixes it then what does that tell me? |
| 20:46.03 | murdock_ut | ManxPower-work: And are you suggesting I do that on all sip peers or just 101 which is the phone with the side car? |
| 20:46.17 | *** part/#asterisk Get_The_Fish (~Get_The_F@c-24-8-50-199.hsd1.co.comcast.net) |
| 20:47.43 | ManxPower-work | murdock_ut, whichever ones have a problem |
| 20:48.17 | murdock_ut | btw this is a sample of my sip.conf |
| 20:48.19 | murdock_ut | http://pastebin.com/rSse4fg4 |
| 20:49.12 | ManxPower-work | it would point to the phone having a bug or config problem with regards to subscribe |
| 20:49.18 | murdock_ut | I'm also wondering if it may be an issue with the phone. Found this: http://forum.snom.com/index.php?showtopic=3882&view=&hl=subscriptions&fromsearch=1 |
| 20:50.10 | murdock_ut | does the sip.conf look about right? No glaring issues? |
| 20:50.27 | ManxPower-work | murdock_ut, Which phone? |
| 20:50.35 | *** join/#asterisk Joel (~jschuweil@unaffiliated/joel) |
| 20:50.40 | murdock_ut | What phone am I using? Snom 37- |
| 20:50.42 | murdock_ut | 370 |
| 20:51.57 | murdock_ut | ManxPower-work: One other thing. Shouldn't the type=friend on a phone since it makes and receives calls? |
| 20:52.14 | ManxPower-work | phones are almost ALWAYS friends |
| 20:52.49 | murdock_ut | ManxPower-work: Disregard that part on my sip.conf. I forgot to change it back to friend last night while trouble shooting. |
| 20:56.01 | *** join/#asterisk waa (~waa@balrog.credipar.com.br) |
| 20:57.03 | *** join/#asterisk lvlolvlo (~lvlolvlo@unaffiliated/lvlolvlo) |
| 20:58.25 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 20:59.38 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:00.47 | Joel | GameGamer43, are you in huntsville this week? |
| 21:05.27 | p3nguin | murdock_ut: peers make and receive calls, too. |
| 21:05.40 | p3nguin | All my phones are peers (now). |
| 21:05.50 | murdock_ut | p3nguin: So confusing. |
| 21:05.58 | p3nguin | yudaisrael1984: What? No, I don't accept unsolicited PMs. |
| 21:06.35 | Joel | murdock_ut, http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer |
| 21:07.22 | murdock_ut | p3nguin: Yea, I saw that. |
| 21:07.31 | murdock_ut | oh sorry joel |
| 21:08.02 | ManxPower-work | peers became much more friend'ly in 1.4 and even more so in 1.6 |
| 21:08.14 | Yudaisrael1984 | so i had a system crash 24 hours ago |
| 21:08.16 | Yudaisrael1984 | segment fault |
| 21:08.19 | Yudaisrael1984 | for no reason |
| 21:08.25 | murdock_ut | So is the growing consensus that we should use peers from now on for phones? |
| 21:08.33 | Yudaisrael1984 | no update was made on the server and no special script was running |
| 21:08.41 | Yudaisrael1984 | im trying to get info on how do i debug it |
| 21:08.57 | Yudaisrael1984 | i downloaded gdb but im trying to figure out how to use it |
| 21:09.07 | p3nguin | murdock_ut: It depends if you want IP/port authentication or username authentication when making calls. |
| 21:10.15 | murdock_ut | p3nguin: That is for making calls, but not for the initial phone registration right? |
| 21:10.47 | *** join/#asterisk sun28 (~light@sun28.ipfw.su) |
| 21:11.32 | lvlolvlo | I like the method of using peers now |
| 21:11.58 | lvlolvlo | It makes more sense in my head, in terms of flexibility |
| 21:11.59 | lvlolvlo | :) |
| 21:12.03 | p3nguin | murdock_ut: registration is for a device(phone) to receive calls. |
| 21:12.34 | lvlolvlo | just my .02 |
| 21:12.35 | *** join/#asterisk blaines (~blaines@67.130.168.2) |
| 21:12.50 | *** join/#asterisk evil_gordita (~evilgordi@ip70-188-50-186.rn.hr.cox.net) |
| 21:12.50 | murdock_ut | p3nguin: Right so that the pbx knows where the phone is. |
| 21:13.12 | p3nguin | What does that have to do with making a call and authing by username or IP/port? |
| 21:16.28 | murdock_ut | Sorry, this is what confused me: "'type=peer' is _never_ matched on username for incoming calls, only matched on IP address/port number" would that mean that I would need to have the ip for each phone in my sip.conf? |
| 21:16.30 | *** join/#asterisk slinksh0t (~slinksh0t@rrcs-24-39-203-130.nys.biz.rr.com) |
| 21:16.41 | p3nguin | murdock_ut: incoming... to asterisk |
