| 00:00.13 | xBull | yes |
| 00:01.06 | Navion | And that's set to your asterisk box's IP? |
| 00:01.13 | ManxPower | .part #non-asterisk-voip-questions |
| 00:01.14 | ManxPower | oops |
| 00:01.25 | xBull | Navion: but there is no configuration for calls incoming from the PSTN and redirect it to SIP server |
| 00:02.01 | ManxPower | xBull: you really should consult the docs for your device. |
| 00:02.03 | *** join/#asterisk implicit_ (n=implicit@vc240226.vpn.uci.edu) |
| 00:02.42 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 00:02.42 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 00:02.48 | Navion | Hmmm... Well, you gotta get there somehow unless it only accepts VOIP -> PSTN calls. |
| 00:03.35 | Navion | Or, yeah, read the docs... Should tell you how to set it up if you're in Austraila. |
| 00:04.40 | xBull | yes... it's very weird |
| 00:04.55 | Navion | Anything like "Stage Method"? |
| 00:06.27 | xBull | in PSTN section it has "switch key" and "digit map" |
| 00:06.47 | xBull | that's all configuration for PSTN |
| 00:07.00 | Navion | Digit map is probably outgoing digit handling. |
| 00:07.15 | Navion | Switch key sounds like a password. |
| 00:07.17 | xBull | yes, it is |
| 00:07.41 | Navion | So, this thing has about a 6 line configuration? |
| 00:07.57 | xBull | switch key is the prefix to force a call through PSTN |
| 00:08.13 | Navion | OK |
| 00:08.34 | *** join/#asterisk tripps (n=ss@72.20.150.196) |
| 00:08.57 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 00:09.02 | *** part/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 00:09.15 | Navion | Well, maybe ManxPower is right. I'm not being much help. Next step is to read the instructions. That hasn't helped me yet but maybe you'll have better luck. |
| 00:10.10 | xBull | yeah, I already read it, I was hoping someone has a similar device, thanks anyway |
| 00:10.30 | Navion | Instructions seem to be written by someone who knows exactly how everything works and all the instructions make sense to them when they read them back to themselves. |
| 00:10.57 | asteriskonsuse | anyone bri working in asterisk-gui? |
| 00:13.56 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 00:14.09 | *** join/#asterisk dijungal (n=kdaniel@209.59.110.30) |
| 00:14.17 | dijungal | hello |
| 00:14.18 | riddlebox | does anyone know if there is a way to unlock the sipura 2100 that broadvoice sent me so I can use both analog ports on it with asterisk? |
| 00:14.34 | dijungal | can i have two hosts= ips on 1 iax2 trunk? |
| 00:15.11 | dijungal | u know like host=xx.xx.zz.1, host=zzz.xxx.ccc.2 |
| 00:15.35 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.138.115) |
| 00:17.04 | *** join/#asterisk tripps (n=ss@72.20.150.196) |
| 00:19.56 | asdx | will asterisk work fine on a virtualized environment (VPS) |
| 00:19.57 | asdx | ? |
| 00:20.31 | blitzrage | not for production use |
| 00:20.40 | russellb | yeah, i've heard mixed results ... |
| 00:20.56 | blitzrage | it DOES work fine ... audio is fine in a lot of my testing... but other times... not so much |
| 00:21.08 | blitzrage | but I would never try it in production without a LOT of testing |
| 00:21.14 | blitzrage | to find out what causes poor audio |
| 00:21.20 | dijungal | asdx: i was just wondering that today, i was watching a video on XEN |
| 00:21.30 | asdx | hm, i see... i will give it a try anyway, if it doesn't work really well then i will go with a "dedicated hosting" |
| 00:21.39 | blitzrage | I've created full out Asterisk clusters on my laptop, but I rarely run more than 1 or 2 calls through it |
| 00:22.15 | asdx | dijungal: nice |
| 00:22.26 | dijungal | blitzrage: using what vitualisation software? on what host? |
| 00:22.31 | asdx | XEN is interesting |
| 00:22.36 | dijungal | very |
| 00:22.41 | blitzrage | I've used xen and VMware Server on Linux |
| 00:22.59 | blitzrage | I love VMware Server -- Xen is a bitch |
| 00:23.07 | *** join/#asterisk tuxd00d (n=tuxd00d@128.187.169.155) |
| 00:23.14 | russellb | vmware server <3 |
| 00:23.17 | asdx | i used to use vmware server on linux |
| 00:23.17 | blitzrage | heck ya |
| 00:23.26 | blitzrage | vmware fusion on OSX |
| 00:23.56 | tuxd00d | I'm not feeling well today.. and my mind is not working right.. i need to figure out if my ISP is, in deed, blocking my incoming IAX2.. could someone help me? |
| 00:24.05 | Mw3 | hm. soxmix has been removed from sox 14.0. is there any alternative mode to record (and join in/out) agent calls? |
| 00:24.21 | blitzrage | tuxd00d: like how... need a port scan? |
| 00:24.36 | asdx | tuxd00d: nmap |
| 00:24.36 | tuxd00d | blitzrage, hey life |
| 00:24.50 | dijungal | vmware server is commercial software? |
| 00:24.56 | blitzrage | tuxd00d: zup |
| 00:25.03 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-13ef6f805d35fbab) |
| 00:25.10 | blitzrage | dijungal: you can download it for free |
| 00:25.16 | dijungal | where? |
| 00:25.26 | dijungal | blitzrage: where? |
| 00:25.38 | Mw3 | you can download anything for free from the internet :) |
| 00:25.39 | blitzrage | www.vmware.com |
| 00:25.45 | dijungal | ohooo |
| 00:25.56 | dijungal | what about vmware player.. that's truly free :) |
| 00:26.05 | Mw3 | so do vmware server |
| 00:26.14 | asdx | i have asterisk running, does this opens a port? |
| 00:26.29 | blitzrage | vmware player lets you play images, but not create them |
| 00:26.37 | asdx | because when i nmap myself i don't see any special port open |
| 00:26.38 | asdx | or listening |
| 00:26.44 | blitzrage | did you open the firewall? |
| 00:26.48 | blitzrage | are you using UDP? |
| 00:28.04 | dijungal | hmm |
| 00:28.12 | asdx | blitzrage: im not sure |
| 00:28.26 | Einsteinium | Howdy, having a dialplan problem. Incoming calls from my PSTN aren't hearing any ringing while waiting for my DIAL to process. |
| 00:28.49 | asdx | how do i see if im using UDP? |
| 00:29.05 | blitzrage | asdx: that statement requires more information |
| 00:29.12 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 00:29.21 | asdx | blitzrage: yeah i know |
| 00:29.28 | blitzrage | ...ok |
| 00:29.41 | Einsteinium | Anyone able to lend a hand |
| 00:30.12 | asdx | im not very sure what UDP is thought |
| 00:30.30 | asdx | transport layer |
| 00:30.38 | asdx | protocol |
| 00:30.39 | blitzrage | asdx: right -- the other one is TCP |
| 00:31.02 | asdx | blitzrage: my asterisk process is probably using TCP |
| 00:31.07 | Einsteinium | blitzrage: any idea how I can make it sound like a ring to an incoming call, if I answer then try dial() |
| 00:31.10 | blitzrage | but Asterisk doesn't support SIP over TCP |
| 00:31.24 | blitzrage | Einsteinium: it should do that automatically |
| 00:31.24 | asdx | hm |
| 00:31.24 | Navion | Unreliable Communictions Protocol. ;) |
| 00:31.40 | blitzrage | nice try -- U *D* P, not C |
| 00:31.42 | Navion | Or Unconnected I forget which |
| 00:31.45 | asdx | blitzrage: right now im running asterisk with IAX2 |
| 00:31.45 | Einsteinium | blitzrage: Nope. Will post my extensions |
| 00:31.49 | Navion | Data |
| 00:31.59 | blitzrage | Einsteinium: to a pastebin |
| 00:32.12 | asdx | how do i know if a process is using TCP/UDP? |
| 00:32.23 | asdx | a network process |
| 00:32.24 | Einsteinium | asdx: trust me, it's using udp |
| 00:32.24 | blitzrage | User Datagram Protocol |
| 00:32.28 | asdx | Einsteinium: ok |
| 00:32.30 | blitzrage | asdx: asterisk only uses UDP |
| 00:32.37 | asdx | blitzrage: ok |
| 00:32.39 | Einsteinium | But in general, check netstat |
| 00:32.45 | Navion | Yeah... That one. I think all the SIP stuff is UDP |
| 00:32.53 | blitzrage | *everything* is UDP |
| 00:33.08 | blitzrage | if you know the difference between udp and tcp, it'll make sense why |
| 00:33.18 | blitzrage | tcp makes no sense for voice traffic |
| 00:33.25 | Einsteinium | blitzrage: http://pastebin.org/7281 |
| 00:33.31 | asdx | i'll have to read wikipedia :-) |
| 00:33.47 | Einsteinium | I added the ringing() recently |
| 00:33.59 | blitzrage | #asterisk: http://pastebin.org/7281 |
| 00:36.28 | *** join/#asterisk BiG^DoG (n=BiG^DoG@c-67-162-233-20.hsd1.de.comcast.net) |
| 00:37.13 | Einsteinium | Anyone? |
| 00:37.41 | BiG^DoG | anyone have any idea why I'm making an FXS to FXO call on my SPA3102 and sometimes (not always) in the middle of the call we get a quick busy and the call is disconnected |
| 00:39.07 | Einsteinium | No one? |
| 00:39.19 | putnopvut | Einsteinium: was blitzrage's pastebin intended for you? |
| 00:39.33 | putnopvut | Oh, wait, it's the same one you posted. oops. |
| 00:39.34 | Einsteinium | That pastebin was mine |
| 00:40.33 | Einsteinium | I am trying to answer first, then have it ring on the users end until I answer, or voicemail picks up |
| 00:46.40 | *** join/#asterisk shmaltz (n=mybox@mail2.dmaven.com) |
| 00:48.57 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 00:49.33 | brookshire | HI! |
| 00:50.00 | shmaltz | hi |
| 00:50.46 | file | brookshire: hai2u |
| 00:50.51 | brookshire | hi! |
| 00:51.15 | brookshire | ha! |
| 00:51.47 | riddlebox | hrmm seems to be a little issue with the asterisk.org site |
| 00:52.58 | brookshire | what's that? |
| 00:53.05 | brookshire | sql error ?? |
| 00:53.12 | Einsteinium | all of digium |
| 00:56.28 | *** join/#asterisk fnordus (n=dnall@24.84.160.227) |
| 00:56.58 | Nugget | mysql blows goats. |
| 00:59.04 | brookshire | mysql is great |
| 01:00.18 | Nugget | mysql is like the windows me of databases. |
| 01:00.49 | Nugget | total rubbush underneath, but cosmetically resembles proper database so people are fooled into trusting it |
| 01:02.57 | brookshire | but at least it scales! |
| 01:03.00 | brookshire | and scales great! |
| 01:03.15 | Nugget | yeah, and backing up your data to /dev/null is really fucking fast too. |
| 01:03.26 | brookshire | why would i do that? |
| 01:03.30 | Nugget | exactly. |
| 01:04.05 | J4k3 | fuck all. |
| 01:04.17 | brookshire | csv is the only way to go |
| 01:06.35 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.ITB.ac.id) |
| 01:08.48 | J4k3 | favorite sql server battle is worse than a distro war |
| 01:08.56 | *** join/#asterisk |dennis| (n=dennis@200.32.236.18) |
| 01:09.12 | *** join/#asterisk zerohalo (n=zeroHalo@c-71-233-235-191.hsd1.ma.comcast.net) |
| 01:09.56 | J4k3 | foxpro. |
| 01:10.06 | *** join/#asterisk Daviey (n=dave@ubuntu/member/daviey) |
| 01:10.30 | BiG^DoG | nyone have any idea why I'm making an FXS to FXO call on my SPA3102 and sometimes (not always) in the middle of the call we get a quick busy and the call is disconnected |
| 01:10.31 | Daviey | Hey, do you think i could expect reliable conference with this zt timing: http://pastebin.ca/764232 thanks |
| 01:15.21 | *** join/#asterisk stridernzl (n=neville@222.155.189.3) |
| 01:17.27 | *** join/#asterisk Raky-2 (n=John@220.157.75.66) |
| 01:18.12 | *** join/#asterisk Trevor_b (n=tbenson@69.12.220.201) |
| 01:20.42 | Trevor_b | If I made a time check to send calls to an external phone systems voicemail (outside of hours) would it be best to do a Transfer instead of a Dial? I never route outside very often. |
| 01:24.46 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 01:26.18 | Trevor_b | damn quiet in here |
| 01:27.51 | *** join/#asterisk hfb (n=hfb@75.80.37.175) |
| 01:36.28 | stridernzl | Trevor_b: It is and I don't know how to answer your Q except i use Dial ... |
| 01:36.56 | Trevor_b | yeah i just went back to dial from transfer ;) |
| 01:43.14 | *** join/#asterisk puck (n=puck@leibniz.catalyst.net.nz) |
| 01:44.41 | puck | Hey, with Asterisk 1.2.x can I have a dialplan after a call to VoiceMail even if the person on the phone hangs up? I'm currently getting spawn extension ... non-zero exit. |
| 01:44.59 | puck | And the dialplan stops being processed |
| 01:45.35 | dijungal | where can i get the 2nd edition Asterisk TFOT ? |
| 01:45.38 | dijungal | to download? |
| 01:49.26 | puck | Hmmm, reading some the logs for this channel it seems that the answer is "no". Suck. |
| 01:50.35 | *** join/#asterisk robl^ (n=robl@pdpc/supporter/monthlybyte/robl) |
| 01:51.36 | Nugget | dijungal: http://safari.oreilly.com/9780596510480/asterisk-BOOKid5949 |
| 01:51.51 | dijungal | thnks |
| 01:53.30 | dijungal | i think u can download the book |
| 01:54.08 | robl^ | no asterisk user should be without a copy on their desk -- digital or dead tree |
| 01:54.30 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 01:56.26 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 01:56.37 | *** join/#asterisk MrTelephone (n=test@S010600195b946fa3.ls.shawcable.net) |
| 01:59.23 | MrTelephone | ~pastebin |
| 01:59.24 | jbot | i guess pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or http://rafb.net/paste/, or http://pastebin.com is usually painfully too slow and unresponsive to use, use one of the other pastebin sites, or dpaste.com is a very nice pastebin as well |
| 02:05.14 | MrTelephone | theres no way those cisco 7960s work behind nat |
| 02:05.22 | MrTelephone | if someone can proove it that would be nice |
| 02:05.58 | MrTelephone | guess i'll sell them on ebay and get some polycom 501s |
| 02:06.12 | MrTelephone | how much do they go for the 7960's? |
| 02:08.13 | *** join/#asterisk BeeBuu (n=chatzill@125.95.28.84) |
| 02:08.46 | BeeBuu | is there anyone here? |
| 02:09.45 | MrTelephone | im not helpin anyone else.. i did a lot of helping and noone can help me with a phone registry |
| 02:09.46 | MrTelephone | :( |
| 02:10.01 | BeeBuu | hi,mrtelphone |
| 02:10.40 | BeeBuu | do you know what's asterisk working by which side? network side or user side? |
| 02:13.16 | *** join/#asterisk gerphimum (n=trekkie@70.125.148.108) |
| 02:14.39 | *** join/#asterisk nighty^^ (n=nighty@sushi.rural-networks.com) |
| 02:14.51 | MrTelephone | i don't understand the question |
| 02:15.20 | BeeBuu | ISDN connect |
| 02:16.05 | MrTelephone | im not sure |
| 02:16.31 | BeeBuu | does on one get my point? |
| 02:18.44 | MrTelephone | the friggin phone is not responding to my god damn messages |
| 02:18.47 | MrTelephone | ($*$*($*$ |
| 02:22.00 | [hC] | so.. on a zap interface (happens to be sangoma) if i press a number while there's audio coming down the line, it doesnt pick up the dtmf. it does in pure silence. is there a common reason for that? |
| 02:22.02 | [hC] | PRI |
| 02:24.36 | [hC] | huh.. a Wait(1) seems to have cured all. that is super super lame. |
| 02:28.49 | [hC] | or, actually the solution was to knock txgain down to -20 |
| 02:28.50 | [hC] | yikes. |
| 02:31.04 | *** join/#asterisk Freman (n=freman@brdr-gw-01.benon.com) |
| 02:31.07 | Freman | ewww... |
| 02:31.15 | Freman | all that work provisioning the other phones |
| 02:31.25 | Freman | and now they want me to make an avaya (of all things) work |
| 02:34.46 | dijungal | hey guys i'm looking for cheap sip/iax provider to nailup my asterisk to and make some test calls |
| 02:34.50 | dijungal | any ideas? |
| 02:36.00 | *** join/#asterisk xBull (n=sabalo@190.3.89.131) |
| 02:40.48 | blitzrage | www.unlimitel.ca, www.nufone.com, www.les.net |
| 02:42.10 | *** part/#asterisk Alowishus (n=jpenix@fwsdo.projectdesign.com) |
| 02:44.04 | *** join/#asterisk Einsteinium (n=99@stjhnf0124w-142162081093.pppoe-dynamic.nl.aliant.net) |
| 02:45.14 | Einsteinium | Anyone have any idea on making it so an inbound call (from PSTN) hears ringing after an Answer() command is issued? |
| 02:46.46 | Freman | eaugh |
| 02:46.51 | Freman | I hate the avaya already |
| 02:47.01 | blitzrage | Einsteinium: what hardware |
| 02:47.29 | Einsteinium | blitzrage: Have SIP coming in from DID provider |
| 02:47.45 | Einsteinium | Asterisk 1.4.13, Debian |
| 02:49.23 | blitzrage | Einsteinium: can you play MoH with the m(class) Dial() option? |
| 02:49.36 | Einsteinium | I can |
| 02:50.04 | Einsteinium | I'd rather avoid MoH if I could, I know there must be a simple way of doing it. |
| 02:50.36 | Einsteinium | Perhaps I could take out the Answer() alltogether, only reason it was there was so if I didn't have my SIP client connected, it would go to voicemail instead of giving a busy tone all the time. |
| 02:50.41 | blitzrage | verify your Asterisk is sending a 180 Ringing back to your service provider -- it might even be the service provider not sending the ringing either to their upstream provider, or indicating it from their switch to the PRI |
| 02:50.44 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.ct.comcast.net) |
| 02:50.59 | Einsteinium | blitzrage: it's only started happening since I added the Answer() |
| 02:51.02 | Einsteinium | before, it would ring |
| 02:51.59 | blitzrage | still check to see if Asterisk is sending the 180 Ringing back to the service provider w/ and w/o the Answer() |
| 02:51.59 | Einsteinium | Besides using a packet monitor, is there an easier way to do that? |
| 02:52.01 | blitzrage | sip debug |
| 02:52.12 | blitzrage | *CLI> sip debug peer my_itsp |
| 02:52.23 | Einsteinium | now, just need to find someone to call me :P |
| 02:52.31 | blitzrage | you don't have a cell or anything? |
| 02:52.39 | Einsteinium | it's at the office |
| 02:53.06 | J4k3 | Einsteinium: skypeout! |
| 02:53.07 | blitzrage | the cell or the anything? |
| 02:53.07 | J4k3 | haha |
| 02:53.30 | Einsteinium | blitzrage: everything except for my SIP client ;) |
| 02:53.35 | blitzrage | Einsteinium: and you should really learn how to use a packet tracer -- it's a necessary skill |
| 02:53.45 | Einsteinium | I know how to use a packet tracer |
| 02:54.13 | Einsteinium | but my Asterisk box is hosted on my colo'd box, and I'd rather not get into installing ethereal/wireshark on it |
| 02:54.19 | blitzrage | I see |
| 02:54.35 | blitzrage | i tend to just install it on all my boxes and use tshark |
| 02:54.44 | Einsteinium | tshark you say? |
| 02:55.03 | blitzrage | ya -- text based version |
| 02:55.06 | Einsteinium | Ah |
| 02:55.10 | blitzrage | I like it better than tcpdump |
| 02:55.18 | Einsteinium | Will look into that |
| 02:57.19 | Einsteinium | Ugh, 180 ringing is being sent |
| 02:57.57 | *** part/#asterisk puck (n=puck@leibniz.catalyst.net.nz) |
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| 03:12.22 | *** join/#asterisk mbit (i=tIRC@218-214-57-65.people.net.au) |
| 03:12.58 | mbit | hey does anyone know if there is a way to turn off the dtmf sound in a conversation? |
| 03:16.03 | *** join/#asterisk Steven_elvisda_ (n=Steven_E@202.47.107.60) |
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| 03:27.08 | *** mode/#asterisk [+o blitzrage] by ChanServ |
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| 03:39.03 | Freman | doh |
| 03:39.10 | Einsteinium | ? |
| 03:39.12 | Freman | now I have to get h323 working with asterisk |
| 03:39.16 | Einsteinium | Why? |
| 03:39.42 | Freman | because they've got 40 of these avaya phones and it's pabx and crap, and they're "attached" |
| 03:39.51 | Einsteinium | Ah |
| 03:40.08 | Freman | easiest way for me to get out of an argument right now is to make asterisk talk to the phones/existing pabx until things start to die (out of production stuff) |
| 04:02.49 | *** join/#asterisk techie (n=techie@adsl-76-214-19-182.dsl.lsan03.sbcglobal.net) |
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| 04:07.47 | *** join/#asterisk antlers (n=antlers@ip70-173-89-173.lv.lv.cox.net) |
| 04:15.35 | *** join/#asterisk dlynes_laptop (n=dlynes@d154-20-9-152.bchsia.telus.net) |
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| 04:17.19 | brookshire | yay.. asterisk.org fixed |
| 04:17.26 | brookshire | stupid drupal caching |
| 04:29.21 | De_Mon | any Team Fortress 2 players in here? |
| 04:31.00 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 04:31.00 | *** mode/#asterisk [+o russellb] by ChanServ |
| 04:34.43 | MrTelephone | not right now |
| 04:34.49 | MrTelephone | demon are u a nat expert? |
| 04:35.01 | De_Mon | no Im smart enuf to avoid it |
| 04:35.15 | De_Mon | I need more friends to play TF with |
| 04:36.03 | [TK]D-Fender | MrTelephone, what about it? |
| 04:36.13 | MrTelephone | does asterisk rewrite the port in sdp upon first packet received from a natted client? |
| 04:36.24 | *** join/#asterisk implicit (n=implicit@c-67-191-24-188.hsd1.fl.comcast.net) |
| 04:36.25 | MrTelephone | and i still think the cisco 7960 has major issues |
| 04:36.26 | MrTelephone | heh |
| 04:37.33 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 04:40.23 | MrTelephone | fender? |
| 04:42.25 | [TK]D-Fender | hrm... maybe a little out of my experience. Last ai heard Cisco + NAT = trouble |
| 04:43.13 | [TK]D-Fender | MrTelephone, I do know that if you tell * that the client is NAT'd it'll send it back to the originating IP, and not the one reported in SIP. |
| 04:43.18 | [TK]D-Fender | (/rtp) |
| 04:46.15 | *** join/#asterisk implicit_ (n=implicit@vc240232.vpn.uci.edu) |
| 04:46.37 | MrTelephone | im curious as to the sdp because the client announces its port before anything is sent |
| 04:47.04 | MrTelephone | d-links try and use the same outside port |
| 04:55.55 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 04:56.13 | [TK]D-Fender | MrTelephone, D-Link's are often very bad with NAT |
| 04:58.37 | *** join/#asterisk thx2000 (n=evan@netblock-208-127-150-56.dslextreme.com) |
| 05:05.35 | Freman | I love my new management... |
| 05:05.42 | Freman | eaugh... this involves recompiling asterisk from scratch, can't just make an addon |
| 05:05.50 | Freman | lets just settle for it "Can be done - if we have to" (c: |
| 05:05.58 | Freman | he says: or even "it can't be done buy new phones" |
| 05:14.12 | *** join/#asterisk peanut- (n=tokarev@cpe-70-113-100-193.austin.res.rr.com) |
| 05:15.42 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 05:16.45 | fujin | blagh. |
| 05:16.53 | fujin | Manager turned down new Polycom phones |
| 05:16.58 | fujin | after I had already applied for an account |
| 05:17.00 | fujin | what a cocknocker. |
| 05:17.20 | fujin | was going go from my current standard, spa942's, to IP330's |
| 05:17.20 | J4k3 | blow up his car, kill his family, and pee on his head |
| 05:17.24 | fujin | man, I will |
| 05:17.30 | fujin | hobbit-looking-motherfucker |
| 05:17.31 | fujin | god damnit. |
| 05:18.05 | fujin | ha |
| 05:18.16 | russellb | we're good friends, you know. |
| 05:18.20 | fujin | I'm sure. |
| 05:18.22 | russellb | jk :-p |
| 05:18.24 | fujin | hehe |
| 05:18.34 | J4k3 | you hang out with hobbits?! |
| 05:18.46 | MrTelephone | stick with polycoms |
| 05:18.59 | russellb | polycoms are definitely my favorite ... |
| 05:19.11 | J4k3 | #qvc |
| 05:19.11 | fujin | I can't stick WITH them, I was trying to change to them! |
| 05:19.15 | J4k3 | #homeshoppingnetwork |
| 05:19.21 | fujin | I have to stick to Linksys :\ |
| 05:19.36 | fujin | bah, if only I had of known there was a local polycom distributor when I purchased 50 SPA942's. |
| 05:20.20 | [TK]D-Fender | fujin : that'll learn ya.... |
| 05:20.27 | fujin | aye |
| 05:20.38 | fujin | ah well, the Linksys have been stable, fortunately |
| 05:20.51 | fujin | I inherited a big box of Mitels when I arrived and there was no damn way I was using those. heh. |
| 05:20.54 | fujin | mitel = cringe |
| 05:20.58 | fujin | I'm sure mitel are right down there with GS |
| 05:21.26 | J4k3 | send them to me |
| 05:21.55 | fujin | ha, they're still sitting around in boxes. I should really get around to hocking them off |
| 05:22.42 | J4k3 | all that I need to work out is the battery control circuit and aquiring the battery itself |
| 05:22.55 | J4k3 | might just go see what radio shack's clearance bin has to offer... they've always got old cell batteries in there |
| 05:23.03 | J4k3 | well |
| 05:23.08 | J4k3 | brand new batteries for 'old' phones |
| 05:24.36 | J4k3 | or, throw a lead acid battery in the thing and call it ballast. |
| 05:24.44 | J4k3 | "it ensures the phone stays on the desk" |
| 05:24.44 | [TK]D-Fender | fujin, Actually I've heard Mitel is supposedly pretty good... |
| 05:25.02 | J4k3 | for d-fender to poop on? |
| 05:25.03 | J4k3 | :D |
| 05:25.10 | fujin | maybe in a Mitel environment |
| 05:25.13 | fujin | the SIP support is terrible |
| 05:25.18 | fujin | we had no end of issue trying to automatically provision them |
| 05:25.29 | fujin | including 10~ suicidal phones |
| 05:26.03 | J4k3 | jtag 4 life |
| 05:26.22 | Mavvie | hmm.... since the upgrade to the latest 1.4 version we have had about 40% less calls and call minutes. |
| 05:26.25 | Mavvie | wonder what goes wrong here. |
| 05:26.39 | Mavvie | could be a CDR thing. |
| 05:27.05 | J4k3 | 1.4 was designed to reduce call center load by forwarding 40% of your calls to a competitor's phone system. |
| 05:27.10 | Raky-2 | hey fellas, i've just installed the cdr mysql module, however it doesn't seem to be writing to my mysql database. |
| 05:27.22 | Raky-2 | i have the correct cdr_mysql.conf, and i also have checked to see that the module is loaded |
| 05:27.32 | Raky-2 | however, i don't see it parsing cdr_mysql.conf, is there some way i can force it to parse the file? |
| 05:27.35 | Mavvie | J4k3: if I had a call center I would believe it :-) |
| 05:27.39 | *** join/#asterisk xpot (n=jim@c-71-195-241-115.hsd1.ut.comcast.net) |
| 05:28.40 | *** join/#asterisk BeeBuu (n=chatzill@125.95.28.84) |
| 05:28.53 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 05:29.05 | linagee | Mavvie: that's interesting |
| 05:29.53 | Mavvie | [Nov 7 16:20:10] NOTICE[5777] cdr.c: CDR on channel 'Zap/93-1' not posted |
| 05:29.55 | Mavvie | wonder if that is related. |
| 05:32.07 | J4k3 | Mavvie: hehe |
| 05:35.16 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 05:37.26 | xpot | can anyone help with the following error? --> [Nov 6 22:36:57] ERROR[14120]: pbx.c:1527 ast_func_read: Function ODBC_CARDNUM not registered |
| 05:38.46 | Mavvie | xpot: seems easy, you are missing a module. |
| 05:39.13 | *** join/#asterisk TJNII (n=TJNII@209.234.89.226) |
| 05:39.25 | xpot | using freetds to connect to MSSQL server, both unixODBC and FreeTDS are installed, when I run "make menuselect" XXX 4. cdr_odbc, [*] 8. cdr_tds |
| 05:39.56 | xpot | do I need cdr_odbc? if so why is the option not available? |
| 05:40.18 | xpot | using latest svn 1.4 |
| 05:40.23 | xpot | branches |
| 05:42.08 | xpot | Mavvie: I felt this is was simple as well, for I have successfully installed this option in the past... I am not sure if this is a latest 1.4 issue or not. |
| 05:42.19 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
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| 05:51.08 | *** join/#asterisk TrentCreek (n=tjones14@cpe-70-117-207-168.rgv.res.rr.com) |
| 05:51.18 | sevard | does anyone have FWD that could do a test call to me? |
| 05:53.59 | *** join/#asterisk macros73_ (n=cs@c-24-3-246-27.hsd1.mn.comcast.net) |
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| 06:06.17 | *** part/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-13ef6f805d35fbab) |
| 06:07.58 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-90711961e706084b) |
| 06:15.28 | *** part/#asterisk axisys (i=iqbala@outbound.silenceisdefeat.org) |
| 06:20.30 | *** join/#asterisk implicit (n=implicit@c-67-191-24-188.hsd1.fl.comcast.net) |
| 06:22.14 | ManxPower | Mavvie: check the CDR to see if it was actually posted. IIRC, someone was talking somewhere (mailing list maybe) about Asterisk giving him that message, but when he looked at the CDRs, it was actually posted. |
| 06:23.18 | ManxPower | Mavvie: lets just hope it's not DROPPING 60% of the calls. |
| 06:24.07 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 06:26.24 | Mavvie | ManxPower: no, it's not dropping, I would have had calls about it. |
| 06:27.22 | Mavvie | ManxPower: it seems (from the source) that its only logged, but not dropped. |
| 06:27.46 | Mavvie | ManxPower: I will have to check at the end of this month with the bill from our uplink (oh oh) |
| 06:28.42 | ManxPower | Mavvie: you have the date, and time, and channel from the console message, go look in the CDR file to see if it's there. |
| 06:31.16 | Mavvie | ManxPower: yes it is there. |
| 06:32.00 | Mavvie | it looks like all calls are unanswered. |
| 06:32.56 | Mavvie | yeah, they're all failed on no answer. |
| 06:34.41 | tzafrir_home | anything wrong with http://asterisk.org/ ? |
| 06:34.55 | tzafrir_home | It returns to me something that looks like garbage |
| 06:35.02 | tzafrir_home | gzipped content? |
| 06:35.28 | Mavvie | tzafrir: somebody posted about a joomla upgrade on it. |
| 06:36.08 | Mavvie | <brookshire> yay.. asterisk.org fixed |
| 06:36.09 | Mavvie | <brookshire> stupid drupal caching |
| 06:37.52 | tzafrir_home | Here's what I get: ��������mo�6�� (a whole bunch of those) |
| 06:38.29 | Mavvie | ManxPower: still doesn't explain why we lost about 40% of calls and call minutes after last weeks upgrade to the latest 1.4 branch... |
| 06:39.22 | tzafrir_home | same with lynx. It mentions something about .html.gz ... |
| 06:39.49 | tzafrir_home | wget gets it right, and the page is 12k, instead of 3k with the others |
| 06:43.16 | *** join/#asterisk Keltus (i=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
| 06:46.18 | ManxPower | Mavvie: well downgrade to what you had and see if you see a 40% increase in calls. |
| 06:47.32 | *** join/#asterisk McDouglas (n=mcd@mmcomp.adsl.datanet.hu) |
| 06:49.06 | tzafrir_home | ah, ok. Had to use ctrl-shift-r to bypass local cache |
| 06:52.09 | hmmhesays | i'll bypass your local cache |
| 06:54.51 | *** join/#asterisk Schumie (i=SteveWri@cpc2-rdng2-0-0-cust382.winn.cable.ntl.com) |
| 07:04.44 | Mavvie | the announcement is a little bit mangled. |
| 07:04.59 | Mavvie | it says 1.4.12 announced, then 1.4.13 announced, then 1.4.11 announced. |
| 07:05.04 | Mavvie | s/announced/released/g |
| 07:05.37 | tzafrir_home | anybody ever ran into the menuselect (or is it mxml) config.status hanging in 100% CPU while generating Makefile? |
| 07:05.48 | tzafrir_home | sed is in a 100% CPU loop |
| 07:07.57 | tzafrir_home | after using ltrace, the loop seems to be in the line ov @INSTALL@ |
| 07:10.40 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 07:12.54 | *** join/#asterisk J4zen (n=Jvan4zen@a213-84-139-244.adsl.xs4all.nl) |
| 07:18.37 | brookshire | tzafrir_home: still problems? |
| 07:19.00 | tzafrir_home | with the site? not after ctrl-shift-r |
| 07:19.34 | tzafrir_home | I think that lynks has a problem dealinng with it, though |
| 07:20.15 | tzafrir_home | hmm.... now works in lynx as well |
| 07:20.36 | J4k3 | I hate cellcos, I hope they all catch something and die |
| 07:20.50 | brookshire | # uptime 01:18:40 up 117 days, 4:15, 3 users, load average: 8.23, 4.53, 3.11 |
| 07:20.53 | brookshire | :( |
| 07:21.01 | brookshire | poor asterisk.org |
| 07:26.01 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 07:33.52 | tzafrir_home | (because you can't blame the users) |
| 07:34.18 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 07:34.22 | *** join/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 07:34.56 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 07:36.22 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 07:37.59 | McDouglas | anyone know how can i reduce on an SPA-921 the time it waits before dialing the number automaticaly? (if the user does not press the dial softkey) |
| 07:45.35 | *** join/#asterisk aki_lidi (i=sonny@202.72.218.14) |
| 07:48.35 | TrentCreek | nope |
| 07:52.51 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 07:57.56 | *** join/#asterisk Boones (i=Boones@port-212-202-42-6.dynamic.qsc.de) |
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| 08:33.18 | *** join/#asterisk BeeBuu (n=chatzill@125.95.28.84) |
| 08:33.24 | BeeBuu | hi,all |
| 08:33.28 | *** join/#asterisk CCFL_Man2 (i=6212e021@pool-70-105-210-30.scr.east.verizon.net) |
| 08:33.30 | *** join/#asterisk yassaccan (n=yassacca@admin189.hgo.se) |
| 08:33.43 | *** part/#asterisk CCFL_Man2 (i=6212e021@pool-70-105-210-30.scr.east.verizon.net) |
| 08:34.05 | BeeBuu | i had a 4E1 card |
| 08:34.44 | BeeBuu | extentsions.conf: |
| 08:34.45 | BeeBuu | [test222] |
| 08:34.47 | BeeBuu | exten=>s,1,Dial(SIP/999) |
| 08:34.48 | BeeBuu | exten=>s,n,HangUp() |
| 08:35.51 | BeeBuu | zapata.conf: |
| 08:35.53 | BeeBuu | [channels] |
| 08:35.55 | BeeBuu | switchtype = euroisdn |
| 08:35.56 | BeeBuu | signalling = pri_cpe |
| 08:35.58 | BeeBuu | group = 0 |
| 08:35.59 | BeeBuu | ;context=test222 |
| 08:36.01 | BeeBuu | channel => 1-15,17-31,32-46,48-52 |
| 08:36.03 | BeeBuu | language=en |
| 08:36.04 | BeeBuu | ;context=from-zaptel |
| 08:36.06 | BeeBuu | context=test222 |
| 08:40.33 | JT | BeeBuu: stop that |
| 08:40.36 | JT | ~pb |
| 08:40.37 | jbot | A Pastebin is a place to paste your stuff without flooding 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://www.rafb.net/paste |
| 08:40.59 | BeeBuu | sorry about that |
| 08:44.30 | BeeBuu | no more flooding... |
| 08:46.46 | *** join/#asterisk fnordus (n=dnall@24.84.160.227) |
| 08:48.51 | *** join/#asterisk _ys (n=yuri@80.70.236.69) |
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| 09:05.46 | McDouglas | is it possible to make asterisk ring back to me if i transfer a call to a different extension and they dont pick it up? |
| 09:06.49 | *** join/#asterisk alpha232 (i=alpha232@198-144-143-60.dyn.megabroadband.net) |
| 09:07.23 | *** join/#asterisk Shaun2222 (n=shaun@ip68-4-127-67.oc.oc.cox.net) |
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| 09:13.00 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 09:14.03 | L|NUX | can some one tell me what is this warrning for |
| 09:14.05 | L|NUX | [Nov 6 14:29:10] WARNING[15205]: sip_write: Asked to transmit frame type 256, while native formats is 0x400 (ilbc)(1024) read/write = 0x400 (ilbc)(1024)/0x400 (ilbc)(1024) |
| 09:26.49 | *** join/#asterisk Miisha (n=Miisha@c103-139.icpnet.pl) |
| 09:27.05 | Miisha | hello :) |
| 09:27.13 | Miisha | i've a little problem |
| 09:27.53 | Miisha | how i can constraint rights for the users ? |
| 09:28.21 | Miisha | you now, i want only one person have permissed to the only one phone |
| 09:28.29 | Miisha | + one head admin |
| 09:29.09 | Miisha | have access* ofcourse :) |
| 09:29.12 | case_ | hello |
| 09:30.04 | Miisha | i've read asteriskguru.com, asterisk.org |
| 09:30.14 | case_ | If i use system() in my dialplan to call a potentialy long and cpu intensive command, can it makes asterisk hang? |
| 09:30.15 | Miisha | but i didn't find inof about it :/ |
| 09:30.40 | *** join/#asterisk qdk (n=qdk@85.235.253.139) |
| 09:30.54 | case_ | Miisha, what kind of phone ? |
| 09:31.10 | Miisha | Linksys SPA921 |
| 09:31.24 | case_ | Miisha, i mean that's SIP phones? |
| 09:31.29 | Miisha | yes |
| 09:31.55 | *** join/#asterisk bantu (n=Miranda@rz-du-fgx-146-148.rz.uni-karlsruhe.de) |
| 09:31.58 | case_ | set up proper login/passwords, and after that you can apply any restrictions you want inside your dialplan |
| 09:32.36 | Miisha | but i'll find it in some .conf file ? |
| 09:32.50 | case_ | what is "it" in your sentence? :) |
| 09:33.18 | *** part/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-90711961e706084b) |
| 09:33.28 | Miisha | add noe users with constraint rights |
| 09:33.31 | Miisha | new* |
| 09:34.03 | Miisha | ouhh... my english sux :] |
| 09:34.08 | case_ | Miisha, you set up the sip accounts in sip.conf, and after that you can constraint rights in extentions.conf |
| 09:34.23 | case_ | (don't talk about mine ;p ) |
| 09:34.30 | Miisha | :P |
| 09:34.40 | Miisha | case_ next stupid question, how :] |
| 09:36.22 | case_ | you have different way to do so, i suggest you read the wiki articles about sip.conf and extentions.conf |
| 09:36.50 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-9aea077da35c646d) |
| 09:36.50 | case_ | sorry i've not the time to discuss it right now (and i'm certainly not the more skilled to do so) |
| 09:37.04 | Miisha | ok case_, no problem, thx for help :) |
| 09:37.24 | case_ | good luck |
| 09:37.40 | Miisha | i'll need, haha :D |
| 09:38.37 | case_ | ...but if someone wish to talk about the System() command in asterisk, it'll help me a lot :) |
| 09:39.43 | case_ | what happends if i put "System(sleep 10)" ? does asterisk hang for 10 seconds or is it forked in the background ? |
| 09:42.38 | Miisha | ok, so in asterisk is possible, than one user have access only for one phone (eg. only) |
| 09:42.54 | Miisha | m'i in right ? |
| 09:45.12 | case_ | what do you mean by "one phone" ? |
| 09:45.52 | Miisha | in callcenter |
| 09:46.22 | case_ | do you mean "one user can only dial to on phone number" ? |
| 09:46.59 | case_ | or "this physical phone can only be used by Mr Foo" ? |
| 09:47.29 | Miisha | no, one person can use only one phone, and don't have acces for some server's settings |
| 09:48.10 | Miisha | i'm not in VoIP, but i have now turtorials in some company ;/ |
| 09:48.30 | Miisha | and this is "my quest" :P |
| 09:48.39 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 09:48.44 | case_ | this has to be set on the sip phone side. i mean. you have user/password set in asterisk, it's up to you to make the phone "forget" the passowrd and ask the user for it each time |
| 09:48.59 | case_ | is that what you want ? |
| 09:50.07 | Miisha | no, i want for example One Head Admin, who have acces for all server's setting |
| 09:50.32 | Miisha | and for example 20 person without acces, who can only use phone |
| 09:50.53 | Miisha | and this 20 person don't have acces for settings |
| 09:51.03 | case_ | but... you don't access to asterisk settings with the phone, do you ??? |
| 09:51.04 | Miisha | it stupid, i know ;/ |
| 09:51.07 | *** join/#asterisk Schumie (n=Steve@cmarfw01.marlow.spinvox.com) |
| 09:51.13 | Miisha | i have |
| 09:51.24 | Miisha | i installed asterisk on my PC |
| 09:51.42 | Miisha | and i have all files, and everything what i need |
| 09:52.42 | case_ | yes... and? |
| 09:53.05 | *** join/#asterisk munmun (n=mun_mun@203.80.176.168) |
| 09:53.32 | *** part/#asterisk munmun (n=mun_mun@203.80.176.168) |
| 09:53.37 | Miisha | so... i have to use sip.conf and extensions.conf, yes ? |
| 09:54.48 | *** part/#asterisk Miisha (n=Miisha@c103-139.icpnet.pl) |
| 10:01.42 | *** join/#asterisk Guggemand (i=guggeman@80.198.131.46) |
| 10:03.20 | Guggemand | anyone know what causes a phone to be "on hold" in a queue? - "Sip/201 with penalty 1 (dynamic) (On Hold)" |
| 10:06.33 | *** join/#asterisk gmustafa (n=gmustafa@40.104.181.58.dynamic.max.com.pk) |
| 10:06.54 | gmustafa | i am having problem configuring asterisk+res_config_ldap |
| 10:07.11 | gmustafa | i have compiled asterisk with res_config_ldap successfully |
| 10:07.26 | gmustafa | res_config_ldap can bind successfully with ldap |
| 10:07.32 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 10:07.56 | gmustafa | but its not quering ldap for sippeers or sipusers |
| 10:15.33 | gmustafa | ??? |
| 10:16.06 | *** join/#asterisk Steven_elvisda_ (n=Steven_E@202.47.107.60) |
| 10:16.45 | *** join/#asterisk IPetrov2 (i=IPetrov2@ppp85-140-232-177.pppoe.mtu-net.ru) |
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| 10:22.39 | Spida | I get reproducible freezes loaading misdn's hfcpci module. how can I fix that? |
| 10:23.46 | *** join/#asterisk af_ (n=getsmart@88-149-240-124.dynamic.ngi.it) |
| 10:26.16 | tzafrir | Spida, post a bug with relevant details? |
| 10:26.36 | case_ | i think the gurus are sleeping, but I have strange problems receiving faxes... |
| 10:26.53 | tzafrir | no european mISDN guru?? |
| 10:28.29 | Spida | tzafrir: I found http://www.mail-archive.com/misdn-asterisk@lists.beronet.com/msg00405.html which seems to be the same issue |
| 10:28.42 | Chris-NB | hi |
| 10:28.48 | tzafrir | Spida, don't ask me about misdn ... |
| 10:29.10 | Chris-NB | anyone uses a SPA3102 ? |
| 10:29.26 | case_ | here's my problem: when i receive a fax, the tif file is OK (seems complete), but the System() just after RxFax() is never called... |
| 10:29.29 | Chris-NB | or know if it is possible to establish a direct sip (t.38) between two of those? |
| 10:29.39 | case_ | and i have theses two warnings: |
| 10:29.41 | case_ | WARNING[28139]: channel.c:2326 set_format: Unable to find a codec translation path from unknown to unknown |
| 10:29.46 | case_ | WARNING[28139]: app_rxfax.c:315 rxfax_exec: Unable to restore write format on 'Zap/53-1' |
| 10:30.23 | tzafrir | case_, what card is it? |
| 10:30.32 | tzafrir | E1 or T1? |
| 10:30.36 | *** join/#asterisk jarod14 (n=jarod14@ns2.viatelecom.com) |
| 10:30.54 | tzafrir | By the time of day I'd guess E1 and alaw format |
| 10:31.08 | case_ | Digium, Inc. Wildcard TE210P (rev 02) |
| 10:31.21 | case_ | tzafrir, E1 i think, i'm in france |
| 10:33.34 | tzafrir | I think you should be able to see the format in 'zap show channel 53' |
| 10:38.45 | case_ | hmmm what format are you talking about ? |
| 10:39.02 | case_ | Default law: alaw |
| 10:39.02 | case_ | Fax Handled: no |
| 10:39.08 | case_ | does it help ? |
| 10:39.22 | case_ | i don't know what it means... |
| 10:40.05 | *** join/#asterisk rigid (n=The@port-212-202-78-56.dynamic.qsc.de) |
| 10:40.07 | rigid | re |
| 10:40.34 | rigid | can someone tell me what I have to ask my SIP provider if I want to send SMS? |
| 10:40.45 | rigid | (is that protocol part in SIP also called SMS?) |
| 10:42.13 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 10:42.20 | JT | no |
| 10:42.27 | JT | sip is not related to sms |
| 10:42.35 | *** join/#asterisk angom (n=Angel@201.170.35.218) |
| 10:42.55 | jwh | you can send messages in sip, how they're dealt with at the upstream is upto them |
| 10:43.09 | jwh | but presumably if they have an SMS gateway, they can relay it onto GSM networks |
| 10:43.43 | JT | and the proportion of ITSPs who support SMS over SIP: almost none |
| 10:44.49 | gmustafa | i have compiled asterisk with res_config_ldap successfully |
| 10:44.53 | gmustafa | res_config_ldap can bind successfully with ldap |
| 10:44.56 | gmustafa | but its not quering ldap for sippeers or sipusers |
| 10:45.03 | gmustafa | any idea? |
| 10:48.14 | dandre | Hello, |
| 10:49.14 | dandre | Is there any documentation or tutorial form MeetMe application other than th book or http://www.voip-info.org/wiki-Asterisk+cmd+MeetMe ? |