| 21:16.49 | p3nguin | That's a phone making a call into asterisk. |
| 21:17.19 | p3nguin | A phone who has type=user cannot receive calls at all. |
| 21:17.59 | murdock_ut | Type user is used to make out bound call to say an ITSP. |
| 21:18.02 | *** join/#asterisk mykhyggz (~col@evolone.org) |
| 21:18.17 | murdock_ut | pbx => itsp |
| 21:18.20 | p3nguin | type=user allows calls into asterisk only |
| 21:18.22 | p3nguin | nothing else |
| 21:18.41 | p3nguin | What asterisk does with the call is, at this point, not relevant. |
| 21:19.01 | murdock_ut | k. I was just worried about the authentication part. |
| 21:19.07 | *** join/#asterisk Brookss (~SSJGotenk@d142-59-61-33.abhsia.telus.net) |
| 21:19.09 | p3nguin | So if the phone is type=user and the ITSP is type=peer, the phone can make a phone call going out via ITSP. |
| 21:19.24 | murdock_ut | k. |
| 21:19.37 | p3nguin | It's simple. Do you want to auth by username? If yes, use type=friend for the phone. |
| 21:20.11 | murdock_ut | p3nguin: That is what I thought, and that is why I had them set as friend. |
| 21:20.33 | p3nguin | Was there a problem that you were second guessing your configurations? |
| 21:21.23 | murdock_ut | I read that page on voip-info that Joel mentioned and didn't know if things had changed. |
| 21:21.29 | *** join/#asterisk outtolunc (~me@c-67-160-192-210.hsd1.ca.comcast.net) |
| 21:21.59 | murdock_ut | Just trying to keep my configs current with best practices. |
| 21:22.30 | Joel | peer is what you should be using. |
| 21:22.52 | p3nguin | unless you insist on authing with a username. |
| 21:23.43 | *** join/#asterisk citywok (~chatzilla@67-134-194-33.dia.static.qwest.net) |
| 21:23.52 | spiceycurry | downloaded x-lite,and setup a sip [1000] just link that that ~book, and getting registration error: 408 - request time out. Do I need a DNS server on my asterisk box? |
| 21:24.06 | murdock_ut | OK, so not to sound dense, but who is authenticating with who when you say that? I would imagine phone to pbx. In which case wouldn't I need to include the phones ip in the sip.conf? |
| 21:24.26 | p3nguin | phone to asterisk, yes... |
| 21:24.49 | p3nguin | but if your phones are registering dynamically, you can't include the IP address. |
| 21:25.42 | spiceycurry | I made the host=dynamic\ |
| 21:25.45 | *** join/#asterisk ReDNeQ (~ReDNeQ@70.114.229.58) |
| 21:25.46 | spiceycurry | host=dynamic |
| 21:26.21 | p3nguin | That will allow the phones to send REGISTERs to asterisk and be accepted (provided that all other configuration is correct and valid). |
| 21:26.25 | spiceycurry | so for "domain" in x-lite, do I need to still specifiy the ip of my asterisk server as the domain? |
| 21:26.40 | murdock_ut | And the reason for doing authentication based on ip would be to prevent someone from outside your network registering to you pbx. If I was using a username/password combo that could possibly be guessed. |
| 21:27.42 | spiceycurry | ok it worked. for reasons unknown to myself... I had Register with domain and receive incoming calls. |
| 21:27.49 | spiceycurry | unclicked it works |
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| 21:56.43 | jhatch | Is there a way with Asterisk to register 2 devices with the same credentials and have both devices ring when the extension is called? Right now, even though they both can dial out, only the last one to register can receive incoming calls. |
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| 21:57.53 | lvlolvlo | @jhatch ring group or follow me comes to mind :/ |
| 21:57.57 | Joel | jhatch, you can accomplish this with a call queue. |
| 21:58.14 | Joel | but for the exact specifics you listed, no. |
| 21:58.20 | jhatch | Doesn't that require separate extensions? |
| 21:58.48 | megalomano | hi people , question , its possible , send SMS messages with asterisk? |
| 21:58.58 | Joel | megalomano, yes. |
| 21:59.09 | lvlolvlo | yes, that would - but that's all I can think of atm to accomplish this. perhaps another user might have a better option |
| 21:59.24 | jhatch | I know some VOIP providers allow multiple devices to register with the same credentials and they all ring. How are they doing it? |