| 10:49.34 | FlatFoot | morning all |
| 10:49.49 | FlatFoot | anyone help with a daft question on callerid(name) ? |
| 10:50.05 | rigid | JT, jwh ah... thanks |
| 10:54.00 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 10:55.15 | jwh | I have SMS as an addon feature, but no one really wants it |
| 10:57.05 | *** join/#asterisk J4zen (n=Jvan4zen@a213-84-139-244.adsl.xs4all.nl) |
| 11:00.16 | J4zen | Could anyone tell me where asterisk stores its audio files ( eg. voicemail? ) |
| 11:00.45 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 11:01.16 | dandre | Is there any documentation or tutorial form MeetMe application other than th book or http://www.voip-info.org/wiki-Asterisk+cmd+MeetMe ? |
| 11:01.44 | IPetrov2 | J4zen: i can see all asterisk folders in /usr/loca/etc/asterisk/asterisk.conf |
| 11:02.07 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 11:07.10 | J4zen | THanks IPetrov2 |
| 11:07.47 | JT | jwh: not over sip anyway |
| 11:07.52 | *** join/#asterisk _ys (i=ys@91.151.196.254) |
| 11:09.27 | J4zen | IPetrov2: I don't see the audio files listed in there |
| 11:09.38 | jwh | JT: well, it is possible to accept text strings in sip messages |
| 11:09.47 | jwh | but i've only implemented it in theory |
| 11:11.43 | dandre | I can't find some good documentation about meetme on the web. I must do something wrong! |
| 11:13.01 | tzafrir | dandre, show application meetme ? |
| 11:13.07 | *** join/#asterisk skymeyer (n=skymeyer@bxlsrvit03.itconnect.be) |
| 11:14.18 | dandre | tzafrir: yes but |
| 11:14.18 | dandre | <PROTECTED> |
| 11:14.18 | dandre | -> what is admin mode and how to use it? |
| 11:14.38 | dandre | same for marked mode and so on |
| 11:14.51 | IPetrov2 | J4zen: u need astspooldir variable |
| 11:15.10 | JT | jwh: heh |
| 11:15.19 | J4zen | i found it IPetrov2 :) Thanks |
| 11:19.02 | *** join/#asterisk Miisha (n=Miisha@c103-139.icpnet.pl) |
| 11:20.50 | FreezeS | hey gys |
| 11:21.01 | FreezeS | how can I trim the spaces at the end of a variable ? |
| 11:24.23 | IPetrov2 | FreezeS: i think u can use agi :) |
| 11:24.44 | FreezeS | isn't there a simpler way ? |
| 11:25.05 | Miisha | in Asterisk GUI, can i make user with some other than "read = all" |
| 11:25.10 | IPetrov2 | u need spaces in other string positions ? |
| 11:25.12 | Miisha | ? |
| 11:26.00 | IPetrov2 | if not u can try FILTER |
| 11:26.31 | J4zen | Hmm odd, when i use sox to convert a WAV file to GSM.. the resulting file has its pitch turned down big time, resulting in a sound file with a very dark and slow voice lol |
| 11:26.37 | *** join/#asterisk blq (n=Bl@dslb-088-064-154-135.pools.arcor-ip.net) |
| 11:26.46 | J4zen | What could have caused this ( or better, how do i prevent/fix this? ) |
| 11:27.44 | *** join/#asterisk michael-i (n=michael-@W8320.w.pppool.de) |
| 11:28.38 | *** part/#asterisk robl^ (n=robl@pdpc/supporter/monthlybyte/robl) |
| 11:29.53 | J4zen | What audio convertor do you all use? |
| 11:29.56 | J4zen | or recommend |
| 11:30.40 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 11:34.21 | J4zen | What audio convertor/recorder do you use or recommend (output: gsm/ulaw)? I'm currently using CoolEdit to record and Sox to convert.. but its not going as i wish lol |
| 11:34.41 | *** join/#asterisk ccesario (n=ccesario@189.20.219.11) |
| 11:34.55 | tzafrir | I use sox , and occasionally audacity |
| 11:35.16 | tzafrir | But mostly sox for conversions |
| 11:35.53 | J4zen | if i do "sox vm-intro.wav vm-intro.gsm" i get a scrambled gsm file |
| 11:36.07 | J4zen | the vm-intro.wav was recorded using cooledit and plays just fine |
| 11:36.16 | J4zen | do u have any clue what could cause this tzafrir? |
| 11:36.24 | *** join/#asterisk penguinFunk (n=penguin@unaffiliated/penguinfunk) |
| 11:37.08 | tzafrir | J4zen, file vm-intro.wav vm-intro.gsm |
| 11:38.06 | J4zen | vm-intro.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 44100 Hz |
| 11:38.12 | J4zen | and:\ |
| 11:39.55 | tzafrir | sox vm-intro.wav -r 8000 vm-intro.gsm |
| 11:39.58 | tzafrir | maybe? |
| 11:40.23 | J4zen | 8000 is the maximum bitrate ?? |
| 11:41.02 | tzafrir | for phone-quality gsm? yes |
| 11:41.13 | Miisha | how i can restrict permission new users ? |
| 11:41.14 | tzafrir | for phone-quality audio? basically yes |
| 11:41.30 | tzafrir | What type of users? |
| 11:41.40 | J4zen | I see, thanks tzafrir. Let me try to alter the bitrate |
| 11:41.41 | Miisha | in asterisk gui |
| 11:41.58 | Miisha | have acces onyl for some function, but not all |
| 11:42.03 | tzafrir | don't give them any access to the manager interface |
| 11:42.22 | J4zen | tzafrir: That was the cause of the problem |
| 11:42.25 | tzafrir | Or at least no write access |
| 11:42.30 | J4zen | bitrate was at 41k |
| 11:42.33 | *** join/#asterisk eurodrew555 (n=eurodrew@83-244-164-130.cust-83.exponential-e.net) |
| 11:43.01 | Miisha | yes, buy if i give write = log, than i can't log by this username and password |
| 11:43.22 | Miisha | but* |
| 11:44.41 | Miisha | i want than user "test" can do smoething only in ring groups, for example |
| 11:44.49 | Miisha | something* |
| 11:45.23 | Miisha | is it possible ? :> |
| 11:48.56 | *** join/#asterisk harpal (n=Harpal@124.125.254.227) |
| 11:55.16 | DrAk0 | is it possible on a middle of a convo invite to another person (like create a dinamic conf?) |
| 11:55.58 | jwh | DrAk0: yes |
| 11:55.58 | DrAk0 | jwh, references please :) |
| 11:56.03 | jwh | DrAk0: standard invite |
| 11:56.13 | jwh | or rather, client side |
| 11:56.33 | jwh | ie; initiate another call on your phone, then bridge, other than that, not sure |
| 11:57.14 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.87.168) |
| 11:57.36 | DrAk0 | how do you bridge? |
| 11:57.41 | DrAk0 | ill read about invite |
| 11:58.27 | CyberSyx^ | anyone using chan_mobile on asterisk ? |
| 11:58.28 | jwh | DrAk0: depends on the phone |
| 11:58.45 | DrAk0 | jwh, using counter path |
| 12:00.28 | Spida | does anybody know how to fix crashes on insertion of misdns hfcpci-module? |
| 12:00.33 | *** join/#asterisk HeMan (n=jimmy@1-1-7-40a.far.sth.bostream.se) |
| 12:01.31 | *** part/#asterisk HeMan (n=jimmy@1-1-7-40a.far.sth.bostream.se) |
| 12:01.34 | *** join/#asterisk HeMan (n=jimmy@1-1-7-40a.far.sth.bostream.se) |
| 12:01.40 | tzafrir | Spida, file a bug report with the relevant details |
| 12:02.00 | tzafrir | that is the best thing to do if noone is available right now |
| 12:04.20 | ai-a | the relevant details might fix his problem ;) |
| 12:04.34 | ai-a | Spida: access rights ? valid module ? |
| 12:05.19 | DrAk0 | jwh, http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO |
| 12:06.35 | HeMan | Any ideas to keep Snom/elmeg-phones from _not_ adding numbers to missed calls when our group-number is called? |
| 12:07.58 | ai-a | HeMan: turn off the missed call notification. |
| 12:08.06 | ai-a | but that means all missed calls are not tracked. |
| 12:08.24 | ai-a | a 'phone' doesnt know the difference between a group ringing and a independent ring. |
| 12:08.40 | ai-a | its in the Identity on Snom.. you could do different identity used for the group. |
| 12:09.04 | HeMan | hmm, good idea! |
| 12:09.06 | ai-a | then that identity (for group A) wont store missed calls, but the identity (for me) will store missed calls. |
| 12:09.48 | Spida | tzafrir: what do you consider as "relevant details"? |
| 12:12.23 | ai-a | Spida: http://asterisk.org/developers/bug-guidelines |
| 12:13.13 | *** join/#asterisk acehunky (n=acehunky@59.184.139.154) |
| 12:13.22 | acehunky | hello |
| 12:13.38 | acehunky | i am having some choppy audio problem when i use ChanSpy |
| 12:13.47 | acehunky | anyone out here is facing similar problem ? |
| 12:15.10 | acehunky | anyone alive ? |
| 12:15.42 | *** join/#asterisk coppice (n=chatzill@39.192.17.210.dyn.pacific.net.hk) |
| 12:16.10 | Spida | ai-a: I have a problem with misdn, not with asterisk. I don't even have asterisk installed (yet) |
| 12:16.45 | coppice | misdn *is* a problem |
| 12:17.03 | acehunky | hi steve |
| 12:17.12 | *** join/#asterisk dijungal (n=kdaniel@63.175.159.171) |
| 12:17.17 | dijungal | good day |
| 12:17.32 | dijungal | what AEL version does asterisk 1.2 use? |
| 12:18.02 | acehunky | anyone over here who can help me with ChanSpy ? |
| 12:18.14 | Spida | coppice: yeah, but zaptel doesn't work for me, either |
| 12:18.15 | dijungal | i use it |
| 12:18.32 | dijungal | i use chanSpy, but it sucks for what i want |
| 12:18.43 | dijungal | u can't exit out of chanspy unless u end the call |
| 12:19.21 | acehunky | umm yeah thats a seperate problem |
| 12:19.28 | acehunky | mebbe anthm needs to write those routines |
| 12:19.52 | acehunky | but my problem is with Choppy Audio on VoIP trunks .. it sounds perfect when i Spy within internal extensions |
| 12:20.08 | dijungal | oooh |
| 12:20.08 | dijungal | ok |
| 12:20.14 | dijungal | yep i have the same problem |
| 12:20.29 | *** join/#asterisk eurodrew555 (n=eurodrew@83-244-164-130.cust-83.exponential-e.net) |
| 12:20.35 | acehunky | i tried to remove b --> Bridged calls |
| 12:20.38 | acehunky | but doesnt help |
| 12:20.48 | dijungal | i have an IAX (gsm) trunk between two servers and I chanspy on server 1 but the listening extension is on server2 |
| 12:20.54 | dijungal | k |
| 12:21.58 | acehunky | infact i even tried SPECIFICALLY setting the Codecs by disallowing all others |
| 12:22.09 | acehunky | to ulaw, alaw, g729, g723 |
| 12:22.20 | dijungal | k |
| 12:22.22 | acehunky | but in all the cases audio still remains choppy on VoIP trunks |
| 12:22.39 | dijungal | are you on 1.2 or 1.4 ? |
| 12:23.12 | acehunky | tried on both |
| 12:23.28 | *** join/#asterisk liri (n=liran@bzq-179-128-126.static.bezeqint.net) |
| 12:23.28 | acehunky | i wasnt able to understand difference between ChanSpy n ExtenSpy |
| 12:25.19 | *** part/#asterisk acehunky (n=acehunky@59.184.139.154) |
| 12:25.27 | *** join/#asterisk acehunky (n=acehunky@59.184.139.154) |
| 12:27.04 | acehunky | so any one who can help me with ChanSpy |
| 12:27.04 | acehunky | ? |
| 12:28.58 | destructure | acehunky: nobody can tell if they can help if they don't know what your problem is |
| 12:29.43 | destructure | oh I see, there is previous chat. (I just got here) |
| 12:29.54 | destructure | still, vague question like that.. hard to answer |
| 12:31.04 | destructure | acehunky: what are you using chanspy for |
| 12:35.09 | acehunky | hi destructure i m using it for Intercepting Agents |
| 12:35.24 | acehunky | ok give you the scenario .. |
| 12:35.41 | acehunky | Agent 1 --> 1001 |
| 12:35.41 | acehunky | Agent 2 --> 1002 |
| 12:35.41 | acehunky | Agent 3 --> 1003 |
| 12:35.52 | acehunky | they all make outbound call through voiptrunk1 |
| 12:35.55 | acehunky | over SIP |
| 12:36.06 | acehunky | and my Admin sits on Extension 1000 |
| 12:36.14 | tzafrir | Spida, a trace , your relevant config files, the model of the card, and such |
| 12:36.30 | acehunky | so when i do chanspy on 1001 (while he is talking on Voiptrunk1) i can hear very bad audio |
| 12:36.54 | acehunky | but when 1001 and 1002 are talking ... and i do chanspy on 1001 or 1002 .. i can hear crystal clear audio |
| 12:37.13 | Spida | tzafrir: how do I get a trace when the machine dies during modprobe? |
| 12:37.21 | acehunky | i tried to set Codec manually one by one .. (ulaw, alaw, g729, 723, gsm) but the situation remains the same .. |
| 12:37.41 | acehunky | destructure did u get my problem statement ? |
| 12:37.43 | tzafrir | Spida, using a serial console :-( |
| 12:37.58 | tzafrir | or take a picture of the screen |
| 12:38.20 | Spida | there is no screen. |
| 12:38.26 | Spida | just serial and ssh. |
| 12:38.29 | coppice | I sometimes get a trace of teardrops down my face when that happens |
| 12:39.49 | dandre | so as I can't find any detailed documentation about meetme, is there some good example on using it? |
| 12:40.12 | acehunky | dandre its the same case with cmd_conference .. app_conference too |
| 12:41.00 | dandre | I don't know these applications |
| 12:42.13 | tzafrir | If you have a serial console then you should have a trace |
| 12:42.35 | tzafrir | Just make sure the terminal that runs the console is logged or has a scrollback |
| 12:42.46 | tzafrir | e.g: minocom -C logfile |
| 12:43.04 | tzafrir | minicom -C logfile # that is |
| 12:44.06 | *** join/#asterisk Op3r (n=edwin@121.97.212.248) |
| 12:53.31 | destructure | acehunky: how about when 1002 is talking to voiptrunk1? |
| 13:00.51 | acehunky | destructure its the same thing .. i get choppy audio |
| 13:04.52 | *** join/#asterisk bantu (n=Miranda@rz-du-mvx-142-44.rz.uni-karlsruhe.de) |
| 13:07.24 | ai-a | how do i add the 'dial' command to the asterisk cli ? i remember seeing it there. I thought it was a version number issue, but i have v1.4.11 installed now. |
| 13:14.14 | *** join/#asterisk uwe (n=uwe@a24-163.adsl.paltel.net) |
| 13:17.30 | *** join/#asterisk Tili (n=tili@203.Red-83-53-146.dynamicIP.rima-tde.net) |
| 13:17.34 | destructure | acehunky: sounds like a timing issue, but hard to say why |
| 13:17.44 | uwe | hello, i've just tried to connect to asterisk behind a router with portforwarding/nat, i forwarded all udp on the public ip on port 5060 to the internal ip to port 5060, i connect from ekiga softphone , from another local network, so am also behind natting, it works, but the strange thing that it takes about 35-40 seconds to reach the asterisk box when dialing, and then it works fine during the call ... i know that sip natting is not straight forward |
| 13:17.45 | uwe | , but can someone give me hint where i can understand why this behaviour is happening ? |
| 13:18.36 | uwe | i mean why it works ... :) ... and with delay |
| 13:20.09 | uwe | or is it just uncommon behaviour which i should look into why it happens on the router for example ? |
| 13:20.13 | *** join/#asterisk Steven_elvisda (n=Steven_E@202.47.107.60) |
| 13:21.29 | McDouglas | if i use the dial cmd and the channel is bussy, shouldnt the processing jump to the n+100th priority command? |
| 13:22.24 | Spida | what is the recommended type of card for a isdn line (2b channel uplink to pstn, thats bri, I believe) for asterisk? |
| 13:22.39 | Tili | what is best opensource solution for sigtran |
| 13:25.51 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.50) |
| 13:26.33 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 13:28.01 | harpal | I have downloaded asterisk-1.4.13 and than try to install while doing make I got error http://pastebin.ca/764654 |
| 13:28.19 | harpal | Any one know what causes that problem? |
| 13:28.20 | *** part/#asterisk acehunky (n=acehunky@59.184.139.154) |
| 13:29.16 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 13:30.27 | *** join/#asterisk zerohalo (n=zeroHalo@pool-71-255-162-167.bstnma.east.verizon.net) |
| 13:30.48 | *** join/#asterisk robl^ (n=robl@pdpc/supporter/monthlybyte/robl) |
| 13:35.40 | *** join/#asterisk dijungal (n=kdaniel@63.175.159.171) |
| 13:40.10 | ai-a | harpal: include the full make |
| 13:40.26 | ai-a | do make clean; make and show the complete process. |
| 13:40.32 | ai-a | did you do ./configure first ? |
| 13:43.38 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 13:54.55 | *** join/#asterisk BezNalogov (n=arjan@cust-79-54.dsl.versateladsl.be) |
| 13:54.57 | harpal | ai-a, Yes I have done ./configure than make menuselect and then make |
| 13:55.21 | ai-a | and the *full* output ? |
| 13:56.06 | ai-a | from ./configure until end where it fails. |
| 13:56.50 | BezNalogov | I bought a grandstream phone gxp 2000. Before I had a linksys ata. With the ata I never had any problems. With the grandstream I use 3 accounts, after it has registered to the asterisk box, the following message appears on asterisk |
| 13:56.55 | BezNalogov | [Nov 7 14:29:18] NOTICE[10195]: chan_sip.c:15422 sip_poke_noanswer: Peer 'arjanbe' is now UNREACHABLE! Last qualify: 75 |
| 13:56.55 | BezNalogov | [Nov 7 14:29:19] NOTICE[10195]: chan_sip.c:15422 sip_poke_noanswer: Peer 'arjanuk' is now UNREACHABLE! Last qualify: 72 |
| 13:56.55 | BezNalogov | [Nov 7 14:29:20] NOTICE[10195]: chan_sip.c:15422 sip_poke_noanswer: Peer 'ispireuk' is now UNREACHABLE! Last qualify: 75 |
| 13:57.12 | BezNalogov | And then -of course- the grandstream is unreachable |
| 13:57.57 | Einsteinium | just for testing purposes, disable qualifying |
| 13:58.01 | BezNalogov | ok |
| 13:59.20 | harpal | ok I am redo that ./configure and make and than give you output |
| 13:59.30 | [TK]D-Fender | BezNalogov: describe in detail the network path between * and your phone. |
| 13:59.45 | harpal | Is it necessary to do make menuselect? |
| 14:00.00 | ai-a | harpal: never heard of menuselect. |
| 14:00.08 | ai-a | what does INSTALL file say ? |
| 14:00.11 | BezNalogov | Ok, the asterisk server is in a NOC. The phone is at my home, there is a firewall |
| 14:00.48 | ai-a | BezNalogov: have you read http://www.aocomputing.net/?p=3 ? |
| 14:01.06 | harpal | In README file they have mentioned |
| 14:01.08 | BezNalogov | In the NAT settings I have forwarded the ports 5082,5004,2070,3478,5060-5065,4569-4569 and 10000-30000 to the phone |
| 14:01.10 | Siya | _00[1N]. the (1N) should match 1-9 right? |
| 14:01.22 | [TK]D-Fender | BezNalogov: pastbin your sip.conf [general] and your phone sections masking only passwords and 1/2 the IP. |
| 14:01.24 | [TK]D-Fender | ~pb |
| 14:01.25 | jbot | A Pastebin is a place to paste your stuff without flooding 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://www.rafb.net/paste |
| 14:01.26 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^ |
| 14:01.26 | harpal | no INSTALL file |
| 14:01.39 | [TK]D-Fender | BezNalogov: you should NOT be forwarding to the phone. |
| 14:01.49 | ai-a | BezNalogov: READ http://www.aocomputing.net/?p=3 ? |
| 14:01.52 | BezNalogov | Ok, then I will remove that |
| 14:03.08 | *** join/#asterisk penguinFunk (n=penguin@unaffiliated/penguinfunk) |
| 14:03.10 | [TK]D-Fender | Siya: Yup |
| 14:03.23 | [TK]D-Fender | ai-a: ;) |
| 14:03.46 | Einsteinium | I'm starting to think my problems are branching from a crappy softphone |
| 14:04.10 | [TK]D-Fender | Einsteinium: And they would be...? |
| 14:04.26 | Einsteinium | Switched from x-lite to gizmo project, it looks like it picks up the call no matter what. (When PSTN creates incoming call, they don't hear ringing) |
| 14:04.50 | Einsteinium | sip debug showed the ringing |
| 14:05.42 | *** join/#asterisk ming_zym (n=ming_zym@123.103.29.251) |
| 14:05.46 | ai-a | Einsteinium: show complete cli debug when a call is comming in. |
| 14:05.56 | ai-a | its not the softphone you nit... |
| 14:06.06 | ai-a | if sip is ringing,, then they should hear ringing. |
| 14:06.18 | Einsteinium | the softphone is picking up the call though, before I accept |
| 14:06.29 | Einsteinium | then using its own proprietary voicemail afterwards |
| 14:06.34 | ai-a | how can you accept, if its already accepted ? |
| 14:07.04 | Einsteinium | the route is accepted (by the looks of it), but local playback and input are disabled until I "accept" |
| 14:07.15 | ai-a | SHOW your *CLI debug output. |
| 14:07.17 | Einsteinium | if I hit "ignore", it automatically pushes to the softphones voicemail |
| 14:07.19 | Einsteinium | Okay. |
| 14:07.25 | ai-a | stop guessing. |
| 14:07.26 | [TK]D-Fender | Einsteinium: Ok, then you've really misconfigured it. |
| 14:07.34 | Einsteinium | This is stock configuration |
| 14:07.43 | Einsteinium | I'd still be using X-lite, but they've broken support for Leopard |
| 14:07.51 | [TK]D-Fender | Einsteinium: you should never have enabled any of that in X-Lite. Just yank it and reinstall from scratch. easier that way :) |
| 14:07.59 | Einsteinium | Not using X-lite. |
| 14:08.01 | BezNalogov | Ok, done that http://pastebin.com/d4e130333 |
| 14:08.01 | Einsteinium | Using Gizmo |
| 14:08.15 | [TK]D-Fender | Einsteinium: Install + 3 settings = working NORMAL X-Lite |
| 14:08.24 | Einsteinium | Again, I am not using X-lite |
| 14:08.25 | ai-a | Einsteinium: as default, no phone auto answers. if you have enabled it,, then well.. what do you expect ? |
| 14:08.44 | Einsteinium | ai-a: as I just said, it's stock configuration + my sip settings |
| 14:08.46 | Einsteinium | I enabled nothing |
| 14:08.55 | ai-a | Einsteinium: right,.. |
| 14:08.59 | [TK]D-Fender | BezNalogov: Go read the guide you were linked to NOW. You're setup is not ready to work behind NAT yet |
| 14:09.00 | ai-a | as i KEEP saying.. SHOW your output. |
| 14:09.12 | [TK]D-Fender | Einsteinium: Gizmo = BLEH |
| 14:09.18 | Einsteinium | I agree |
| 14:09.25 | Einsteinium | but X-Lite won't work yet :( |
| 14:09.31 | ai-a | Einsteinium: want to see sip.conf, extentions.conf, cli output with larg debug level. |
| 14:09.33 | BezNalogov | ok thanks |
| 14:09.34 | [TK]D-Fender | ai-a: No,he said soft-phone based VM picks up, etc... its a shitty soft-phone |
| 14:09.44 | [TK]D-Fender | Einsteinium: And why not? |
| 14:09.46 | ai-a | he said it answers, and then he accepts. |
| 14:10.03 | Einsteinium | [TK]D-Fender: Google for X-Lite and MacOSX Leopard |
| 14:10.08 | ai-a | on both xlite and gimzo. |
| 14:10.12 | Einsteinium | It's not compatible yet |
| 14:10.14 | Einsteinium | NOT ON X-LITE |
| 14:10.21 | [TK]D-Fender | Einsteinium: Google for something OTHER than Gizmo :) |
| 14:10.46 | [TK]D-Fender | Einsteinium: http://www.zoiper.com/zmac.php |
| 14:10.48 | Einsteinium | [TK]D-Fender: I wish there were more clients for Mac, but there aren't a lot of options |
| 14:10.50 | BezNalogov | Ok, that is clear |
| 14:10.57 | BezNalogov | Thanks |
| 14:11.06 | [TK]D-Fender | BezNalogov: np |
| 14:11.27 | BezNalogov | Wait, one question |
| 14:11.38 | BezNalogov | My asterisk server has a public address |
| 14:11.50 | BezNalogov | eth0 Link encap:Ethernet HWaddr 00:30:48:42:6F:FE |
| 14:11.50 | BezNalogov | <PROTECTED> |
| 14:12.05 | ai-a | then you dont need to forward any ports on any side. |
| 14:12.13 | ai-a | you might need your iptables set up. |
| 14:12.14 | BezNalogov | Should I fill in that then at both the WAN and internal network address? |
| 14:12.45 | *** join/#asterisk michael-i (n=michael-@W8047.w.pppool.de) |
| 14:13.02 | *** join/#asterisk zeppelin_ (n=zeppelin@189.32.104.12) |
| 14:13.08 | ai-a | also, shouldnt put asterisk on direct internet,, use a proxy to prevent pw crack attempts on your sip accounts.. |
| 14:13.12 | zeppelin_ | ~book |
| 14:13.12 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 14:13.16 | *** join/#asterisk IPetrov2 (i=IPetrov2@ppp91-76-141-110.pppoe.mtu-net.ru) |
| 14:13.48 | ai-a | OpenSER |
| 14:14.28 | [TK]D-Fender | BezNalogov: Here : http://pastebin.com/m5218229 |
| 14:19.01 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:19.25 | Einsteinium | [TK]D-Fender: Doesn't look like Voiper is compatible with Leopard yet either |
| 14:19.28 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 14:19.52 | [TK]D-Fender | Einsteinium: no compatability mode in there? |
| 14:19.57 | Einsteinium | not yet :( |
| 14:20.03 | [TK]D-Fender | Einsteinium: So basically Apple jsut "broke everything" again? |
| 14:20.04 | *** join/#asterisk Naeem (n=chatzill@62.240.47.161) |
| 14:20.05 | Einsteinium | giving same error as X-Lite is |
| 14:20.18 | *** join/#asterisk qdk (n=qdk@85.235.253.139) |
| 14:20.24 | Einsteinium | Apples conforming to the protocol, apparently those clients were not |
| 14:20.26 | Einsteinium | some were not affected |
| 14:20.32 | [TK]D-Fender | Einsteinium: when you go "bleeding edge", expect to get cut. |
| 14:20.46 | Einsteinium | bleeding edge would have been the beta :P |
| 14:22.02 | BezNalogov | Ok, ai-a, I changed my sip.conf now, reloaded it, rebooted the phone |
| 14:22.14 | [TK]D-Fender | BezNalogov: What router is it behind? |
| 14:22.17 | BezNalogov | But after a minute all accounts are again unreachable |
| 14:22.33 | BezNalogov | Speedtouch 706 |
| 14:22.34 | dijungal | is there a way to have 2 or more asterisk servers writing to the same queue.log and csv file ? |
| 14:22.44 | ai-a | BezNalogov: put sip debugging on,, and capture the output of the sip messages.. |
| 14:22.50 | ai-a | also sip debuggin on the sip phone if it allows it. |
| 14:23.02 | ai-a | i guess its giving an internal IP address for the softphone. |
| 14:23.10 | ai-a | have you put NAT = yes, in the sip.conf of the sipphone ? |
| 14:23.33 | BezNalogov | Yes I have |
| 14:24.10 | BezNalogov | arjanbe/arjanbe 82.174.54.79 D N 49698 UNREACHABLE |
| 14:24.18 | [TK]D-Fender | BezNalogov: And have you modified your sip.conf as per my pastebin? |
| 14:24.22 | BezNalogov | The strange thing is that in sip show peers, the NAT setting is N |
| 14:24.24 | BezNalogov | Yes I have |
| 14:24.34 | Einsteinium | do you have firewall log access? |
| 14:24.56 | [TK]D-Fender | BezNalogov: pastebin it again.... and if you have a users.conf, "mv" it out of the way |
| 14:25.25 | harpal | ai-a, here is output of from ./configure http://pastebin.ca/764704 |
| 14:25.51 | harpal | It also include error and whole output of make |
| 14:25.59 | BezNalogov | http://pastebin.com/d2b50d2ae |
| 14:26.13 | BezNalogov | Any useful information there about what might be wrong? |
| 14:26.47 | robl^ | anyone know the max length for a SIP secret in Asterisk? |
| 14:26.54 | rantsh | hi all |
| 14:27.37 | rantsh | listen this might be more of a linux question than anything, but anyone knows how to get the lenght of an mp3 without playing it? using sox maybe? |
| 14:28.05 | coppice | robl^: if we told you it wouldn't be a secret, would it? |
| 14:28.33 | robl^ | coppice: I promise not to tell anyone. ;-) |
| 14:28.53 | coppice | everyone says that on the internet |
| 14:29.11 | [TK]D-Fender | BezNalogov: it could be your router. If you've turned off all forwarding on it I'm not sure what to suggest. Check your server's firewall as well. |
| 14:29.14 | Qwell | play it. got a stop watch? |
| 14:29.15 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.50) |
| 14:29.41 | *** join/#asterisk felix_da_catz (n=felix@PixNationalGroup.phonoscope.com) |
| 14:30.00 | [TK]D-Fender | rantsh: In most OS GUI's you can get properties on files like that.. |
| 14:30.08 | McDouglas | i have some trouble with this voice menu: http://pastebin.com/d7c17d677 |
| 14:30.39 | *** join/#asterisk l2trace99 (n=asd@75.112.133.235) |
| 14:30.47 | McDouglas | why does it wait about 5 seconds after i press the 2 button on lin 12 ? |
| 14:30.48 | McDouglas | *line |
| 14:31.00 | dijungal | is there a way to have 2 or more asterisk servers writing to the same queue.log and csv file ? |
| 14:31.12 | rantsh | [TK]D-Fender, yup, but I need to run a batch and store the filename of a recorded queue call (transcoded to mp3) with its length in a database table, |
| 14:31.30 | harpal | ai-a, Have you find anything why I am getting that error? |
| 14:31.37 | [TK]D-Fender | dijungal: Never lets 2 systems figth for rights over a file. You CAN however allow them to share a DB |
| 14:31.46 | ai-a | harpal: no idea.. what about an older version ? does that compile ? |
| 14:31.53 | dijungal | k |
| 14:32.04 | [TK]D-Fender | rantsh: Keep looking for trouble, and you're likely to find it :) |
| 14:32.09 | dijungal | so i store the queue data in a DB and let two systems write to that table |
| 14:32.27 | rantsh | : ) |
| 14:32.36 | [TK]D-Fender | dijungal: Sure. DB lock handling will let that work out normally. |
| 14:33.01 | BezNalogov | http://pastebin.com/m7adc5e1b |
| 14:33.07 | BezNalogov | There is the debug output |
| 14:33.08 | CyberSyx^ | <PROTECTED> |
| 14:33.18 | CyberSyx^ | but the "s" extension where is ? A: |
| 14:33.36 | dijungal | ok i'll have to do that |
| 14:33.38 | harpal | ai-a, which older version? |
| 14:33.47 | harpal | I have astrisk 1.4.13 |
| 14:33.50 | BezNalogov | If there is nothing special there, I will do a hard reset on this router, maybe it's having troubles again. I had problems with it before |
| 14:33.54 | ai-a | so, try 1.4.11 |
| 14:34.10 | dijungal | next question... hos do i store the queue data in a mysql table? |
| 14:34.19 | [TK]D-Fender | BezNalogov: ouch. could be that your router is just nasty with NAT |
| 14:34.28 | [TK]D-Fender | dijungal: go read... THE BOOK |
| 14:34.30 | [TK]D-Fender | ~book |
| 14:34.30 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 14:34.31 | datachomper | Custom katana, lol |
| 14:34.42 | harpal | ai-a, Is it not stable? |
| 14:34.52 | *** join/#asterisk shafique (n=shafique@203.215.176.186) |
| 14:35.05 | harpal | ok just try that 1.4.11 |
| 14:35.07 | ai-a | harpal: i installed 1.4.11 few days ago. 1.4.13 is newer this week. |
| 14:35.07 | datachomper | Is it Hanzo steel? |
| 14:35.25 | harpal | ok. let me try to install it |
| 14:35.44 | BezNalogov | This router isn't only nasty with NAT, it is a preconfigured router from my ISP for internet and telephony |
| 14:36.22 | shafique | hi |
| 14:36.38 | dijungal | thnks |
| 14:36.49 | [TK]D-Fender | datachomper: based on this one : http://www.roninswords.com/custom%20horse%20and%20plum.htm |
| 14:37.01 | dandre | I am trying to use the manager UpdateConfig but I can't understand how to format Match-XXXXXX. |
| 14:37.27 | dandre | I want to use some wildcard match, is this possible? |
| 14:37.36 | *** join/#asterisk DarkFlib (n=root@host90-152-23-30.ipv4.regusnet.com) |
| 14:37.48 | CyberSyx^ | [TK]D-Fender, excuse me but where do i view the "default" context if i write extensions.conf from 0 ? |
| 14:38.17 | dandre | for instance, match all line whose value starts with some pattern |
| 14:38.36 | [TK]D-Fender | CyberSyx^: ...huh? what "default" context? There is no such thing. the only 2 fixed name contexts are [globals] and [general] |
| 14:39.26 | CyberSyx^ | [TK]D-Fender, looks this: http://pastebin.com/m736e44b3 |
| 14:40.18 | [TK]D-Fender | CyberSyx^: Thats just a place * tries to fall back because you didn't create the context for incoming calls you TOLD IT you were using. |
| 14:40.19 | DarkFlib | got a quick issue... we were using 1.4.11 with odbc backed voicemail, (peers reg'd with OpenSER) and every so often we keep getting callerid wedged (it comes from a fixed number that bears little resemblance to the callerid in the sip packet) and is only cleared by a restart |
| 14:40.30 | CyberSyx^ | [TK]D-Fender, |
| 14:40.40 | CyberSyx^ | the context for incoming calls is this... |
| 14:40.42 | CyberSyx^ | whait |
| 14:40.44 | DarkFlib | we upgraded to 1.4.13 this morning and its still occuring |
| 14:40.55 | [TK]D-Fender | CyberSyx^: == Starting Mobile/n73-ef38 at bluetooth,s,1 failed so falling back to exten 's' <-- its trying to land on "bluetooth,s,1" and it doesn't exist. So why not? |
| 14:40.59 | DarkFlib | any ideas? |
| 14:41.30 | CyberSyx^ | wait |
| 14:42.04 | CyberSyx^ | [TK]D-Fender, look this: http://pastebin.com/m736e44b3 |
| 14:43.13 | [TK]D-Fender | CyberSyx^: YOU look at it! Its telling you EXACTLY what context & exten its expecting. Get off your ass and MAKE IT! |
| 14:43.31 | *** join/#asterisk irule (n=irule@200.53.61.4) |
| 14:43.42 | CyberSyx^ | ... |
| 14:43.55 | Einsteinium | Show SIP.conf |
| 14:44.19 | DarkFlib | hmmm... it also looks like we have no database traffic from our func_odbc stuff... |
| 14:44.34 | [TK]D-Fender | CyberSyx^: Seriously its basically telling you "Push the big red button", and you're going around asking "what does it want me to do, I don't understand!?" |
| 14:45.12 | DarkFlib | [TK]D-Fender: the big red button with the words 'Push me' on it? :P |
| 14:46.46 | *** join/#asterisk harpal (n=Harpal@124.125.254.227) |
| 14:47.27 | McDouglas | anyone could help with this dialplan? http://pastebin.com/d7c17d677 if i press button 2 on line 11 while the sound file is still palying it waits 5 seconds, and only after that jumps to line 2 |
| 14:47.34 | *** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-a9fe3df50482cc44) |
| 14:50.43 | [TK]D-Fender | McDouglas: Your "menu" is really messed up |
| 14:50.56 | [TK]D-Fender | McDouglas: Why multiple waits? |
| 14:51.00 | McDouglas | [TK]D-Fender: its jsut a test, but i'm open for suggestions |
| 14:51.13 | [TK]D-Fender | McDouglas: And your "goto" is leading to a dead end |
| 14:51.19 | McDouglas | [TK]D-Fender: the first wait waits for direct extension, and after that i wait for a menu selection |
| 14:51.52 | McDouglas | [TK]D-Fender: why? if i rpess button 2 it does go to the menu-szerviz context |
| 14:51.53 | [TK]D-Fender | McDouglas: That idea is not valid. Remove it and just play your prompts back to back. |
| 14:52.13 | [TK]D-Fender | you can press 2 at ANY TIME. |
| 14:52.43 | *** join/#asterisk socken23 (n=socken@ip-213-189-154-029.fix.magnet.ch) |
| 14:52.59 | *** join/#asterisk fskrotzki (n=fskrot@host198.textwise.com) |
| 14:53.03 | socken23 | I don't have any voice prompts any more! :-( Any idea where I should start looking?? |
| 14:53.17 | dandre | I want to delete from manager a conference room without knowing the PIN and the admin code. How should I do? |
| 14:53.32 | McDouglas | [TK]D-Fender i want to wait a bit longer after telling the caller that he can dial a direxct extension before start listing the menu |
| 14:54.12 | [TK]D-Fender | McDouglas: makes no difference if you wait to TELL them the options, they can DO it at any time, so sperates waitextens is pointless. |
| 14:54.24 | [TK]D-Fender | separate* |
| 14:54.53 | De_Mon | [TK]D-Fender he still wants to add a delay before playing the 2nd voice prompt |
| 14:54.55 | McDouglas | i know they can do it any time, but i dont want to confuse them by start listing the menu |
| 14:55.26 | [TK]D-Fender | De_Mon: thats what Background is for.... playing back stuff while accepting input. |
| 14:55.43 | [TK]D-Fender | McDouglas: So Background a few seconds of silence! |
| 14:55.52 | McDouglas | well,t hats an option |
| 14:55.56 | *** join/#asterisk NovceGuru (i=shelby@ballmung.easymac.org) |
| 14:56.08 | De_Mon | whats the difference between WaitExten(.5) and Background(500ms) on thats right one doesnt exist |
| 14:56.14 | McDouglas | still, no idea why the delay after the 2nd background |
| 14:56.56 | *** join/#asterisk acunningham (n=acunning@nic06-3-88-173-73-90.fbx.proxad.net) |
| 14:56.58 | [TK]D-Fender | McDouglas: Go fix it and add a realistic delay. |
| 14:57.50 | acunningham | Can someone running 1.4.13, please run "show channels" twice from a console connected via "asterisk -r", and tell me if the second one works? |
| 14:58.06 | *** join/#asterisk irule (n=irule@200.53.61.4) |
| 14:58.27 | Einsteinium | Yes it does |
| 14:58.33 | acunningham | Hmm. |
| 14:58.42 | acunningham | For me, the second run produces no output |
| 14:59.06 | acunningham | If you connect to the manager interface, and do: |
| 14:59.11 | acunningham | action: command |
| 14:59.14 | acunningham | command: show channels |
| 14:59.17 | acunningham | Does that work? |
| 14:59.22 | McDouglas | [TK]D-Fender: modified to this: http://pastebin.com/d5851b867 still got a huge delay after pressing button 2 |
| 14:59.24 | harpal | ai-a, That works fine. I have installed asterisk 1.4.11. |
| 14:59.51 | [TK]D-Fender | McDouglas: WORSE. UGH |
| 15:00.45 | [TK]D-Fender | McDouglas: "show function TIMEOUT" , learn to ANSWER you calls explicitly, set timeouts, use WaitExten where appropriate or use "s" like you're supopsed to with "autofallthrough=no", etc... |
| 15:00.54 | *** join/#asterisk Runlvl (n=gustavo@190.2.40.57) |
| 15:00.55 | Runlvl | <PROTECTED> |
| 15:01.19 | ai-a | harpal: do a diff. |
| 15:01.22 | *** join/#asterisk freezey (n=freezey@maher.mercy.edu) |
| 15:01.30 | *** join/#asterisk ManxPower (n=manxpowe@247.sub-70-223-210.myvzw.com) |
| 15:01.34 | harpal | ok |
| 15:01.51 | Runlvl | hehe :) |
| 15:01.59 | irule | Runlvl do you also irc somewhere else? |
| 15:02.15 | Runlvl | irule, yes, we are using #asterisk-es |
| 15:02.29 | JerJer | anyone good with PHP and want to deal with a really annoying bit of PHP code? :) |
| 15:02.36 | harpal | ai-a, there are many differences |
| 15:02.46 | JerJer | http://www.pastebin.ca/764736 |
| 15:02.48 | freezey | i am wondering what the soft fax reciever for asterisk is... is the conf file already in asterisk or do i need to grab the seperate package from somewhere? |
| 15:03.41 | [TK]D-Fender | freezey: Google up "spandsp" |
| 15:03.53 | [TK]D-Fender | freezey: And yes it is 3rd party, not part of * |
| 15:04.16 | coppice | its GPL, so it can't going into * |
| 15:04.36 | harpal | ai-a, now how to teste that? I am new to this. i done know more about that. I have downloaded asterisk book, but is there another way to do that and I start to use that? |
| 15:05.20 | *** join/#asterisk qdk (n=qdk@85.235.253.139) |
| 15:05.22 | freezey | [TK]D-Fender: another questions is i am running what i guess is an older version of asterisk... non gui.. is their a webbased app that can handle asterisk? |
| 15:05.54 | [TK]D-Fender | freezey: All sorts of GUI's are listed in the channel topic. Use any and this channel is dead to you. |
| 15:06.10 | McDouglas | [TK]D-Fender: what do you mean by "use "s" like you're supopsed to with "autofallthrough=no"" ? |
| 15:06.57 | freezey | [TK]D-Fender: yeah not using one now was just curious... for my faxing i am using hylafax and its just not working how it should be |
| 15:07.24 | [TK]D-Fender | McDouglas: Go read THE BOOK, and learn about Asterisk Standard Extensions. |
| 15:07.25 | ai-a | harpal: you have two folders asterisk1.14.11 and 1.14.13 do diff -r ...11 ...13 |
| 15:07.32 | ai-a | then paste that into pastebin :) |
| 15:07.36 | McDouglas | [TK]D-Fender: i already read the book |
| 15:07.37 | coppice | freezey: in what way is hylafax not working? |
| 15:07.47 | [TK]D-Fender | McDouglas: then try this : |
| 15:07.49 | [TK]D-Fender | ~osmosis |
| 15:07.50 | jbot | i guess osmosis is the act of beating yourself on the head repeatedly with THE BOOK, until some measure of absorption has occured ... or at least until your unconsciousness restores peace to the channel ... |
| 15:07.55 | [TK]D-Fender | :D |
| 15:08.00 | freezey | coppice: it seems that the extensions i had recently setup in FaxDispatch were at one point working and now they just endlessly ring but sometimes u can get the faxline but rarely |
| 15:08.11 | freezey | coppice: like i am calling the faxline right now and its just endlessly rining |
| 15:08.11 | McDouglas | thanks. |
| 15:08.24 | [TK]D-Fender | McDouglas: Go try and follow the rest of what I advised you to look at |
| 15:08.48 | *** join/#asterisk twoshadetod (n=twoshade@c-76-123-96-239.hsd1.fl.comcast.net) |
| 15:08.48 | ManxPower | freezey: You are not helpless. Asterisk empowers you to take charge of your life and do some basic troubleshooting. |
| 15:09.00 | twoshadetod | anyone here use the polycom 501? |
| 15:09.11 | ManxPower | twoshadetod: only about 1/2 the channel. |
| 15:09.22 | ai-a | harpal: maybe not ;) |
| 15:09.35 | McDouglas | i am reading it. i suppose my main problem comes from the digit timout. But still, i know what is the s extension, and thats wgy i dont understand why do i have to use it? |
| 15:09.43 | freezey | ManxPower: ha well i got asterisk running pretty well it handles about 3000 extensions and departments so i wanted to use something seperate for faxing |
| 15:09.45 | twoshadetod | Can it function as a standalone, without asterisk jjust with a "direct line" to a sipprovider? |
| 15:10.00 | freezey | coppice: you ever run into a situation like this? |
| 15:10.02 | ManxPower | freezey: Are the fax calls going thru Asterisk? |
| 15:10.14 | [TK]D-Fender | McDouglas: That was only a small PART of what I told you to look at. |
| 15:10.24 | ManxPower | freezey: I'm pretty sure you can troubleshoot it a bit more than "it just rings" |
| 15:10.40 | [TK]D-Fender | twoshadetod: Its a SIP phone and can connect to an ITSP jsut like any other SIP device. |
| 15:11.04 | twoshadetod | [TK]D-Fender, awsome thats what i was thinking /hoping |
| 15:11.17 | ManxPower | McDouglas: the only times you use the "s" extension is with immediate=yes, or if the call comes in an FXO port without DID |
| 15:11.18 | freezey | ManxPower: well see the problem is is that somestimes i get the fax line other times i dont... and i grepped the entire asterisk dir for the extensions that were made in hylafax and nothing comes up so i think those extensions go through my PBX or my AVAYA switch |
| 15:11.33 | [TK]D-Fender | ManxPower: We're talking IVR's here |
| 15:11.35 | [TK]D-Fender | ~stdextens |
| 15:11.36 | jbot | stdextens is probably "s" Standard Extension : Where a call goes to when * does not know the destination of the call. Ex : Calls coming in on FXO ports (no DID), a call coming in from an ITSP that doesn't specify or where it was not set in the REGISTER line, or FXS port goes off-hook and "immediate=yes" in zapata.conf. "s" is also used to make IVRs & macros. The ... |
| 15:11.52 | ManxPower | freezey: if you don't know where it rings you are far, far beyong our help. |
| 15:12.19 | freezey | ManxPower: let me find out where it rings from... and i will get back to you.. |
| 15:12.29 | ManxPower | [TK]D-Fender: Ah. I work on finishing my first cup of coffee then. |
| 15:12.40 | McDouglas | [TK]D-Fender: well, i use did, so i know the call will come in on extension 16 |
| 15:12.41 | [TK]D-Fender | ManxPower: Yuo do that :) |
| 15:12.50 | *** join/#asterisk cpina (n=carles@ip23498.bcn.altecom.net) |
| 15:12.59 | cpina | hello! |
| 15:13.06 | [TK]D-Fender | McDouglas: Ok, get off the topic of "s" and follow the REST like I've told you repeatedly. |
| 15:13.26 | cpina | one short question: after load ztdummy module, background doesn't play any sounds (silence, without error message). How we can use ztdummy and background on same time? |
| 15:13.51 | ManxPower | cpina: why do you need ztdummy? |
| 15:14.00 | cpina | ManxPower: for meetme |
| 15:14.10 | cpina | (without hardware, in this computer) |
| 15:14.20 | cpina | (specific hardware i mean :-) ) |
| 15:14.40 | ManxPower | cpina: there have been various reports recently about ztdummy causing that problem. |
| 15:14.53 | harpal | ai-a, here it is http://pastebin.ca/764753 |
| 15:15.07 | ManxPower | no fixes that I remember, except for one person fixed the issue by changing the kernel HZ to 1000 from 250Hz |