| 21:59.57 | Joel | jhatch, ask the voip provider? |
| 22:00.07 | jhatch | :) |
| 22:00.12 | Joel | jhatch, are you certain they use asterisk to handle registrations? |
| 22:00.14 | sawgood | In a nutshell what is the difference between 'dial pattern' for outside route and dial rules for a SIP trunk? |
| 22:00.24 | jhatch | Not certain they use asterisk. |
| 22:00.33 | Joel | sawgood, ? freepbx terms your speaking in, or? |
| 22:00.35 | sawgood | can they both have the exact same entries without conflict? |
| 22:00.43 | jhatch | *my* voip provider does use asterisk and I cannot register 2 devices. |
| 22:01.04 | jhatch | The one in question, not sure what they use., |
| 22:02.16 | Joel | sawgood, what you're asking makes no sense from an asterisk perspective. |
| 22:02.31 | megalomano | Joel: thanks , is necessary some module ? |
| 22:02.44 | sawgood | right .. Joel ... I think i got it figured out |
| 22:03.08 | Joel | sawgood, that's a very freepbx specific question |
| 22:03.25 | Joel | sawgood, never huts to ask in #freepbx |
| 22:03.28 | Joel | hurts* |
| 22:04.15 | Joel | megalomano, depends on how you want to do sms, this may get the ball rolling for you http://www.voip-info.org/wiki/view/Asterisk+cmd+Sms |
| 22:06.22 | megalomano | Joel: thanks again , i have gsm equipment to make it terminator |
| 22:08.15 | *** join/#asterisk beardy (~beardy@unaffiliated/beardy) |
| 22:14.05 | jhatch | Looks like the voip provider that allows multiple registrations is using Kamailio, not Asterisk. |
| 22:14.41 | jhatch | Thanks for your answers |
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| 22:15.47 | p3nguin | Asterisk doesn't handle multiple registrations from the same peer like you would hope. People that configure networks with this capability are often using things like OpenSEP/OpenSIPS in front of Asterisk. |
| 22:16.05 | p3nguin | OpenSER, I meant |
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| 22:17.06 | sawgood | I'm sure I have my dial rules and dial plan setup correctly, If I make a call using a softphone (external to the Asterisk box things work great) |
| 22:17.16 | sawgood | If I call from a Snom 320 on the LAN, I get a fast busy |
| 22:17.28 | sawgood | I guess maybe the Snom 320 phones have some keymap or something |
| 22:17.33 | p3nguin | Where's the sip debug to show this? |
| 22:17.34 | sawgood | anyone know by chance? |
| 22:17.34 | Joel | sawgood, they do |
| 22:17.40 | sawgood | digit map I think |
| 22:17.44 | Joel | sawgood, do you see the call hitting asterisk, or not? |
| 22:17.51 | p3nguin | Where's the sip debug? |
| 22:17.53 | sawgood | yes/no ... |
| 22:18.06 | Joel | yes/no isn't a valid answer to my question. |
| 22:18.07 | sawgood | I see the call does something in Asterisk but it is not as inteneded |
| 22:18.24 | Joel | if the call is hitting asterisk, then it's not a phone digit map issue. |
| 22:18.40 | sawgood | If someone dials 7 digits from the Snom phone, I see Asterisk do 'other' things like 'change extension state', etc |
| 22:19.07 | p3nguin | Where's the sip debug? |
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| 22:19.25 | Joel | sip debug isn't the first thing I would ask for in trouble shooting this *shrug* |
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| 22:20.07 | Joel | I'd be far more interested in the cli with verbose enabled then anything else |
| 22:20.16 | sawgood | I do not clearly see anyting labeled as 'digit map' in the Snom GUI |
| 22:20.34 | Joel | sawgood, did you not see what I said? |
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| 22:21.11 | Joel | additionally, if you chose to ignore my last comment, at the very least try googling "snom digit map" |
| 22:21.19 | sawgood | Joel: I have the Asterisk CLI open (this is a VERY busy Asterisk box) ... I have a user on his cell phone ... I ask him to call out from his Snom |
| 22:21.46 | sawgood | I hear the speakphone ... when he dials 7 digits and hits the CHECK key, I see output reaching the Asterisk box |
| 22:22.10 | sawgood | its not a 'normal' cal coming to the Asterisk box ... its like the Asterisk box is trying to do something else with the digits recevied |