| 15:15.18 | cpina | ManxPower: well, we are using quite old asterisk :-) so should not be recent... |
| 15:15.19 | cpina | ah ok |
| 15:15.21 | cpina | i take note |
| 15:15.28 | cpina | (1.2.14 asterisk version) |
| 15:15.39 | [TK]D-Fender | cpina: ANCIENT |
| 15:15.43 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 15:15.46 | ManxPower | cpina: well you know the first thing you must do then. |
| 15:16.01 | *** join/#asterisk slowshutt (n=d@196.211.34.2) |
| 15:16.02 | ManxPower | cpina: when did the problem start? |
| 15:16.21 | cpina | ManxPower: now, setting up this new Asterisk server |
| 15:16.29 | socken23 | Mhh... I still don't get any prompts! |
| 15:16.43 | Katty | mew? |
| 15:16.45 | socken23 | I see all the logs in CLI, but don't hear anything.. |
| 15:16.56 | ManxPower | cpina: at least upgrade to the latest 1.2.x zaptel and asterisk |
| 15:16.58 | [TK]D-Fender | socken23: And you've still shown us nothing of value! But at least there is symmetry... |
| 15:17.07 | cpina | ManxPower: yes, i will do it now :-) |
| 15:17.18 | cpina | and report here if it worked :-) |
| 15:17.22 | socken23 | [TK]D-Fender: Can you give me a hint where to start looking? |
| 15:17.32 | [TK]D-Fender | socken23: www.pastebin.com |
| 15:17.42 | McDouglas | [TK]D-Fender: exten => 16,1,Set(TIMEOUT(digit)=1) solved my problem, thanks |
| 15:17.54 | socken23 | [TK]D-Fender: Yeah thanks, but what do you need to see? |
| 15:18.04 | socken23 | I'll paste some output of the CLI... |
| 15:18.14 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.50) |
| 15:18.19 | [TK]D-Fender | socken23: how about the complete CLI output of your failed call at verbose 10 & channel debug enabled. |
| 15:18.23 | freezey | ManxPower: it goes through the PBX |
| 15:18.25 | ManxPower | McDouglas: I doubt that actually solved your problem. |
| 15:18.36 | [TK]D-Fender | McDouglas: pastebin where you've gotten yourself to so far. |
| 15:18.42 | ManxPower | freezey: then there is nothing we can do to help you, since it does not go thru asterisk |
| 15:19.28 | harpal | ai-a, Where I can start so that It starts working and than I can start changing options and check that? |
| 15:19.46 | ai-a | harpal: .13 compiles for me. |
| 15:19.47 | Katty | mew :< |
| 15:19.48 | McDouglas | [TK]D-Fender: http://pastebin.com/d6383f77c |
| 15:19.57 | [TK]D-Fender | Katty: Mew. |
| 15:20.00 | *** join/#asterisk hfb (n=hfb@pool-71-106-219-180.lsanca.dsl-w.verizon.net) |
| 15:20.01 | Katty | mew :> |
| 15:20.01 | ai-a | Hey Katty :) |
| 15:20.26 | socken23 | [TK]D-Fender: Here we go: http://pastebin.com/m616f40fd |
| 15:20.41 | *** join/#asterisk docelmo (n=vircuser@c-76-99-153-242.hsd1.de.comcast.net) |
| 15:21.04 | harpal | oh greats, than whats problem in my system? |
| 15:21.25 | xpot | can anyone help with the following error? --> [Nov 6 22:36:57] ERROR[14120]: pbx.c:1527 ast_func_read: Function ODBC_CARDNUM not registered |
| 15:21.27 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 15:21.40 | xpot | using freetds to connect to MSSQL server, both unixODBC and FreeTDS are installed, when I run "make menuselect" XXX 4. cdr_odbc, [*] 8. cdr_tds |
| 15:22.13 | [TK]D-Fender | McDouglas: http://pastebin.com/m56bc03c8 <- better |
| 15:22.21 | *** join/#asterisk Naeem (n=chatzill@62.240.47.161) |
| 15:22.28 | slowshutt | asterisk voicemail uses sendmail to send voicemail notifications, mine was working fine until a few days ago |
| 15:22.48 | [TK]D-Fender | socken23: you don't here it asking for a password? |
| 15:22.48 | McDouglas | [TK]D-Fender: somewhere i read that i dont have to sue answer() because background and palyback does that automaticaly |
| 15:23.06 | McDouglas | *use |
| 15:23.08 | *** join/#asterisk Naeem (n=chatzill@62.240.47.161) |
| 15:23.09 | [TK]D-Fender | McDouglas: Never trust stuff like that. If its worth doing, its worth doing YOURSELF |
| 15:23.10 | socken23 | [TK]D-Fender: No! that's my problem ;-) |
| 15:23.26 | [TK]D-Fender | socken23: And where is this SIP device relative to *? |
| 15:23.26 | Kobaz | yeap |
| 15:23.27 | Kobaz | er |
| 15:23.54 | socken23 | directly connected |
| 15:23.58 | slowshutt | hi there need to connect to a telfree server anyone tried it before? |
| 15:24.01 | ManxPower | McDouglas: hopefully you READ THAT IN THE CLI when you did a "show application background" and "show application playback" |
| 15:24.13 | socken23 | [TK]D-Fender: when I enter my password, I get ... Playing 'vm-youhave' (language 'de') |
| 15:24.42 | *** join/#asterisk eurodrew555 (n=eurodrew@83-244-164-130.cust-83.exponential-e.net) |
| 15:24.45 | ai-a | socken23: you getting one way audio ? |
| 15:24.53 | socken23 | [TK]D-Fender: Calling another sip client also works (with audio)... |
| 15:24.55 | slowshutt | i can connect using the username and password supplied by telfree using the xlite softphone, but can not seem to get it to work with asterisk |
| 15:24.59 | slowshutt | any help welcome |
| 15:25.04 | [TK]D-Fender | socken23: Try using english to confirm that its all prompts, and not language specific |
| 15:25.07 | harpal | ai-a, whats causing problem to install 1.4.13? have you any idea? |
| 15:25.13 | socken23 | [TK]D-Fender: good idea, let me try that |
| 15:25.19 | ai-a | harpal: nope. but if you find out we wou dlike to know ;) |
| 15:25.27 | Katty | [TK]D-Fender: so i fought with that server for 3 weeks. |
| 15:25.28 | harpal | ok sure |
| 15:25.31 | Katty | [TK]D-Fender: and now it finally works. |
| 15:25.36 | Katty | [TK]D-Fender: except, no one seems to care... |
| 15:25.39 | [TK]D-Fender | Katty: Yee-haw |
| 15:25.44 | *** join/#asterisk grtrfgthy (n=dddddd@123-2-65-142.static.dsl.dodo.com.au) |
| 15:25.46 | [TK]D-Fender | Katty: \o/ |
| 15:25.54 | Katty | [TK]D-Fender: that makes me all sad inside. |
| 15:25.56 | harpal | ai-a, where can I start? so that I can just test |
| 15:26.09 | ai-a | harpal: open the failing .c file, and find the error. |
| 15:26.10 | socken23 | [TK]D-Fender: Mhh.. same problem Playing 'vm-password' (language 'en') .. -> No audio |
| 15:26.11 | [TK]D-Fender | Katty: Eviscerate them then :) |
| 15:26.18 | Katty | [TK]D-Fender: maybe i'll just quit. |
| 15:26.21 | ai-a | harpal: ill have a play. |
| 15:26.29 | socken23 | ai-a: Mhh.. one-way audio? It's a voicemail.. I don't hear the voicemail... |
| 15:26.32 | grtrfgthy | is there a a way to have sound file speedup/slowdown in playback (for long messages) |
| 15:26.41 | [TK]D-Fender | socken23: while in a call pastebin the output of "sip show channels" |
| 15:26.42 | ManxPower | socken23: stop asterisk, rmmod ztdummy, start asterisk. is the problem still there? |
| 15:27.13 | [TK]D-Fender | grtrfgthy: in voicemailmain, yes. |
| 15:27.20 | socken23 | ManxPower: rmmod ztdummy: Module ztdummy does not exist in /proc/modules |
| 15:27.36 | ManxPower | socken23: it is not listed in "lsmod" either? |
| 15:27.50 | socken23 | nope. I do have a zaptel card though |
| 15:27.55 | ManxPower | socken23: Oh! |
| 15:27.58 | socken23 | I have zaptel loaded |
| 15:28.10 | socken23 | altough I think my problem is related to sip... |
| 15:28.28 | ManxPower | if you have a zaptel card installed and the drivers loaded, but it is not configured or there is line connected to it, that can cause audio problems on ALL channel types, espectially VoIP ones. |
| 15:28.42 | socken23 | Aha! Interesting! |
| 15:28.42 | harpal | ai-a, means? how to test that? |
| 15:28.43 | ai-a | harpal: has #ifdef DEBUG_THREADS around that error line. |
| 15:28.46 | ManxPower | socken23: rmmod all the zaptel stuff, then start asterisk and see if that fixes it. |
| 15:28.52 | socken23 | I do have some problems with the external channels |
| 15:28.55 | socken23 | let me try that |
| 15:29.17 | ManxPower | I HATE COLD WEATHER! |
| 15:29.26 | harpal | #ifdef DEBUG_THREADS |
| 15:29.26 | harpal | <PROTECTED> |
| 15:29.26 | harpal | #endif |
| 15:29.38 | socken23 | ManxPower: You're my hero!! That does fix the problem!!!! |
| 15:29.38 | harpal | This ast_cli line I getting error |
| 15:29.43 | *** join/#asterisk docelmo (n=vircuser@c-76-99-153-242.hsd1.de.comcast.net) |
| 15:29.53 | ManxPower | socken23: I've been doing this stuff for a long, long time. |
| 15:29.57 | socken23 | ManxPower: Didn't occur to me that problems with zaptel might inflict with the other channels |
| 15:30.06 | ai-a | harpal: whats what i said. |
| 15:30.14 | ai-a | why is DEBUG_THREADS defined in your make ? |
| 15:30.15 | socken23 | ManxPower: Looks like! So I'll start looking with zaptel problems |
| 15:30.21 | socken23 | ManxPower: Thanks for the tip!! |
| 15:30.37 | datachomper | errr body in the club gettin tips |
| 15:30.39 | ai-a | harpal: was it your that did make menu<soemthing> ? |
| 15:30.42 | ManxPower | socken23: chances are Asterisk saw the card, and so used the sync timing from the card to time RTP packets, but the card was not generating interrupts. |
| 15:30.54 | harpal | ai-a, yes |
| 15:31.05 | ai-a | harpal: then that has messed the install. |
| 15:31.12 | ai-a | delete the folder, download / untar it again. |
| 15:31.19 | harpal | but I have taken that output without aplying that command |
| 15:31.27 | harpal | okok |
| 15:31.43 | socken23 | ManxPower: Ah ok. You really do know your stuff ;-) |
| 15:31.46 | harpal | I have 1.4.11 installed. should I remove that? |
| 15:31.54 | harpal | how to remove that? |
| 15:31.58 | ManxPower | socken23: it has not made me rich. |
| 15:32.15 | ManxPower | harpal: are you upgrading odowngrading from 1.4.11? |
| 15:32.26 | ai-a | harpal: a make clean might fix it :) |
| 15:32.57 | ai-a | ManxPower: he did make menuselect which seemed to have messed it up. |
| 15:32.59 | harpal | If I do make clean than it removes that 1.4.11 version? |
| 15:33.30 | harpal | ai-a, I have 1.4.11 installed that how to install that 1.4.13? |
| 15:33.33 | ai-a | nope, its in a different folder. |
| 15:33.46 | ai-a | you need to uninstall it then. |
| 15:33.53 | Siya | [TK]D-Fender: tx (was a while back... 2hr conf call *pfew*) |
| 15:33.57 | ManxPower | harpal: just install 1.4.13 over 1.4.11 |
| 15:33.59 | ai-a | but a make install on .13 should upgrade it automaticly. |
| 15:34.07 | harpal | ok |
| 15:34.23 | *** join/#asterisk mrtelephone (n=MrTeleph@h697179-171.picriverisp.net) |
| 15:34.36 | ManxPower | If you were going between MAJOR versions 1.2.x to/from 1.4.x it *could* be a bit more complicated. |
| 15:34.47 | mrtelephone | CISCO 7960's send out sip messages on a random port and expect the message back on 5060 :( |
| 15:35.07 | mrtelephone | very interesting |
| 15:35.11 | ManxPower | mrtelephone: if it did, then they would not work. |
| 15:35.18 | mrtelephone | thats why I couldn't get it working |
| 15:35.30 | [TK]D-Fender | mrtelephone: Go run Novell 5.0 while you're at it then for full protocol wackiness :p |
| 15:35.36 | mrtelephone | asterisk was trying to send back on the port it came from |
| 15:35.44 | ManxPower | hundreds of people (including myself) had 79xx phones working just fine with Asterisk |
| 15:35.50 | mrtelephone | on a nat? |
| 15:35.56 | robl^ | yup |
| 15:36.14 | mrtelephone | cisco -> nat -> internet <- asterisk |
| 15:36.23 | ManxPower | mrtelephone: there are two things at work. SIP (port 5060 for the destination of the packet, random for source of packet) and RTP (the ports used are decided during the call setup in SIP) |
| 15:36.46 | ManxPower | mrtelephone: not cisco -> nat -> internet -> nat -> asterisk? |
| 15:36.57 | mrtelephone | can't you tell the cisco to accept sip packets on the port its sending from? |
| 15:37.11 | robl^ | I used to have 5 Cisco 7960s on a NAT .. 5x phones -> NAT -> Intenets <- Asterisk (public IP) |
| 15:37.22 | asteriskonsuse | anyone bri working in asterisk-gui? |
| 15:37.24 | mrtelephone | what was your config then? |
| 15:37.26 | asteriskonsuse | anyone bri working in asterisk-gui? |
| 15:37.35 | ManxPower | mrtelephone: you didn't do something stupid like enable NAT support on the phone did you? |
| 15:37.41 | mrtelephone | robl, were you forwarding the sip port to the phones on the router? |
| 15:37.50 | ManxPower | asteriskonsuse: nobody uses guis here. try a gui channel |
| 15:37.51 | mrtelephone | manxpower, yes |
| 15:38.05 | ManxPower | mrtelephone: that breaks asterisks nat support. don't do that. |
| 15:38.22 | ManxPower | phone nat support + asterisk nat support = no nat support. |
| 15:38.30 | mrtelephone | but cisco won't accept the packets on the port asterisk sends to |
| 15:38.33 | robl^ | mrtelephone: not sure anymore. I haven't used the phones in about 3 yrs. I moved away from Cisco long ago. We use Polycom and Aastra only |
| 15:38.48 | ManxPower | mrtelephone: asterisk is sending on the wrong port because the phone is set up for NAT |
| 15:38.48 | mrtelephone | or the router blocks it |
| 15:39.11 | mrtelephone | so if I turn off nat and try again with asterisk nat=yes then it should still connect? |
| 15:39.11 | ManxPower | mrtelephone: turn off nat support in the phone config |
| 15:39.17 | ManxPower | mrtelephone: correct. |
| 15:39.19 | mrtelephone | ok I have it working on the local network now |
| 15:39.32 | ManxPower | you may have to stop and start asterisk and reboot the phone since all that stuff is wrong right now. |
| 15:39.53 | mrtelephone | reload won't cut it? |
| 15:40.05 | ManxPower | mrtelephone: asterisk caches all the ip and port info across a reload. |
| 15:40.16 | ManxPower | I did not say "will" I said "might" have to stop and start asterisk. |
| 15:40.42 | ManxPower | that info SHOULD be updated when the phone reregisters when you reboot the phone. |
| 15:40.57 | mrtelephone | the cisco phoen is sending this for contact info Contact: <sip:504@192.168.123.111:5060; |
| 15:41.10 | mrtelephone | 5060.. asterisk will send to 51118 or whatever port the message came in on |
| 15:41.15 | *** join/#asterisk Pauliukas (n=Paulius@bas1-montrealak-1128581474.dsl.bell.ca) |
| 15:41.23 | mrtelephone | but the phone wont' punch through 5060 because its sending on 51118 |
| 15:41.27 | *** join/#asterisk alphanet (i=ircuser@shakotay.alphanet.ch) |
| 15:41.40 | ManxPower | mrtelephone: and it will do that, because asterisk will see thru that fake address. |
| 15:42.22 | ManxPower | mrtelephone: how do you know the phone is sending on 51118 and not sending on 5060 and the NAT router changing it to 51118. |
| 15:42.57 | ManxPower | Asterisk's nat=yes makes asterisk ignore the ip and port info that is in the SIP headers and makes asterisk use the source ip/port in the PACKET header. |
| 15:43.07 | socken23 | Hey something else. I've won a linksys CIT200 phone (skype). Is there ANY way to modify the firmware?? |
| 15:43.10 | alphanet | hello! assuming a call came to an extension, but was NOT Answer()ed yet, is there a way to do a Dial(), and when the callee answers, Dial() would NOT answer the original call but instead process some context instead? I saw the G Dial option in v1.2 and wonder if it is applicable here? |
| 15:43.11 | mrtelephone | because I have it on the localnetwork and asterisk is reading from 192.168.123.111:51118 but the contact info is 192.168.123.111:5060 |
| 15:43.25 | socken23 | It communicates over DECT but I can't use it with my normal dect base station... stupid... |
| 15:43.37 | ManxPower | mrtelephone: Then the phone is setup wrong. |
| 15:43.43 | mrtelephone | usually nat=yes works even on a localnetwork because the actual ip is the actual.. but not with cisco junk |
| 15:43.51 | mrtelephone | ok |
| 15:43.58 | mrtelephone | theres a setting for voip control port |
| 15:44.00 | mrtelephone | maybe thats doing it |
| 15:44.16 | ManxPower | mrtelephone: Cisco phones are just as good as Polycom. It's just that the LICENSING for the Cisco phones is horrible. |
| 15:44.29 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177582435.dsl.bell.ca) |
| 15:44.36 | ManxPower | mrtelephone: I already told you to clear the config of the phone and ONLY change the min required. |
| 15:45.23 | cpina | hi... ManxPower and everybody. Using last bristuff with asterisk 1.2.24 and ztdummy, background (or any play) is not working (completely silence) |
| 15:45.39 | cpina | why do we have problems using ztdummy and just a playing? |
| 15:45.53 | tzafrir | cpina, what do you get for zttest ? |
| 15:46.00 | tzafrir | no output at all? |
| 15:46.11 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 15:46.44 | cpina | tzafrir: waiting for result... |
| 15:46.50 | cpina | Opened pseudo zap interface, measuring accuracy... |
| 15:46.56 | Siya | somehow => '_00[1N][1N].' => 1. Macro(trunkdial|${trunk_4}/${EXTEN:0}) [pbx_config] |
| 15:46.57 | cpina | (already for a minute) |
| 15:47.10 | Siya | doesn't match 003312345678 |
| 15:47.34 | cpina | (i pressed ctrl+c and then:) |
| 15:47.40 | Siya | I'd like it to match all international number I don;t already match specifically |
| 15:47.45 | cpina | --- Results after 0 passes --- , Best: 0.000000 -- Worst: 100.000000 -- Average: 100.000000 |
| 15:47.52 | Siya | what am I missing? |
| 15:48.10 | [TK]D-Fender | Siya: OH, I don't think you can use N, X, etc inside [] |
| 15:48.26 | [TK]D-Fender | Siya: just do [1-9] |
| 15:48.54 | FlatFoot | afternoon all , anyone recommend a good sip softphone ? |
| 15:49.15 | Siya | [TK]D-Fender: cool was wondering that |
| 15:49.29 | [TK]D-Fender | FlatFoot: Ekiga or eyeBeam |
| 15:49.30 | xpot | is there a known issue in svn branch asterisk 1.4 with cdr_tds using func_odbc and MSSQL? |
| 15:49.53 | FlatFoot | [TK]D-Fender: cheers for that |
| 15:49.56 | Siya | [TK]D-Fender: perfect, problem solved \o/ |
| 15:50.28 | [TK]D-Fender | Siya: NP |
| 15:50.30 | Siya | [TK]D-Fender: how do ekiga and eyebeam compare to x-lite? |
| 15:50.32 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.218.nw.nuvox.net) |
| 15:50.50 | [TK]D-Fender | Siya: X-Lite is the light version of eyebeam with all the good stuff cut out |
| 15:51.07 | [TK]D-Fender | Siya: Ekia is like eyebeam with G.729 cut out |
| 15:51.22 | [TK]D-Fender | Siya: And any other licensed codecs |
| 15:51.26 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.133.162) |
| 15:52.07 | *** join/#asterisk simond (n=simon@208.68.95.5) |
| 15:52.09 | Siya | the good stuff like G.927 and video :) |
| 15:52.16 | Siya | [TK]D-Fender: tx again :) |
| 15:52.49 | Siya | Mothers that are feeling ill are not to be messed with |
| 15:52.51 | Siya | ... |
| 15:52.58 | simond | I'm running into an odd problem with asterisk 1.2.24 when trying to route calls based on callerid |
| 15:52.59 | coppice | don't interfere when someone is busy |
| 15:53.06 | [TK]D-Fender | Siya: no, they both have video.... jsut H.263+ only |
| 15:53.13 | [TK]D-Fender | (or was that non +?) |
| 15:53.39 | simond | if I have exten => XXX/222,1,NoOp(hi), and I dial something from a phone with callerid 222, it isn't matched |
| 15:53.59 | simond | if however, I had another line that just says exten => XXX,1,NoOp(blah), the previous line suddenly starts working |
| 15:54.03 | simond | is this normal? |
| 15:54.46 | [TK]D-Fender | simond: is that the EXACT line pasted? |
| 15:55.28 | dandre | How can I trace manager commands and reponses form CLI interface? |
| 15:55.55 | simond | [TK]D-Fender: it was exactly this: |
| 15:56.01 | *** join/#asterisk uwe (n=uwe@a24-163.adsl.paltel.net) |
| 15:56.10 | simond | exten => 123/4166666666,1,Dial(SIP/foo) |
| 15:56.17 | simond | exten => 123,1,NoOp() |
| 15:56.28 | mrtelephone | if the port isn't in the <sip:504@192.168.123.111:5060; contact header then what do sip proxys/clients usually do? |
| 15:56.34 | [TK]D-Fender | simond: pastebin your full context and the CLI output of a failed call with channel debug enabled |
| 15:56.43 | mrtelephone | is the port option mandatory? |
| 15:56.51 | [TK]D-Fender | ~pb |
| 15:56.52 | jbot | A Pastebin is a place to paste your stuff without flooding 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://www.rafb.net/paste |
| 15:56.53 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^ |
| 15:56.55 | mrtelephone | asterisk doesn't send out a port |
| 15:57.21 | simond | [TK]D-Fender: what do you mean by 'channel debug' ? |
| 15:57.26 | mrtelephone | so Im guessing its not mandatory and the client or peer should be smart enough to send back to port it came on by default? |
| 15:57.52 | [TK]D-Fender | simond: if your call is coming in via SIP do "sip debug", IAx2, etc... |
| 15:58.17 | simond | k |
| 15:59.02 | *** join/#asterisk rpm (n=russell@75.153.47.179) |
| 16:02.24 | dandre | How can I trace manager commands and reponses form CLI interface? |
| 16:05.00 | [TK]D-Fender | dandre: unload chan_brokenrecord.so |
| 16:05.02 | simond | [TK]D-Fender: http://rafb.net/p/8gdBvS84.html |
| 16:05.05 | *** part/#asterisk Pauliukas (n=Paulius@bas1-montrealak-1128581474.dsl.bell.ca) |
| 16:05.31 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 16:05.31 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 16:05.53 | dandre | [TK]D-Fender: what is it supposed to do? |
| 16:06.13 | shido6 | how do you pass a variable to an external application using 2 priorities, one to set the variable and the other to run the System app which fires off your appliaction? |
| 16:08.08 | [TK]D-Fender | simond: Where do you see 4166666666 in that incoming call? So far it shouldn't match what you pasted and is reacting pretty normally. |
| 16:08.23 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 16:08.30 | simond | I don't see it matching, that's the problem. |
| 16:08.44 | [TK]D-Fender | dandre: its supposed to give you a subtle behavior modification hint. |
| 16:09.07 | simond | I put the callerid in the SIP profile, it's displayed on the endpoint I'm calling -- however * isn't seeing it for some reason. |
| 16:09.15 | *** join/#asterisk ghenry (n=ghenry@ghenry.plus.com) |
| 16:09.19 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 16:09.25 | [TK]D-Fender | simond: And since the only thing you showed was a CID restricted exten to which the CID does NOT match... why SHOULD it match the exten? |
| 16:10.17 | harpal | ai-a, now asterisk 1.4.13 installed on my system |
| 16:10.35 | harpal | I have done make clean earlier but that has no effect |
| 16:10.35 | ai-a | good'o. |
| 16:10.40 | FlatFoot | [TK]D-Fender: tried those softphones , not overly keen . BUT have just been told about one from nch ( Express Talk ) that seems quite nice |
| 16:10.44 | ai-a | hope its not a live system ;) |
| 16:10.55 | simond | [TK]D-Fender: one sec, I'll grab the output where it works for some myserious reason |
| 16:10.59 | [TK]D-Fender | FlatFoot: Zoiper is pretty decent... |
| 16:11.06 | harpal | I have removed that folder and untar new asterisks |
| 16:11.10 | FlatFoot | Zoiper i shall have a look |
| 16:11.16 | harpal | Yes I have installed to one vmware machine |
| 16:11.22 | harpal | its not live |
| 16:11.23 | [TK]D-Fender | simond: and pastebiin your sip config for the affected phone. |
| 16:11.28 | harpal | now how to test that ? |
| 16:11.33 | simond | sure thing |
| 16:12.04 | *** join/#asterisk LakeSolon (n=blake@64-83-209-86.dhcp.stcd.mn.charter.com) |
| 16:12.17 | FlatFoot | [TK]D-Fender: i see it was IDEFISK used that one before was quite impressed |
| 16:13.38 | harpal | ai-a, how to test that? |
| 16:14.12 | *** join/#asterisk techie (n=techie@adsl-76-214-19-182.dsl.lsan03.sbcglobal.net) |
| 16:15.05 | dandre | is there any plan to make Match parameter of UpdateConfig manager interface to be clever about regex or wildcar? |
| 16:15.47 | [TK]D-Fender | dandre: Go check Mantis |
| 16:16.35 | simond | [TK]D-Fender: http://rafb.net/p/7usG8L56.html |
| 16:17.11 | *** join/#asterisk bantu (n=Miranda@p54A32C6E.dip0.t-ipconnect.de) |
| 16:18.48 | [TK]D-Fender | simond: :/ |
| 16:19.06 | xpot | anyone know what would cause "xxx func_odbc" if unixODBC is installed and FreeTDS? |
| 16:19.56 | ccesario | ~book |
| 16:19.57 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 16:20.20 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:20.32 | *** join/#asterisk Anthro (i=gss@pdpc/supporter/active/Anthro) |
| 16:22.09 | cpina | hi again |
| 16:22.17 | Anthro | I have a Digium IAX to phone adapter (i.e. it has an RJ45 for ethernet and an RJ11 for phone) from a couple of years ago, and I don't remember what it's called or how to configure it. Can anyone point me to the docs? |
| 16:22.33 | cpina | i'm using ztdummy in two boxes, in one works fine and zttest is working fine... but other one nothiing |
| 16:22.36 | cpina | how i can start to debug? |
| 16:22.45 | cpina | same kernel version, different CPU... |
| 16:23.03 | cpina | in not-working one i see some rtc: lost some interrupts at 1024Hz. |
| 16:23.11 | cpina | any help is appreciated :-) |
| 16:23.22 | mrtelephone | should phones send sip messages on the same port they receive? |
| 16:23.25 | xpot | FYI: resolution to "XXX func_odbc" was to install unixODBC-dev |
| 16:23.34 | simond | [TK]D-Fender: exactly; is there some way to see why asterisk didn't match a particular rule? |
| 16:23.48 | [TK]D-Fender | simond: Really not sure. I'm stumped on this.... |
| 16:24.51 | simond | I suppose file a bug? |
| 16:25.32 | Qwell | Anthro: IAXy |
| 16:25.54 | Anthro | Qwell: Ah! Yes. That's it. Now I can google for it effectively :-) |
| 16:26.54 | tzafrir | what are the two CPUs? different systems? |
| 16:26.56 | *** join/#asterisk shinao1 (n=shinao1@217.20.242.50) |
| 16:27.00 | tzafrir | cpina, --^ |
| 16:27.20 | tzafrir | What kernel? |
| 16:27.28 | cpina | tzafrir: yes, different CPU's |
| 16:27.34 | mrtelephone | voip_control_port should make cisco send from 5060 or just listen on 5060? |
| 16:27.40 | tzafrir | what CPUs? |
| 16:28.33 | cpina | Intel(R) Pentium(R) 4 CPU 3.00GHz |
| 16:28.37 | cpina | in non-working one |
| 16:28.53 | cpina | and in working one: Intel(R) Pentium(R) 4 CPU 3.00GHz |
| 16:28.54 | cpina | same one :-) |
| 16:29.04 | cpina | sorry, i was confused (i thought different cpu's but no) |
| 16:29.20 | cpina | so same cpu, same kernel, i will install now in non-working one a previous bristuff (same than there) |
| 16:29.58 | *** join/#asterisk Anthro (i=gss@pdpc/supporter/active/Anthro) |
| 16:30.41 | mrtelephone | its gotta be a setting somewhere but cisco doesn't have detailed docs for sip 8.2 |
| 16:31.45 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 16:31.45 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 16:31.53 | cpina | in one system, there isn't g729 codecs installed, this is the only difference... |
| 16:32.00 | cpina | tzafrir: any other idea? |
| 16:33.11 | *** join/#asterisk CmaX7 (n=neo@dsl-241-71-72.telkomadsl.co.za) |
| 16:33.15 | CmaX7 | ahoy! |
| 16:33.47 | *** join/#asterisk e` (n=e@38.102.196.202) |
| 16:36.56 | [TK]D-Fender | chips |
| 16:37.09 | Qwell | salsa |
| 16:37.45 | [TK]D-Fender | merengue |
| 16:37.53 | Qwell | pie |
| 16:37.59 | [TK]D-Fender | rat |
| 16:38.08 | Qwell | you don't want to know the first thing I thought of |
| 16:38.13 | [TK]D-Fender | :O |
| 16:38.15 | Qwell | skip it |
| 16:38.16 | [TK]D-Fender | heh |
| 16:38.25 | [TK]D-Fender | <- w1nn4r |
| 16:38.49 | Anthro | Okay, my IAXy isn't responding to broadcast pings, and doesn't seem to have requested an IP address from my DHCP server. Any suggestions? |
| 16:39.06 | Qwell | droppings |
| 16:39.29 | Qwell | Anthro: It may have a static IP. I'd reset it |
| 16:39.51 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 16:39.58 | Anthro | Qwell: I pushed the reset button (at least, the only available button). Do I have to hold it for some length of time? |
| 16:40.08 | Qwell | Anthro: yes, but I don't recall exactly how |
| 16:40.17 | Qwell | it's in the docs for it |
| 16:41.57 | Katty | oh |
| 16:41.58 | Katty | look |
| 16:42.01 | Katty | it's Qwell! |
| 16:42.05 | Qwell | where?! |
| 16:42.06 | Anthro | Qwell: Ah! http://kb.digium.com/entry/27/92/ |
| 16:42.07 | Qwell | [TK]D-Fender: next word? |
| 16:44.07 | Anthro | Qwell: And that did it. Woo! |
| 16:44.27 | CmaX7 | Can anyone please help me? |
| 16:44.38 | CmaX7 | I'd like to set up a PPPoE client on AsteriskNow |
| 16:44.43 | CmaX7 | How do I do that? |
| 16:45.09 | [TK]D-Fender | Qwell: bird |
| 16:47.18 | *** join/#asterisk killfill (n=killfill@200.55.220.3) |
| 16:47.24 | killfill | hey.. |
| 16:47.35 | CmaX7 | hey killfill |
| 16:47.43 | killfill | asterisk is binding itself to 127.0.0.1:8088... where do i tell it to do it on 0.0.0.0?... |
| 16:48.28 | Qwell | [TK]D-Fender: who opened that callerid bug? |
| 16:48.31 | killfill | ooh how fool.. :P i was wathcing somethign else.. |
| 16:48.36 | killfill | (never mind..) |
| 16:49.10 | [TK]D-Fender | Qwell: simond has a dialplan CID filter issue.... |
| 16:49.37 | Qwell | simond: Please see the note I added to your report. You'll need to reproduce with 1.4. |
| 16:49.41 | Qwell | [TK]D-Fender: thanks |
| 16:52.21 | *** join/#asterisk XJMAC (n=XJMAC@189.164.87.71) |
| 16:52.29 | XJMAC | hi |
| 16:52.40 | XJMAC | good morning, |
| 16:52.55 | simond | ugh |
| 16:53.19 | CmaX7 | anyone have any idea how to set up PPPoE on AsteriskNow? It only has DHCP or Static IP options in the web gui... |
| 16:53.55 | *** part/#asterisk XJMAC (n=XJMAC@189.164.87.71) |
| 16:53.57 | [TK]D-Fender | CmaX7: You're in the wrong channel, please read the topic. |
| 16:54.37 | CmaX7 | [TK]D-Fender: err ?? This is the Asterisk channel right? |
| 16:55.08 | [TK]D-Fender | CmaX7: AsteriskNOW has its own support channel and * has nothing to do with your networking and distro specific questions. |
| 16:55.13 | cpina | hello again... zttest in one computer is not working at all (so meetme is not working) |
| 16:55.15 | *** part/#asterisk simond (n=simon@208.68.95.5) |
| 16:55.20 | cpina | i have /proc/zaptel/1 too, etc. |
| 16:55.23 | CmaX7 | [TK]D-Fender: aah, thanks... |
| 16:55.23 | cpina | any idea how to debug it? |
| 16:55.48 | Daviey | Anybody having problems with trunk at the moment? |
| 16:55.52 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 16:55.56 | Qwell | Daviey: how...vague |
| 16:56.08 | Daviey | chan_sip.c:16439 sipsock_read: SIP transaction failed: |
| 16:56.31 | Qwell | that looks familiar, actually. I think there is a bug open for that |
| 16:56.55 | Daviey | I'm trying to work out when it started, what revision |
| 16:56.55 | Qwell | oh, nope |
| 16:57.20 | Daviey | :( |
| 16:58.18 | *** part/#asterisk CmaX7 (n=neo@dsl-241-71-72.telkomadsl.co.za) |
| 16:59.46 | *** join/#asterisk Navion (n=billp@75-105-41-123.cust.wildblue.net) |
| 17:02.15 | *** join/#asterisk sacitec (n=tobi@189.149.99.172) |
| 17:05.06 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 17:06.26 | *** join/#asterisk sheldonh (n=sheldonh@66.219.59.32) |
| 17:06.58 | sheldonh | how can i confirm from the asterisk shell that trunking is enabled for an iax2 peer? (asterisk-1.4.11) |
| 17:09.06 | ManxPower | sheldonh: if you set trunk=yes for the entry in iax.conf then IAX2 trunking is enabled. |
| 17:09.23 | ManxPower | it will whine it it can't enable IAX2 trunking (like if you don't have a zaptel device) |
| 17:14.02 | sheldonh | hmmm. we have trunk=yes in iax.conf, but are seeing one-way trunking if we look at packet and bit rates (we're pretty confident it's one-way trunking) |
| 17:15.31 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 17:18.40 | *** join/#asterisk pepo-- (n=pepOSX@190.72.153.45) |
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| 17:24.10 | *** join/#asterisk linxroute (i=linxrout@117.0.26.118) |
| 17:24.45 | *** join/#asterisk darkskiez (i=mhb@cpc1-broo3-0-0-cust659.renf.cable.ntl.com) |
| 17:24.46 | linxroute | . |
| 17:25.47 | lmadsen | .. |
| 17:26.24 | *** join/#asterisk Clayg (n=clay@c-76-123-96-239.hsd1.fl.comcast.net) |
| 17:27.01 | Clayg | anyone having problems with polycom 501 appearing to crash? after 1-2 minutes the screen gets "screwy" \ |
| 17:27.19 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 17:27.28 | *** join/#asterisk Dovid (n=Dovid@bzq-79-181-174-187.red.bezeqint.net) |
| 17:27.51 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 17:28.31 | Dovid | in the US can anyone tell me the difference between a T1 and a Intergrated T1 ? |
| 17:29.16 | linxroute | T1 and "fractional T1" is correct term, i think |
| 17:29.44 | Dovid | linuxroute: is one analog and the other digital or ae they both the same? |
| 17:30.02 | *** join/#asterisk Op3r (n=edwin@121.97.212.248) |
| 17:30.09 | lmadsen | same -- one has 24 channels, the other... less than 24 |
| 17:30.17 | Dovid | ah ok. |
| 17:30.38 | Dovid | and I assume that this card: http://www.telephonydepot.com/product_p/105-020-235.html wont work for it? |
| 17:30.45 | Dovid | oops* |
| 17:30.45 | Dovid | http://www.telephonydepot.com/product_p/105-020-235.htm |
| 17:31.12 | lmadsen | no idea... never used one of those cards in asterisk |
| 17:31.28 | lmadsen | my guess is.... no unless they supply a driver for zaptel |
| 17:31.42 | Dovid | lmadsen: what cards do u reccomend for a PRI ? |
| 17:31.47 | [TK]D-Fender | Dovid: Usually provider give you an IAD to deal with their T1. |
| 17:32.22 | Dovid | TK: I am new to PRI's. what is IAD? |
| 17:32.27 | lmadsen | Dovid: I will recommend Digium cards with the disclaimer that I am a consultant for Digium |
| 17:32.57 | [TK]D-Fender | Dovid: Integrated T1 is typically a mixed voice & data T1, sometimes even dynamic. |
| 17:33.04 | Dovid | lmadsen: I figured that out. which one of digium's card ? |
| 17:33.07 | lmadsen | oooo, dynamic :) |
| 17:33.16 | Alan_Hicks | dynamic T1? |
| 17:33.18 | lmadsen | Dovid: ummmm... one of them :) |
| 17:33.19 | Dovid | the price is $100 difference |
| 17:33.24 | Dovid | for half t1 or full |
| 17:33.27 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 17:33.27 | *** mode/#asterisk [+o anthm] by ChanServ |
| 17:33.39 | lmadsen | honestly I don't do PRI / analog -- I do SIP |
| 17:33.42 | [TK]D-Fender | Dovid: At which point your provider usually gives (or rents) you something like a Cisco 2430 to handle the split off. You'd plug your T1 card into that and the IAD would handle the channel usage |
| 17:35.28 | lmadsen | Dovid: if you are only going to put in a fractional T1 (probably like 8 or 10 channels I'd guess), then just a single port card would be fine I would think, unless you plan on upgrading past that |
| 17:35.31 | *** join/#asterisk Defraz (n=t0tal@fw.fuzecore.com) |
| 17:35.32 | Dovid | TK: which cards would u reccoment for a T1 ? |
| 17:36.16 | Dovid | lmadsen: I think they are gona for thr full T |
| 17:36.26 | [TK]D-Fender | Dovid: Sangoma A101d |
| 17:37.18 | linxroute | ~ |
| 17:37.34 | *** join/#asterisk thix_rj_br (n=Basic@c934c2fc.virtua.com.br) |
| 17:39.04 | Dovid | TK: what is the difference between the different cards that they have ? |
| 17:39.24 | [TK]D-Fender | Dovid: Gee I dunno, go READ and you tell ME <--- |
| 17:41.14 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 17:42.15 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 17:44.49 | Dovid | TK: I dont know much about em hence y i ask you (the expert). hey u even got jbot to be ur B*tch |
| 17:45.17 | *** part/#asterisk techie (n=techie@adsl-76-214-19-182.dsl.lsan03.sbcglobal.net) |
| 17:45.53 | *** join/#asterisk TrentCreek (n=tjones14@cpe-70-117-207-168.rgv.res.rr.com) |
| 17:46.06 | [TK]D-Fender | Dovid: go READ, this stuff is very apparent. |
| 17:46.56 | Dovid | lol tk. i see wut ur saying |
| 17:49.13 | thix_rj_br | Hi, everyone. I live in Brazil and had two TDM400 (conected at 8 analog pstn lines) on my * box. Recently, a bought a TDM2400 and the Caller Id stopped working. Can anyone explain me why? Nothing (except the channel numbers) was changed on zaptel or zapata configurations. |
| 17:49.38 | *** join/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 17:50.17 | *** join/#asterisk gvasterisk (n=prueba@200.69.249.33) |
| 17:50.42 | gvasterisk | hi, I've a little question, Trixbox says that Physical memory reachs up 92% why does it happen? |
| 17:51.13 | thix_rj_br | The important lines at the zapata.conf are: cidsignalling=DTMF |
| 17:51.13 | thix_rj_br | cidstart=polarity |
| 17:51.27 | thix_rj_br | callerid=asreceived |
| 17:51.27 | thix_rj_br | usecallerid=yes |
| 17:52.07 | thix_rj_br | Everything was working fine till the boards switch |
| 17:52.35 | linxroute | have anyone here been using SRTP with asterisk, i google but found very limited information |
| 17:53.37 | *** join/#asterisk penguinFunk_ (n=cylon@unaffiliated/penguinfunk) |
| 17:53.53 | *** join/#asterisk simond (n=simon@208.68.95.5) |
| 17:54.06 | tzafrir | gvasterisk, what is the output of 'free'? |
| 17:54.21 | *** join/#asterisk fiXXXerMet (n=kjohnson@69.85.26.2) |
| 17:54.24 | tzafrir | 92% inclludes buffered+cached? |
| 17:54.46 | tzafrir | gvasterisk, also: run 'top' and press 'm' |
| 17:55.04 | gvasterisk | Kernel + Buffer + Cached |
| 17:55.35 | gvasterisk | top? what's that? |
| 17:56.03 | tzafrir | your kernel tries to use most of the available memory for I/O buffers and such, so it won't be wasted |
| 17:56.11 | tzafrir | unused memory is wasted memory |
| 17:56.50 | gvasterisk | how can I configure it? |
| 17:57.34 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 17:58.30 | *** join/#asterisk hexorx (n=joshr@63-211-239-34.teliax.com) |
| 17:58.52 | *** join/#asterisk gvasterisk (n=prueba@200.69.249.33) |
| 17:59.00 | gvasterisk | here again, anybody can tell me? |
| 17:59.11 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 17:59.17 | Alan_Hicks | Howdy. I'm not entirely sure what I should be looking for, or if it's possible, so I thought I'd ask here before googling. I've got a client that currently has a proprietary Nortel PBX system with four PSTN lines coming in. Each phone has buttons for lines 1 through 4. |
| 18:00.03 | Alan_Hicks | Inbound calls ring all phones. Some one answers and the call is bridged to that one phone. Later, that user can put them on hold, intercom another person, and ask them to "Pick up on Line 3." to talk to the caller. |
| 18:00.39 | [TK]D-Fender | Alan_Hicks: Forget that key-system SLA functionality. Go lookup Parking on the WIKI |
| 18:00.43 | [TK]D-Fender | ~wikis |
| 18:00.44 | jbot | wikis is probably http://www.voip-info.org |
| 18:00.50 | Alan_Hicks | They want this same functionality with Asterisk and SIP phones (using the same 4 incoming PSTN lines). Is this even possible? I've already figured out call parking and transfers, but they seem set in their ways on this. |
| 18:01.11 | Alan_Hicks | [TK]D-Fender: I've already done call parking, but they don't want that. It's "too confusing". |
| 18:01.22 | [TK]D-Fender | Alan_Hicks: pretty much TFB. |
| 18:01.28 | Alan_Hicks | TFB? |
| 18:02.21 | *** join/#asterisk implicit (n=implicit@c-67-191-24-188.hsd1.fl.comcast.net) |
| 18:02.40 | [TK]D-Fender | ~tfb |
| 18:02.41 | jbot | it has been said that tfb is Too #&^$ing bad.... |
| 18:02.45 | [TK]D-Fender | :) |
| 18:02.51 | gvasterisk | any idea what could be happening with the physical memory? |
| 18:03.01 | [TK]D-Fender | gvasterisk: Gremlins ate it |
| 18:03.05 | Alan_Hicks | Shit. This may be a deal-breaker for them. |
| 18:03.08 | thix_rj_br | did anyone sucessfully used DTMF caller id in Brazil via TDM2400? |
| 18:03.30 | gvasterisk | ~nat |
| 18:03.31 | jbot | [nat] Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 18:03.46 | gvasterisk | where can I get info about nat problems? |
| 18:03.58 | linxroute | google : asterisk nat |
| 18:03.59 | linxroute | :) |
| 18:04.09 | linxroute | that's what i used to find |
| 18:05.44 | [TK]D-Fender | Alan_Hicks: Sad.... |
| 18:06.15 | Alan_Hicks | Yeah. I don't get it myself. I think they're just scared to change, even though they really *need* this asterisk PBX. |
| 18:06.48 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 18:07.28 | Alan_Hicks | Besides, call transfer handles everything for them already. Hit "trans", dial the extension, and the caller is put on hold. extension rings, user2 picks up and can accept or deny the transfer. |
| 18:07.38 | Alan_Hicks | It's essentially EXACTLY what they're doing now. |
| 18:07.46 | linxroute | ;) |
| 18:08.46 | simond | Qwell: I'm able to reproduce the callerid based routing problem in trunk |
| 18:09.27 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 18:09.30 | *** join/#asterisk myiagy (n=myiagy@189.34.11.211) |
| 18:09.43 | linxroute | Alan_ , just wonder what is your sip phone ? |
| 18:10.00 | Runlvl | Comunidad de Asterisk en español! http://www.asterisk-la.org |
| 18:10.03 | Alan_Hicks | linxroute: I'm using Polycom IP 320s right now. |
| 18:10.09 | linxroute | oh |
| 18:10.17 | Alan_Hicks | What do you ask? |
| 18:10.19 | linxroute | so it's 2 lines ? |
| 18:10.29 | Qwell | simond: you can reopen that bug then |
| 18:10.30 | Alan_Hicks | Yes, but I'm not married to them. |
| 18:11.34 | ai-a | Is there anyway to reject / busy a call if the sip phone is already busy on another call ? |
| 18:11.53 | ai-a | Dial(SIP/john) always works,, but thats not good, we want it to fail. |
| 18:12.07 | *** join/#asterisk captiancrash (n=jonmoore@70.159.118.70) |
| 18:12.23 | Alan_Hicks | ai-a: Dial() will exit and set DIALSTATUS if the line is busy. |
| 18:12.41 | Alan_Hicks | SIP/john is probably a multi-line SIP phone. |
| 18:12.48 | ai-a | Alan_Hicks: the sip phone has multi-line.. |
| 18:13.05 | ai-a | i have put call-waiting indication to none,, but it still rings.. yet the called phone doesnt see the call. |
| 18:13.08 | Alan_Hicks | You could re-configure it to use only a single line. |
| 18:13.14 | linxroute | well , i just thought that with phone with something like 4 lines , you can config line 3-4 always use for pickup call , these lines are configured to connect to connect to parked call service directly |
| 18:13.16 | ai-a | Alan_Hicks: then transfers stop working :) |
| 18:13.40 | *** join/#asterisk Navion (n=billp@75-105-41-123.cust.wildblue.net) |
| 18:13.48 | linxroute | so they just press line 3 or 4 to pickup the call |
| 18:13.52 | Alan_Hicks | Interesting question then. I'm really not sure how to do it myself. Maybe some one with more experience will come along. |
| 18:14.33 | Alan_Hicks | linxroute: That's entirely different. Besides, what if they have three parked calls at a time? |
| 18:14.53 | linxroute | yeap |
| 18:14.58 | linxroute | just a silly thought |