| 22:22.33 | p3nguin | If only there were some way to debug such problems. |
| 22:22.43 | Joel | sawgood, I'm not terribly interested in repeating myself. |
| 22:22.54 | p3nguin | Fuck, that's what the devs need to work on... some type of debug commands or something. |
| 22:23.07 | Joel | p3nguin, no need to be so harsh. |
| 22:23.10 | sawgood | I have verbose set to 12 |
| 22:23.12 | Joel | p3nguin, if he doesn't want help, so be it. |
| 22:23.24 | p3nguin | Yep, that's why I stopped asking. |
| 22:23.35 | sawgood | I'm trying to work with like 4 people at onces ... two on the phone, relaying to another, and chatting at the same time |
| 22:23.43 | sawgood | its not as easy as it might seem |
| 22:23.51 | Joel | p3nguin, additionally a sip debug is going to tell you NOTHING about call flow through asterisk, just fyi. |
| 22:24.29 | Joel | sawgood, well, if you can't understand what I'm saying, and provide the different types of output several of us are asking for, I believe freepbx has a commercial support option. |
| 22:24.51 | Joel | p3nguin, I'll bet a small child it's a dial plan issue in freepbx. |
| 22:25.43 | sawgood | Ok .. here is the deal ... |
| 22:26.01 | Joel | p3nguin, but I don't have the patience for teeth pulling or the attention span to pay attention much longer :) |
| 22:26.02 | sawgood | speakerphone on Snom 320 ... client dials 7 digits and hits the check on the phone |
| 22:26.17 | sawgood | The call does not do anything at the CLI ... the Snom plays a fast busy |
| 22:26.59 | sawgood | So, its like the 7 digits do not leave the phone and make it to the Asterisk box |
| 22:27.11 | [TK]D-Fender | sawgood: Stop. If Verbose shows nothing at CLI, then look at the SIP DEBUG. |
| 22:27.18 | p3nguin | So the whole "call is reaching asterisk" bit wasn't right. Good. |
| 22:27.29 | murdock_ut | sawgood: Does the status sa OK on the system information page on the snom? |
| 22:28.05 | sawgood | yes, the Snom 320 phone works (in fact, you can dial out 'sometimes') its only if certain combinations of numbers are dialed will it not work |
| 22:28.21 | sawgood | 7 digit dial never works ... and only some 11 digit dialing works |
| 22:28.57 | murdock_ut | Have you looked at sip debug? |
| 22:29.05 | Joel | somebody sounds super confused. |
| 22:29.12 | *** join/#asterisk joobie (~joobie@mx01.anric.com.au) |
| 22:29.35 | carrar | heh |
| 22:29.41 | sawgood | sip debug from the Snom or from the CLI? |
| 22:29.48 | murdock_ut | cli |
| 22:30.05 | lvlolvlo | I would defn look at either the SIP debug or grab a tcpdump from the system to look at the call legs (e.g. from ext-->pbx and pbx--->sip provider) |
| 22:30.12 | Joel | p3nguin, I take my bet back based on the ever changing description of the problem :) |
| 22:30.25 | joobie | guys ive got a voip phone with 2 lines (polycom 320).. line 1 is setup to register with my sip box, but line 2 i cant use.. what's the proper way to set this up? Do i need to create a second extension for line 2 or can i just extend line 1 so that it gives the option to use line 2 also?? |
| 22:31.15 | sawgood | sip debug from the CLI is not the correct syntax |
| 22:31.20 | paulc | joobie: do you want 2 buttons to represent the same extension? |
| 22:31.27 | [TK]D-Fender | joobie: linekeys / callsperlinekey |
| 22:31.41 | paulc | joobie: or two different "extensions" (each that can take multiple calls stil) but with different numbers/usages/etc |
| 22:32.34 | carrar | joobie, lok at reg.1.lineKeys in the Polycom Admin Guide |
| 22:33.56 | vik_ | which python module is required to call answer function in python AGI ? |
| 22:34.25 | sawgood | You would 'think' digit map would be something 'easy' to find in the Snom 320 web GUI |
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| 22:35.19 | joobie | sorry phone, back |
| 22:35.37 | joobie | paulc, ideally.. i sorta think one extension per phone would be best - unless there's reason not do to this |
| 22:36.09 | joobie | cheers paulc TK, carrar |