| 18:15.25 | [TK]D-Fender | Alan_Hicks: this is really very sad. |
| 18:15.47 | Alan_Hicks | Yeah. I'm gonna do my damndest to convince them that re-learning won't be hard at all. |
| 18:15.53 | [TK]D-Fender | Alan_Hicks: parking works and any 2-bit hack is ugly, causes a LOSS of functionality and saves little if anything at all. |
| 18:16.12 | Alan_Hicks | And... since they're looking to hire a new person to handle calls, and I put my sister in for an interview with them, it might be easier. |
| 18:16.22 | linxroute | :) |
| 18:16.29 | *** join/#asterisk Tili (n=tili@153.Red-80-38-134.staticIP.rima-tde.net) |
| 18:16.49 | Alan_Hicks | [TK]D-Fender: Correct, but these people just stopped turning the handle and ringing Sara over in Mayberry to get the sheriff's office. |
| 18:17.21 | [TK]D-Fender | Alan_Hicks: Welcom to the 21st century! |
| 18:17.30 | Alan_Hicks | I know. |
| 18:17.41 | hmmhesays | more mullets than you can throw a schmidt at |
| 18:17.48 | Alan_Hicks | It looks like Asterisk has support for SLA, but it'll be a PITA to get done. |
| 18:17.49 | hmmhesays | oh wait, wrong era |
| 18:18.05 | Alan_Hicks | hmmhesays: That's schtick. ;-) |
| 18:18.21 | hmmhesays | I think you missed the reference |
| 18:18.40 | Alan_Hicks | Maybe I did. |
| 18:18.44 | *** join/#asterisk arguile (i=user224@KTNRON06-1242488957.sdsl.bell.ca) |
| 18:18.52 | hmmhesays | schmidt is a crappy beer |
| 18:19.23 | hmmhesays | my stereotyping says people with mullets drink crappy beer |
| 18:19.26 | *** join/#asterisk ViKing78 (n=ViKing78@cerberus.franklinamerican.com) |
| 18:20.42 | [TK]D-Fender | Alan_Hicks: SLA in *? lol.... NO |
| 18:20.48 | *** join/#asterisk gvasterisk (n=prueba@200.69.249.33) |
| 18:21.17 | Alan_Hicks | [TK]D-Fender: According to this, it's possible: http://www.asterisk.org/node/48342 |
| 18:21.37 | draygon | Has anyone here used Asterisk@Home? |
| 18:21.42 | [TK]D-Fender | Alan_Hicks: NO :) |
| 18:21.53 | Alan_Hicks | Could you explain please? |
| 18:22.31 | [TK]D-Fender | Alan_Hicks: that is not SLA.... that is a FUGLY 2-bit hack that would require a MINIMUM of an IP 601 to eve TRY to pull of in your scenario, and would confuse them MORE witht he call handling it'd require. |
| 18:23.07 | Alan_Hicks | ok |
| 18:23.15 | [TK]D-Fender | draygon: A@H is no more and has been replaced by Trixbox. Read the channel topic. |
| 18:23.26 | lmadsen | hmmhesays: Pabst Blue Ribbon! |
| 18:23.35 | draygon | Yep, I meant Trixbox |
| 18:23.48 | [TK]D-Fender | draygon: again, please read the channel topic. |
| 18:23.58 | Qwell | lmadsen: so, I was out at a bar downtown one night a few months back... |
| 18:24.10 | lmadsen | seems likely |
| 18:24.21 | Qwell | we left the bar, and the PBR truck was parked out there. It had a tap in the back. Full. Working. |
| 18:24.24 | Qwell | Hilarity ensued. |
| 18:24.29 | lmadsen | lol |
| 18:24.31 | lmadsen | niiiiice |
| 18:24.39 | lmadsen | they can give that stuff away because it's so cheap |
| 18:24.41 | Qwell | heh |
| 18:24.46 | lmadsen | the funny thing is... I got it in college once, and it wasn't really all that bad |
| 18:24.49 | lmadsen | I was shocked |
| 18:25.04 | Qwell | heh |
| 18:25.04 | lmadsen | not nearly as bad a Labatt or Molson products |
| 18:25.44 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 18:25.51 | hmmhesays | yeah PBR hangover in a can |
| 18:25.57 | hmmhesays | while you're drinking it! |
| 18:26.39 | Alan_Hicks | Pabst is good stuff, simply because it's so cheap. |
| 18:26.52 | Qwell | it's like $3 for a 24 pack, right? |
| 18:26.54 | Qwell | :p |
| 18:27.03 | Alan_Hicks | $3.50. |
| 18:27.09 | Qwell | inflation sucks |
| 18:27.15 | lmadsen | :) |
| 18:27.18 | Alan_Hicks | Half that is the cost of the aluminum cans. ;-) |
| 18:27.36 | Qwell | lmadsen: I actually have a video of that night somewhere, btw.. |
| 18:27.37 | *** join/#asterisk implicit (n=implicit@c-67-191-24-188.hsd1.fl.comcast.net) |
| 18:28.03 | lmadsen | would cost like $2 a can if I bought it with Canadian money in the US |
| 18:28.03 | lmadsen | s/can/case |
| 18:28.11 | lmadsen | well that seems dangerous :) |
| 18:28.16 | lmadsen | would be funny to see for sure |
| 18:28.32 | jameswf | $30 in mexican money |
| 18:28.35 | lmadsen | welp, I'm off to drop some stuff off to my accountant for my business end of year, then to drink some lowenbrau or something |
| 18:28.58 | hmmhesays | what does trixbox use for its web voicemail? |
| 18:29.07 | jameswf | freepbx |
| 18:29.20 | jameswf | << #trixbox |
| 18:30.15 | hmmhesays | yea yeah |
| 18:30.40 | jameswf | trixbox is kinda a four letter eord here.... |
| 18:30.49 | Qwell | jameswf: lame? |
| 18:30.52 | jameswf | s/eord/word |
| 18:31.13 | jameswf | Qwell: more graphic |
| 18:31.27 | hmmhesays | yeah I'm aware, I figured someone would know off the top of their head |
| 18:32.30 | gvasterisk | how do I install mystun server? |
| 18:32.34 | jameswf | trixbox is a compilation , in it self it doesnt do anything really and what it does do it does poorly, #freepbx does most of the legwork configuration wise so maybe that is a better venue |
| 18:33.12 | [TK]D-Fender | hmmhesays: http://www.youtube.com/watch?v=dtl2MBS-frk |
| 18:34.05 | Qwell | jameswf: If you ever meet Andrew, ask him how much of the money he got he shared with the freepbx folks. |
| 18:35.58 | jameswf | andrew = andy brown? |
| 18:36.07 | Qwell | gillis, or whatever |
| 18:36.20 | Qwell | of trixbox shame |
| 18:36.34 | *** join/#asterisk murdmath (n=vircuser@64-42-64-98.atgi.net) |
| 18:37.10 | murdmath | Has anyone here used e&m? |
| 18:37.17 | hmmhesays | that guy is crazy |
| 18:37.33 | murdmath | e&m signalling that is. |
| 18:38.08 | linxroute | ear and mouth :) |
| 18:38.17 | murdmath | I've got a digium t1 card and as soon as I plug in the t1 line into all of my incoming trunks start rining. |
| 18:38.30 | jameswf | someone flamed kerry, he did a features thread and he said "thats a freepbx feature" they were like so what develop it and give back to freepbx/// |
| 18:39.52 | [TK]D-Fender | Kerry is a 2-bit sell-out who wrote chumpy articles pimping chump products and target the whole thing to equally chump users. |
| 18:39.55 | jameswf | andrew looks like the revenge of the nerds guy.... chris lynman looks kinda like a serial killer |
| 18:40.04 | *** join/#asterisk fs-unlocked (i=fs-unloc@200.234.206.130) |
| 18:40.37 | [TK]D-Fender | Cater to idiots and you will attract them.... |
| 18:42.02 | draygon | "Cater to idiots and you will attract them...." |
| 18:42.06 | *** join/#asterisk _ys (n=yuri@80.70.236.69) |
| 18:42.06 | draygon | true words ^^ |
| 18:42.08 | *** join/#asterisk Dan0maN_Work (n=dschuh@64.149.174.136) |
| 18:42.12 | Qwell | murdmath: You know Digium gives free install support, right? |
| 18:42.27 | murdmath | Qwell: I do, does it cover this? |
| 18:42.37 | Qwell | you're installing it, right? :) |
| 18:42.48 | draygon | heh |
| 18:42.57 | murdmath | Qwell: I'll give them a call. |
| 18:43.02 | Qwell | yeah, it does initial setup, for common things. I assume E&M is common |
| 18:43.10 | [TK]D-Fender | ~assume |
| 18:43.11 | jbot | About assumptions : http://www.moviewavs.com/php/sounds/?id=gog&media=WAVS&type=Movies&movie=Long_Kiss_Goodnight"e=assume.txt&file=assume.wav |
| 18:43.13 | jameswf | Neil Zaza - I'm Alright << was pretty bad arse, audio went to he77 at the end though |
| 18:44.08 | [TK]D-Fender | jameswf: Wasn't for studio value... this is a perfect example of my own style. |
| 18:44.17 | Dan0maN_Work | i have an off the wall question: the license file for the * moh says it was from www.freemusicplay.com, which no longer exists apparently. anyone know where this site/these people that * got the free music from has moved to? |
| 18:44.55 | *** join/#asterisk Rhinoo_ah (n=ahonea@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 18:47.37 | Dan0maN_Work | background for that question: we currently have a contract (ridiculously expensive one) for our MOH for our lucent definity. it's expiring. we want to find some free music to play until we make our move to * |
| 18:48.33 | *** join/#asterisk kkn088 (n=kikoun@77.204.108.68) |
| 18:48.57 | twisted | whoa |
| 18:49.06 | twisted | Qwell: don't make assumptions in telecom |
| 18:49.52 | davevg-btwtech | i think i've seen the current * moh on http://freeplaymusic.com/ |
| 18:50.28 | Alan_Hicks | Dan0maN_Work: There's also http://www.royaltyfreemusic.com/ |
| 18:51.16 | Dan0maN_Work | ah |
| 18:51.20 | Dan0maN_Work | thanks guys |
| 18:53.00 | *** join/#asterisk ManxPower (n=manxpowe@247.sub-70-223-210.myvzw.com) |
| 18:53.58 | *** join/#asterisk saftsack (n=saftsack@pD9E06EC0.dip.t-dialin.net) |
| 18:54.01 | jameswf | I like the loops makes me wanna get in an elivator |
| 18:54.29 | Qwell | twisted: I like to make assumptions in code too. |
| 18:54.43 | Qwell | if (a || !a) /* a is probably true here */ |
| 18:55.40 | *** join/#asterisk eject_ck (n=eject_ck@serenadely-waterfront.volia.net) |
| 18:55.49 | [TK]D-Fender | (o) (o) |
| 18:55.53 | ManxPower | I just crashed Asterisk. |
| 18:56.38 | eject_ck | I added into indications.conf section [ua] (copy of [ru]) and now I see in log [Nov 7 20:55:14] NOTICE[25248]: indications.c:505 ast_unregister_indication_country: Removed default indication country 'ua' |
| 18:56.42 | eject_ck | what it mean ? |
| 18:57.01 | ManxPower | eject_ck: that means you did a reload |
| 18:57.15 | ManxPower | Notice that it is a NOTICE and not an ERROR or WARNING. |
| 18:58.01 | eject_ck | yeah reload |
| 19:00.15 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 19:00.18 | *** join/#asterisk Anthro (i=gss@pdpc/supporter/active/Anthro) |
| 19:03.35 | mrtelephone | shaw cable needs new network techs |
| 19:03.38 | mrtelephone | im applying for a job |
| 19:04.04 | ManxPower | I'm glad I'm not a saw cable customer |
| 19:04.11 | ManxPower | or a shaw cable customer, either. |
| 19:04.40 | mrtelephone | the reason why im having trouble with the cisco phones IS the fact that sip firmware 8.2 wants to use random local ports for sending sip messages |
| 19:04.51 | mrtelephone | i hooked up my polycoms to the localnet and their local port is 5060 |
| 19:05.10 | *** join/#asterisk mmlj4 (n=jkelly@ip70-171-94-246.no.no.cox.net) |
| 19:05.22 | mrtelephone | hey can you ping 24.109.106.80 |
| 19:06.39 | mrtelephone | shaw routes there packets through from thunder bay to winnipeg then back to toronto |
| 19:07.00 | IPetrov2 | C:\Program Files\Far>ping 24.109.106.80 |
| 19:07.01 | IPetrov2 | ïÂÍÅÎ ÐÁËÅÔÁÍÉ Ó 24.109.106.80 ÐÏ 32 ÂÁÊÔ: |
| 19:07.01 | IPetrov2 | ðÒÅ×ÙÛÅÎ ÉÎÔÅÒ×ÁÌ ÏÖÉÄÁÎÉÑ ÄÌÑ ÚÁÐÒÏÓÁ. |
| 19:07.03 | mrtelephone | thats an extra 2400 km |
| 19:07.06 | IPetrov2 | timeout :) |
| 19:07.14 | mrtelephone | idiots |
| 19:07.18 | mrtelephone | how hard is it to route really |
| 19:09.37 | mrtelephone | if you load balance proxies how do you avoid looped calls.. in the dialplan you would have to check sip2 and sip2 will in turn check sip1 |
| 19:09.50 | *** join/#asterisk agx (n=badpengu@81-174-8-217.dynamic.ngi.it) |
| 19:11.14 | variable_office | what is a common cause for the caller hearing the callee but the callee not hearing the calleer? |
| 19:11.48 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 19:12.00 | mrtelephone | nat with rtp |
| 19:12.05 | mrtelephone | nat and rtp traffic |
| 19:12.21 | mrtelephone | the callee's phone isn't getting the rtp stream from the caller |
| 19:12.56 | variable_office | mrtelephone, where would the nat be? on the callers side? |
| 19:13.15 | mrtelephone | the callee if he isn't hearing |
| 19:13.37 | mrtelephone | the callee is sittin behind a residential gateway or osmething and the rtp isn't punching through the router |
| 19:14.00 | mrtelephone | if your using asterisk make sure canreinvite=no and nat=yes on in the clients settings in sip.conf |
| 19:14.39 | De_Mon | what happens if you set nat=yes and they arn't natted? |
| 19:15.10 | mrtelephone | it still works if the device is sending sip from port 5060 |
| 19:15.12 | De_Mon | I'm thinking there arnt any side effects |
| 19:15.30 | variable_office | mrtelephone, bother of those are set correctly |
| 19:15.34 | mrtelephone | if the phone sends sip messages from another port nat=yes will screw up a device on a local network |
| 19:15.34 | agx | with 1.4 i think you should use canreinvite=nonat |
| 19:15.39 | *** join/#asterisk VoicePulse (n=contact@unaffiliated/voicepulse) |
| 19:15.51 | variable_office | the problem is with the caller behind the nat, and the callee on pots |
| 19:16.04 | Qwell | ~sipnat |
| 19:16.05 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 19:16.08 | mrtelephone | callee is on pots and can't hear? |
| 19:16.26 | mrtelephone | is asterisk behind a nat? |
| 19:16.49 | mrtelephone | to be honest i had a girl complain a few days ago saying noone she called could hear her |
| 19:17.04 | mrtelephone | time for an asterisk restart |
| 19:17.11 | jameswf | http://video.google.com/googleplayer.swf?docId=8583540508309604526&hl=en << tcp/ip this is the coolest video... |
| 19:17.38 | mrtelephone | time for a server reboot, its been 109 day |
| 19:17.39 | mrtelephone | s |
| 19:18.00 | variable_office | mrtelephone, the caller is behind a nat, but the asterisk box is public |
| 19:18.10 | variable_office | the asterisk box forwards the calls over sip trunks to pots |
| 19:18.15 | codeshah | hi guys, should I get asteriskNOW or install it on an existing LINUX BOX? I have a free box, and was wondering what the better option is |
| 19:18.34 | mrtelephone | variable_office.. your directing calls out through a sip provider? |
| 19:18.36 | mrtelephone | ~siptrunk |
| 19:18.37 | jbot | i guess siptrunk is Asterisk does not support SIP Trunks. Set trunk=no in sip.conf and then set up the device normally in sip.conf. |
| 19:18.37 | jameswf | codeshah: the best option depends on use |
| 19:18.45 | mrtelephone | haha |
| 19:18.56 | mrtelephone | im startin to act like qwell |
| 19:19.04 | codeshah | jameswf, I would use this for asterisk only . I want to essentially play with it, setup a small home system etc . |
| 19:19.21 | jameswf | with cards or pure voip |
| 19:19.21 | codeshah | jameswf, it wont be for production . just dev / test . |
| 19:19.25 | Qwell | codeshah: are you familiar with linux? Do you feel comfortable editing configs by hand? |
| 19:19.44 | codeshah | qwell, yeah I am on ubuntu right now . I code in linux mainly . just dont want to put asterisk on my work machine |
| 19:20.06 | jameswf | If it is pure VOIP or your using digium cards. switchvox looks neat |
| 19:20.09 | codeshah | jameswf, pure VOIP to start probably, but I do want the option for both . |
| 19:20.13 | mrtelephone | editing configs by hand is a must :-/ |
| 19:20.16 | codeshah | switchvox eh? |
| 19:20.19 | mrtelephone | you have to know whats going on |
| 19:20.23 | variable_office | mrtelephone, yes a sip provider |
| 19:20.48 | mrtelephone | variable_office, thats complicated.. does it happen to every call? |
| 19:21.03 | jameswf | I think it is best "for learning" to start with a distro and build ecerything... |
| 19:21.21 | variable_office | mrtelephone, no only to some, and it doesnt happen when i am using it |
| 19:21.37 | mrtelephone | the biggest annoyance of buying proprietory stuff is the configurability is limited.. when you have access to every config file you can do anything :( |
| 19:21.39 | codeshah | jameswf, thanks . |
| 19:21.41 | variable_office | I dont have enough users yet to report any kindof statistics, but that user is the only one who has complained |
| 19:21.59 | mrtelephone | variable_office, it never frigs up when the tech is around :( |
| 19:22.09 | codeshah | so, if I am using pure VOIP, I don't need a card, eh? |
| 19:22.50 | mrtelephone | variable_office, it only happens to him sometimes? |
| 19:23.00 | mrtelephone | damn those problems are hard to fix if its intermitten |
| 19:23.13 | variable_office | mrtelephone, yep, and not that often from what I hear |
| 19:23.28 | variable_office | ya, i know the inconsistency of the problem is what is causing me not to fix it |
| 19:23.37 | codeshah | ... you guys think GotVMail.com uses ASTERISK? Or something similar ? I am using them for service, and then found out about asterisk . |
| 19:24.31 | *** join/#asterisk asdx (n=diego@adsl-139-171.click.com.py) |
| 19:24.35 | asdx | hi |
| 19:24.53 | variable_office | mrtelephone, i keep trying to run wireshark and then analyzing the calls when he says this call is the problem, but wireshark eats up 2gb of ram and then crashes my machine. there any better tools for this? |
| 19:25.32 | asdx | our telephone company is blocking SIP, and they are a monopoly of internet, is there a way to bypass this? |
| 19:25.40 | asdx | can i use SIP over SSH or VPN? |
| 19:26.07 | hmmhesays | openvpn works quite well |
| 19:26.10 | |omni| | just proxy it |
| 19:26.23 | |omni| | you can do VPN, but then you'll have VPN overhead as well |
| 19:26.32 | |omni| | if you have decent bandwidth and latency..not a big deal |
| 19:26.44 | *** join/#asterisk speekac (n=alwin@124.110.48.60.brk01-home.tm.net.my) |
| 19:26.47 | speekac | hi there |
| 19:26.59 | speekac | anyone here familiar with grandstream configuration file ? |
| 19:27.19 | cpm | ~grandstream |
| 19:27.19 | jbot | it has been said that grandstream is the Yugo of VoIP hardware. Run. Run away now. |
| 19:27.29 | asdx | ok thanks |
| 19:27.44 | mrtelephone | 80%variable_office.. rtp debug ip <ip> |
| 19:28.02 | mrtelephone | in asterisk? |
| 19:28.08 | cpm | speekac, that said, check http://www.voip-info.org/wiki/view/Grandstream+Configuration+Tool |
| 19:28.10 | variable_office | mrtelephone, is that within asterisk? |
| 19:28.13 | mrtelephone | yeah |
| 19:28.48 | mrtelephone | are you trying to use g729? |
| 19:29.01 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 19:29.14 | variable_office | g711 |
| 19:29.51 | hmmhesays | anyone ever configured ari standalone? |
| 19:29.56 | speekac | cpm: i'm having problem getting the CRC for the config file |
| 19:30.10 | jameswf | ari is kinda crap |
| 19:30.28 | speekac | cpm: i'm already on that page, and able to get 14bytes of the header, except for the CRC checksum |
| 19:31.24 | hmmhesays | yeah show me something better for web based voicemail |
| 19:32.24 | jameswf | I am unaware of any apps that fill the gap for what ARI does but it is still crap. |
| 19:32.43 | hmmhesays | heh |
| 19:33.45 | mrtelephone | variable_office, i wouldn't be surprised if it was your provider |
| 19:33.52 | De_Mon | doesnt asterisk already have its own crappy voicemail app? |
| 19:33.55 | De_Mon | web voicemail app |
| 19:34.23 | ai-a | http://pastebin.ca/765045 - Zap is failing at asterisk start 'Unable to specify channel 1: No such device or address' |
| 19:34.45 | jameswf | I have ARI replacement on my list of things to write, I would rather use something else... |
| 19:35.05 | variable_office | mrtelephone, ya, more likely than not; i am struggling to get hooked up to the pstn soon, but for now I have to make due |
| 19:35.18 | variable_office | mrtelephone, are you hooked up directly with the pstn? |
| 19:35.25 | mrtelephone | yeah i have a small t1 |
| 19:35.32 | mrtelephone | teeny weeny |
| 19:35.46 | *** join/#asterisk rantsh (n=rantsh@190.36.185.139) |
| 19:36.00 | mrtelephone | i tried voicepuls and gtcomm for voip providers and they worked pretty good |
| 19:36.07 | mrtelephone | im going to go back using them in case the t1 overflows |
| 19:36.31 | asdx | |omni|: what can i use for proxy? |
| 19:36.59 | hmmhesays | jameswf: that doesn't really help me at the moment now does it |
| 19:37.45 | variable_office | mrtelephone, how are you porting numbers onto the t1? |
| 19:37.55 | variable_office | that was my big thing; that and if the t1 failed |
| 19:38.04 | mrtelephone | im just with the local phone company.. im only handing out local service |
| 19:38.29 | mrtelephone | if the t1 fails then Im tits up |
| 19:38.39 | mrtelephone | i can pay for a backup channel but I can't afford it right now |
| 19:39.28 | mrtelephone | im not sure how backup d channels work |
| 19:41.35 | gvasterisk | any good billing system? |
| 19:41.56 | gvasterisk | ( free ) |
| 19:41.56 | gvasterisk | :P |
| 19:50.32 | *** join/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 19:51.26 | Anthro | Okay, I clearly don't know what the hell I'm doing here. I have a feeling that the problem is conceptual rather than technical. I have an IAXy at 192.168.1.10. I have AsteriskNOW running (in a VM) at 192.168.1.8. I used "iax2 provision 192.168.1.10 default" on the asterisk console to provision the IAXy, and I think it worked. I have a phone hooked up to the IAXy and get only silence (though I do have power). Right now I just want dialtone. Wh |
| 19:51.28 | Anthro | at am I missing? |
| 19:52.51 | lirakis | im trying to install zaptel-1.4 from svn. I did make menuselect and just selected ztdummy (i have no tdm hardware) ... i then did make && make install. .. at make install i get "install -d /sbin install /sbin/ - install: too few arguments" ... its not a verbose error.. i dont know if i need to select some thing i didnt or what |