| 22:37.19 | paulc | joobie: It's sometimes useful to have multiple extensions per phone. Example: My home phone. Polycom 601. Line 1 = my home number. Line 2 = door entry system. Line 3 = extension that I send a bunch of toll free and other numbers to, prefixing the caller ID name to let me know which number I'm answering |
| 22:41.57 | murdock_ut | sawgood: did you find your digitmap? |
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| 22:53.02 | vik_ | which python module is required to call answer function in python AGI ? |
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| 23:01.15 | Joel | paulc / joobie a more simple case is home v. work extensions |
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| 23:10.53 | joobie | this is just a work phone |
| 23:11.06 | joobie | just need the ability to say accept 2 calls at once.. ideally if it could be same extension it woul dbe better |
| 23:11.11 | joobie | will look into the linekeys |
| 23:13.56 | Joel | joobie, http://<phone ip>/reg_1.htm -> Polycom/456 -> "Num Linke Keys" : 2 -> "Submit" |
| 23:14.07 | Joel | ignoring the typo in there. |
| 23:15.03 | joobie | Thanks Joel |
| 23:15.06 | joobie | exactly what i needed :P |
| 23:15.16 | joobie | and in the sip.conf, do i just need to change call-limit to 2? |
| 23:15.39 | [TK]D-Fender | joobie: You shouldn't put limits in * effectively |
| 23:15.53 | [TK]D-Fender | joobie: 99 is a good idea. Only so presence works |
| 23:16.17 | joobie | what do you mean by only so presence works? |
| 23:16.27 | joobie | i put the limit in because we have one client ont eh system that pays per line |
| 23:16.44 | joobie | so didnt want them grabbing too many lines at one time.. ended up carrying this config across to all the other sip users |
| 23:17.05 | Joel | I don't set call limits in sip.conf *shrug* |
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| 23:29.51 | sawgood | got ... it ... darn Snom 'dial-plan' settings preventing 7 digit dialing from workign |
| 23:29.56 | sawgood | all working now ... whew! |
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| 23:49.19 | TJNII | ISPs should offer country blocking as a feature. I'd pay more to not get traffic from China and Russia. |
| 23:49.43 | lvlolvlo | @TJNII why not just that in your asterisk server? |
| 23:50.34 | TJNII | I whitelist now, but it would be nice to just have an umbrella ban across all my IPs. |
| 23:51.12 | TJNII | Blocking a country isn't as easy as it sounds, as they seldom have one contiguous IP range. Hence paying someone else. |
| 23:51.20 | lvlolvlo | are you talking about VoIP traffic or just regular internet traffic? |
| 23:51.32 | TJNII | Everything. |
| 23:52.02 | TJNII | Most of the VoIP attacks I saw came from Russia. China really wants to get into the one box I have with ssh on port 22. |
| 23:53.32 | carrar | TJNII, block a country is pretty eacy |
| 23:53.33 | carrar | easy |
| 23:54.26 | carrar | say for example china |
| 23:54.36 | carrar | download ftp://ftp.apnic.net/apnic/dbase/data/country-asn.lst |
| 23:54.53 | carrar | grep ": cn :" country-ipv4.lst | awk -F: '{print $2}' | sed "s/ //g" | sort -n |
| 23:55.32 | carrar | add that to your filter table |
| 23:55.44 | carrar | scrip it to update daily |
| 23:56.34 | TJNII | That's still a pain. My point is that it would be a nice feature from an ISP. |
| 23:56.40 | carrar | true |
| 23:56.52 | carrar | just as much as a pain for a ISP |
| 23:56.53 | Joel | doesn't sound like an isps responsibility |
| 23:56.57 | ariel_ | yes and you also have iptables on your system |
| 23:57.00 | Joel | carrar, fyi, your link doesn't work. |
| 23:57.46 | carrar | ftp://ftp.apnic.net/apnic/dbase/data/country-ipv4.lst |
| 23:58.04 | carrar | Looks like they changed the name of the file |
| 23:58.08 | ariel_ | I don't think isp's should be doing any blockings |
| 23:58.20 | carrar | we would if the customer pays us |
| 23:58.27 | carrar | and says it's ok |
| 23:58.36 | ariel_ | and I don't want the government into it either, little by little we keep loosing our freedom |
| 23:58.37 | carrar | and pays for whatever resources we need to do that |
| 23:58.39 | Joel | ariel_, agreed. |
| 23:59.07 | carrar | I can script a ACL just as easy anyone else |
| 23:59.17 | carrar | might as well get paid for it |