| 19:55.40 | lirakis | tzafrir: any info on this? .. i see a bug report |
| 19:57.24 | tzafrir | lirakis, got it. Working on a fix |
| 19:59.20 | lirakis | tzafrir: i wasnt sure if it had been corrected.. by looking at the bug report... |
| 20:00.19 | tzafrir | I don't think anybody before has actively deselected all the programs :-) |
| 20:00.40 | *** join/#asterisk skirmisha (n=viki@87-126-55-7.btc-net.bg) |
| 20:00.45 | skirmisha | hello guys |
| 20:01.00 | tzafrir | lirakis, could you please try http://rapid.tzafrir.org.il/~tzafrir/zaptel_fix_make_install.diff |
| 20:01.02 | skirmisha | any suggestions about virtual T.38 support |
| 20:01.10 | skirmisha | or software fax |
| 20:02.00 | *** join/#asterisk asdx (n=diego@adsl-158-235.click.com.py) |
| 20:02.03 | asdx | if i use SIP over VPN the performance will decrease? |
| 20:02.36 | *** join/#asterisk saftsack (n=saftsack@p57A7512D.dip.t-dialin.net) |
| 20:02.48 | [TK]D-Fender | asdx: Naturally. |
| 20:02.57 | lirakis | asdx: SIP is a messaging protocol ... not the media |
| 20:03.08 | destructure | if I use SIP over carrier pigeon, will performance decrease? |
| 20:03.18 | lirakis | lol destructure++ |
| 20:03.28 | fujin | I've used SIP/uLaw over a PPTP VPN, low latency though (30ms~) and it's been fine. |
| 20:03.59 | fujin | softphone->pc->pptp->VPNTRONS->asterisk |
| 20:04.05 | lirakis | I had a "sales engineer" from global crossing send an email asking if we could do "h323 to g729 conversion" ... i was like ... uhhh.... |
| 20:04.06 | lirakis | ha ha |
| 20:04.17 | fujin | lol |
| 20:05.15 | tzafrir | lirakis, does it work? |
| 20:05.44 | lirakis | tzafrir: hold on.. just did a ./configure .. let me menuselect etc. |
| 20:06.04 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 20:07.20 | skirmisha | come on guys |
| 20:07.32 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 20:07.57 | tzafrir | skirmisha, rxfx/txfax? iaxmodem? |
| 20:08.22 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 20:09.09 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 20:10.48 | lirakis | tzafrir: yeah after your patch, it worked fine. Thank you! .. can you give me a shorthand version of the problem? |
| 20:11.04 | tzafrir | BINS got an empty value |
| 20:11.15 | tzafrir | follow it in Makefile |
| 20:11.19 | skirmisha | tzafrir i need solution for voip fax |
| 20:11.25 | skirmisha | no FXO cards |
| 20:11.40 | skirmisha | just sip faxes and sending as pdf to email |
| 20:11.49 | tzafrir | skirmisha, callweaver? |
| 20:12.02 | skirmisha | that's openpbx right |
| 20:12.31 | skirmisha | i need something that work with asterisk |
| 20:12.35 | *** join/#asterisk WindBack (n=jorge@host133.190-136-231.telecom.net.ar) |
| 20:12.53 | WindBack | I have a doubt |
| 20:13.06 | *** join/#asterisk n0rus (n=ruslan@router.valiyev.net) |
| 20:13.10 | n0rus | Hello all |
| 20:13.14 | WindBack | I have a doubt |
| 20:13.27 | WindBack | When I put canreinvite=yes |
| 20:14.06 | WindBack | I'm making asterisk found as only sip redirect server ?? |
| 20:15.17 | n0rus | I'm running 1.2 on a Debian Etch machine with zaptel-1.2.21 built from sources (need ztdummy). The module is loaded and Asterisk finds it. When I however play something like "exten => 123,1,MP3Player(/usr/share/asterisk/sounds/angel.mp3)", I get a very choppy sound. "set debug" is set to 1000 and doesn't show anything related to this problem. |
| 20:15.47 | n0rus | Where to dig, folks? |
| 20:16.41 | tzafrir | n0rus, if you suspect zaptel timing problem, try zttest |
| 20:17.25 | *** join/#asterisk yannj_fr (i=yannj_fr@vpn.intelunix.fr) |
| 20:18.01 | n0rus | tzafrir, yeah, that's the first think I suspect. Here are some results: 99.975586% 99.926758% 99.975586% 99.926758% 99.975586% 99.975586% 99.926758% 99.975586% |
| 20:18.08 | n0rus | sorry |
| 20:18.13 | tzafrir | n0rus, looks ok |
| 20:18.15 | n0rus | tht's the first thing I suspected |
| 20:18.19 | n0rus | damn |
| 20:18.21 | n0rus | :) |
| 20:18.47 | tzafrir | lirakis, svn up # and thanks |
| 20:19.09 | n0rus | Maybe I should try an alternative to mpg123? Like madplay? What do you think? |
| 20:19.37 | tzafrir | but you're not using mpg123, right? you're using format_mp3 |
| 20:19.50 | tzafrir | anyway, why do you use mp3 files and not a saner format? |
| 20:20.03 | tzafrir | convert the format off-line to wav |
| 20:20.41 | tzafrir | sox file.mp3 -s -w -r 8000 -c 1 file.wav |
| 20:20.55 | tzafrir | you will waste much less CPU on conversion then |
| 20:21.45 | n0rus | tzafrir, my musiconhold.conf is by default... I have mpg123 installed. "ps -ef" shows mpg123 -q -s -f 8192 --mono -r 8000 /usr/share/asterisk/mohmp3/angel.mp3 |
| 20:21.53 | n0rus | Yeah, I think I should try the converting thing |
| 20:22.11 | tzafrir | n0rus, use native moh |
| 20:22.21 | n0rus | ok |
| 20:22.22 | n0rus | thanks |
| 20:22.25 | n0rus | I'll try |
| 20:24.22 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 20:24.51 | *** join/#asterisk The_X (i=chris@true.fiberpimp.net) |
| 20:25.00 | The_X | anyone ever got asterisk working with an acme packet box? |
| 20:25.46 | *** join/#asterisk lemanal (n=lemanal@adsl-068-016-134-023.sip.msy.bellsouth.net) |
| 20:27.44 | ManxPower | n0rus: native MoH does not support mp3 files. |
| 20:28.12 | ManxPower | n0rus: I don't recommend mpg123 based MoH, but if you insist on using it, make SURE you are on mpg123 v0.59r |
| 20:28.31 | [TK]D-Fender | ManxPower: It can. |
| 20:28.47 | ManxPower | [TK]D-Fender: not without installing asterisk-addons mp3 |
| 20:28.54 | ManxPower | but I don't consider that "part of Asterisk". |
| 20:29.28 | [TK]D-Fender | ManxPower: I never said "as supplied stock with the *-only tarball" now did I? :) Addon's is something of a given... |
| 20:29.38 | outtolunc | catfight! |
| 20:30.20 | ManxPower | I think that if you are going to go thru the hassle of Native MoH mp3 support, you might as well find a sox that supports mp3 and transcode the files |
| 20:30.36 | gvasterisk | is there anyway to put an mp3 as musiconhold? |
| 20:31.16 | ManxPower | gvasterisk: Didn't I just say what you need to do to do that. |
| 20:31.32 | gvasterisk | sorry I wasn't reading |
| 20:31.38 | ManxPower | gvasterisk: well go read |
| 20:31.52 | gvasterisk | ok, sorry :) It wasn't intentional |
| 20:31.59 | [TK]D-Fender | gvasterisk: And we do NOT support Trixbox here. |
| 20:32.19 | ManxPower | gvasterisk is using Trixbox. Time to throw his sorry ass the the 'gators. |
| 20:32.38 | gvasterisk | I wasn't asking for trixbox |
| 20:32.41 | ManxPower | ~zeeek |
| 20:32.41 | jbot | well, zeeek is someone who once said "learning asterisk using a GUI is like learning sex through masturbation. You'll never get to the good stuff." |
| 20:32.43 | *** part/#asterisk fiXXXerMet (n=kjohnson@69.85.26.2) |
| 20:32.52 | gvasterisk | lol, that's great |
| 20:33.11 | ManxPower | gvasterisk: don't worry. They are specially trained gators, they only like meat contaminated with trixbox. |
| 20:33.17 | gvasterisk | ok, I promise I'll start learning without gui |
| 20:35.31 | [TK]D-Fender | Yeah. Ok. Fine. Sure. |
| 20:36.39 | *** join/#asterisk mascool (n=george@c-68-43-139-85.hsd1.mi.comcast.net) |
| 20:37.37 | mascool | can anyone tell me how I can list the number/address of a customer in to the public directory (555-1212) ? My provider doesn't offer it |
| 20:38.05 | [TK]D-Fender | mascool: as they own the # my first guess is you're screwed |
| 20:38.27 | mascool | damn |
| 20:38.38 | mascool | I feared that was the case |
| 20:39.42 | mascool | thanks [TK]D-Fender |
| 20:40.18 | mascool | oh another thing, you know where I can find some samples of SER + asterisk configs ? |
| 20:40.49 | Runlvl | Asterisk Community in Spanish, just visit http://www.asterisk-la.org - The Asterisk Latin America Team =) |
| 20:41.33 | gvasterisk | who asked for spanish? |
| 20:41.46 | draygon | hrm? |
| 20:42.13 | *** join/#asterisk agx (n=badpengu@81-174-8-217.dynamic.ngi.it) |
| 20:42.50 | *** join/#asterisk jetlagmk2 (n=jetlag@pool-151-204-7-155.pskn.east.verizon.net) |
| 20:43.36 | Runlvl | gvasterisk, any one, i'm just talking ;) |
| 20:43.53 | [TK]D-Fender | SPAM.... its not just a luncheon meat..... |
| 20:44.52 | *** join/#asterisk Schumie (i=SteveWri@cpc2-rdng2-0-0-cust382.winn.cable.ntl.com) |
| 20:45.50 | jameswf | spam is good, especialy the grease layer |
| 20:45.56 | *** join/#asterisk IOscanner (n=IOscanne@216.88.109.2) |
| 20:46.11 | Runlvl | come on :P |
| 20:46.15 | [TK]D-Fender | ~asteriskspanish |
| 20:46.15 | jbot | [~asteriskspanish] Asterisk Community in Spanish, just visit http://www.asterisk-la.org -=- IRC channel #asterisk-es |
| 20:46.18 | [TK]D-Fender | There |
| 20:46.32 | [TK]D-Fender | now everyone can ask for that when somebody CARES. |
| 20:46.50 | mascool | ~where'smylighter |
| 20:46.52 | [TK]D-Fender | Runlvl: Now please stop spamming :) |
| 20:46.53 | Runlvl | [TK]D-Fender, thanks (= |
| 20:47.00 | IOscanner | anyone know how asterisk handles DNS lookups. I am trying to find out if it will lookup an address when the TTL has expired. |
| 20:47.04 | Runlvl | Okey! :P |
| 20:47.52 | gvasterisk | Runlvl you speak spanish? |
| 20:48.10 | ManxPower | IOscanner: don't assume that Asterisk even knows about the DNS TTL. |
| 20:48.12 | Runlvl | gvasterisk, yes , i can |
| 20:48.14 | lirakis | _. will match ANY extension correct? |
| 20:48.26 | Qwell | lirakis: no, but it'll match everything useful (and then some) |
| 20:48.29 | Runlvl | if i want to |
| 20:48.44 | ManxPower | lirakis: Yes, including h, i, T, t extensions, which you DO NOT WANT. |
| 20:48.59 | IOscanner | I have a carrier that based fail-over on DNS. Trying to find out if there is away to deal with it other then a reload. |
| 20:49.15 | ManxPower | for example if you did an exten => _.,1,Dial(Zap/g1/555121) then when you hang up, asterisk will jump to exten "h" and redial the call. |
| 20:49.36 | lirakis | ManxPower: ... hmm okay .. sooo.. i want to match all dialable numbers.. NANP, ILD, etc. |
| 20:50.01 | ManxPower | IOscanner: as I'm sure you know, Asterisk's support of DNS is abysmal. |
| 20:50.18 | ManxPower | lirakis: exten => _XX.,1,Dial(.... |
| 20:50.34 | lirakis | ManxPower: got it... thanks |
| 20:50.36 | ManxPower | as the shortest dialable number in NANPA (other then 0) is 3-digits. |
| 20:50.37 | IOscanner | I was not sure. I was hoping not |
| 20:51.38 | mascool | so anyone that can help me out with some info on SER + asterisk? :D |
| 20:51.44 | ManxPower | if you don't care about 911, 411, etc, then you could use exten => _XXXXXX.,1,Dial(.... |
| 20:52.14 | ManxPower | of course, if you are using zap channels that will add a significant delay to all calls to start ringing. |
| 20:54.12 | ManxPower | mascool: your best bet is google, not many people use SER on this channel |
| 20:54.15 | *** join/#asterisk Voicemeup (n=VoiceMeU@modemcable132.108-83-70.mc.videotron.ca) |
| 20:54.17 | freezey | <PROTECTED> |
| 20:54.19 | *** part/#asterisk Voicemeup (n=VoiceMeU@modemcable132.108-83-70.mc.videotron.ca) |
| 20:54.19 | freezey | the main fax line called "FaxMaster" is 7010 and thats the only extension i notice inside of asterisk none of the other ones are in their 7012 7011 7017 none of them |
| 20:54.50 | ManxPower | freezey: why are you telling us this? |
| 20:54.52 | mascool | ManxPower, thanks I've checked google, couldn't find too much info about this |
| 20:55.33 | freezey | ManxPower: cause i am having a problem that when i call in for the exentsion it continues to just ring and the fax line does not pick it up |
| 20:55.44 | freezey | ManxPower:L i am pretty sure i am narrowed it down to an asterisk problem |
| 20:56.08 | freezey | ManxPower: cause sometimes when i ring the fax line from my lan line let it ring and then call from my cellphone i can hit the fax line and then others no |
| 20:56.10 | ManxPower | You have a lot more Asterisk trouble shooting to do before asking here. |
| 20:56.25 | De_Mon | pissh |
| 20:56.46 | ManxPower | you need to watch the CLI for a failed call, put up a pastebin if the CLI output, put up a pastebin of the relevant parts of extensions.conf, etc. |
| 20:57.06 | freezey | whos that towards? |
| 20:57.14 | ManxPower | freezey: you |
| 20:57.20 | tzanger | bleh |
| 20:57.33 | ManxPower | you don't even know that the call was sent to the fax from Asterisk, do you? |
| 20:57.42 | freezey | ok |
| 20:58.05 | ManxPower | I'll bet that the problem is you are doing a blind dial and never checking the status of DIALRESULT to provide a busy to the caller. |
| 20:58.31 | freezey | ManxPower: no i am not sure if it was sent with asterisk but i know its being routed through the asterisk box and none of those extensions are sittin in extensions.conf |
| 20:58.58 | ManxPower | freezey: well find out for sure. |
| 20:59.00 | freezey | ManxPower: i am not sure exactly whats going on i kjust know i have been troubleshooting hylafax all day... how exactly can i checlk the output ffrom the asterisk cli and how do i get to that |
| 20:59.27 | ManxPower | freezey: you are here, therefore you are trying to work with asterisk. now, what does the CLI show for a failed call??? |
| 20:59.42 | freezey | i am not sure on what your asking me to look at tho |
| 20:59.47 | Shaun2222 | [TK]D-Fender: around? |
| 20:59.56 | ManxPower | freezey: you did not install asterisk, did you? |
| 21:00.09 | ManxPower | freezey: do you know what the Asterisk CLI is? |
| 21:00.17 | [TK]D-Fender | Shaun2222: Yes... |
| 21:00.56 | Shaun2222 | [TK]D-Fender: getting a PRI T1 from TW, they are asking me if i want associated signaling or non-facilty associated signaling.. with that 101d card what should i tell them. |
| 21:01.27 | ManxPower | Shaun2222: you want associated signalling, as NFAS (non-facility) makes no sense unless you have 2 or mote PRIs |
| 21:01.36 | freezey | ManxPower: i didnt install it myself but i do know the asterisk cli |
| 21:01.45 | freezey | ManxPower: well howto get their atleast |
| 21:01.56 | ManxPower | freezey: and you know that the CLI prints out everything that happens in asterisk for every call, right? |
| 21:02.18 | freezey | didnt know that no |
| 21:02.19 | ManxPower | so get to the cli, try to reproduce the problem, copy the CLI output and put it on pastebin.ca for people to look at. |
| 21:02.29 | ManxPower | freezey: then you need to read The Book first. |
| 21:02.31 | ManxPower | ~book |
| 21:02.32 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 21:03.02 | Shaun2222 | ManxPower: whats the difference between these two and why would one want it vs not wanting it |
| 21:03.04 | ManxPower | you are like a 4-year old trying to drive a car. you don't even know enough about cars to start one up. |
| 21:03.12 | asdx | nice, i've got a vps :-) |
| 21:03.17 | asdx | horray! |
| 21:03.28 | Shaun2222 | with who? |
| 21:03.34 | asdx | slicehost |
| 21:03.38 | ManxPower | Shaun2222: NFAS lets you use one D-channel for than one PRI. Of course, that means that if the D-channel goes down, all PRIs that use the D-channel will go down. |
| 21:03.46 | freezey | ManxPower: ok so whast the command to see the output? |
| 21:03.59 | ManxPower | freezey: the command "is go read the book" |
| 21:04.05 | ManxPower | when you are done with the book, then come back. |
| 21:04.25 | Shaun2222 | asdx: havnt been aroudn very long from the looks of it.. |
| 21:04.40 | freezey | ManxPower: learn better with experience then reading a book |
| 21:04.59 | Mw3 | then get some experience :) |
| 21:05.29 | ManxPower | freezey: Tough. Read the book. |
| 21:05.31 | freezey | and thats what i am doing here trying to ask questions to troubleshoot issues that in the end will result in me having more experience using the system |
| 21:05.54 | ManxPower | freezey: and what *I* am saying is that you don't even know enough about asterisk to ask valid questions. |
| 21:05.57 | Shaun2222 | ManxPower: can you have more than one D channel on a PRI? i really dont need 23 voice channels, would there be a benifit to having more D channels.. |
| 21:05.57 | *** part/#asterisk sheldonh (n=sheldonh@66.219.59.32) |
| 21:06.21 | draygon | Shaun2222, get back to cpanel! |
| 21:06.21 | ManxPower | Shaun2222: the max number of D-channels per PRI is 1. The max PRIs per d-channel is 1 or more. |
| 21:06.56 | freezey | ManxPower: ok hows this question... whast the command so i can watch the output of a failed call via the asterisk CLI |
| 21:07.14 | ManxPower | freezey: the most basic asterisk command of all "asterisk -rvvv" |
| 21:07.32 | freezey | ManxPower: now was that dificult to answer? |
| 21:07.33 | Shaun2222 | draygon: :) |
| 21:07.39 | ManxPower | but I will not help you further, as you refuse to read. |
| 21:07.44 | ManxPower | I am putting you on /ignore. |
| 21:07.45 | jameswf | my favorite new command is originate.... |
| 21:08.20 | freezey | ManxPower: i will read it i am just saying i am not gonna read a 500 page book when i have a problem as we speak |
| 21:08.24 | jameswf | with a simple loop and a bach script 50 phones ringing and screaming like monkeys :) |
| 21:08.33 | Shaun2222 | ManxPower: you talking to me about the ignore or sombody else.. |
| 21:08.41 | jameswf | *bash |
| 21:08.52 | [TK]D-Fender | Shaun2222: You're a little backwards. NFAS is for letting you use 1 D channel across multiple PRI's instead of 1 each allowing you to RECOUP 1 channel for use as a B for each supplemental PRI added. This of course increases your risk of failure. |
| 21:08.56 | ManxPower | Shaun2222: well you are not being an idiot so there is no reason for me to /ignore you. |
| 21:09.01 | Qwell | jameswf: look at russell's res_monkeys branch |
| 21:09.08 | [TK]D-Fender | Shaun2222: And you don't even WANT that, let alone need a FULL PRI as it is. |
| 21:09.44 | twisted | Qwell: dare I ask? |
| 21:09.55 | Qwell | twisted: randomly injects monkeys into calls. |
| 21:10.02 | Qwell | ...to ONLY one side |
| 21:10.02 | twisted | lolol |
| 21:10.08 | twisted | i need that. |
| 21:11.03 | Qwell | "WTF was that?" "What was what?" "The monkeys!" "wtf are you talking about?" "You didn't hear that?" "You...hear...monkeys?" |
| 21:11.03 | Shaun2222 | [TK]D-Fender: ok, so i want FAS and not NFAS. |
| 21:11.03 | [TK]D-Fender | Qwell: Don't need a brach for that... easy enough to make a daemon to scan for a channel and chan_local a Chanspy w/ whisper into it ;) |
| 21:11.25 | De_Mon | mmm chanspy whisper, I need to play with that |
| 21:11.30 | [TK]D-Fender | Qwell: ....not that I would ever perpetrate such heinousness ;) |
| 21:11.33 | twisted | Qwell: i don't see that branch |
| 21:11.38 | *** part/#asterisk thix_rj_br (n=Basic@c934c2fc.virtua.com.br) |
| 21:12.52 | Qwell | maybe he committed it to trunk, heh |
| 21:13.09 | TJNII | Hmmmm... chanspy whisper. I'll keep that in mind if the telemarketers ever start getting through |
| 21:13.10 | Qwell | I forget what it was called |
| 21:13.20 | twisted | i don't see a res_monkeys in trunk :P |
| 21:13.25 | jameswf | res_monkeys looks like what I wanted to put together... anyone who is familiar with the multix cookie monster app.... |
| 21:13.29 | *** join/#asterisk slowshutt (n=d@196.211.34.2) |
| 21:13.35 | file | he did not commit it |
| 21:13.38 | slowshutt | hi there people |
| 21:14.22 | slowshutt | anyone tried installing asterisk 1.4 on a clack connect box using centos? |
| 21:14.45 | slowshutt | sorry thats a clarkconnect box |
| 21:15.34 | shido6 | "Sucka Free" |
| 21:15.38 | slowshutt | how important is bison-devel? |
| 21:15.52 | jameswf | mmmm buffalo meat |
| 21:16.06 | slowshutt | :> |
| 21:16.32 | [TK]D-Fender | jameswf: I've been having fillet mignon all week now :) |
| 21:16.38 | slowshutt | anyone? |
| 21:16.49 | De_Mon | I've setup a small call center with asterisk an asterisk queue of dynamic members. If nobody answers the call goes to a MeetMe conference and a remote callcenter is joined to the conference. |
| 21:17.24 | slowshutt | well done i'm still trying to load asterisk |
| 21:17.50 | De_Mon | I want call the remote callcenter into a local channel since it goes into their own queue system, and when an agent answers, have some sort of dialog of "Press 1 to accept this call" |
| 21:19.08 | De_Mon | I'm not really sure how to get music on hold / frequent announcements of the accept this call message to work |
| 21:19.34 | De_Mon | and then join that channel to the customer... |
| 21:24.07 | slowshutt | De_Mon: hi there seems like you know your way around, mind helping me please, can't see to install bison-devel on my clarkconnect box, one of sterisk dependencies. Do I need it ? |
| 21:24.14 | De_Mon | lol |
| 21:24.23 | slowshutt | ? |
| 21:24.26 | slowshutt | why lol |
| 21:24.52 | De_Mon | sorry, I was just thinking that I scared everyone off and then you piped up |
| 21:25.29 | De_Mon | if its a dependancy, it is probably required. |
| 21:25.30 | slowshutt | do you think i need bison-devel? |
| 21:25.52 | slowshutt | k i'm of to the linux channel |
| 21:25.53 | De_Mon | You could always compile asterisk on a different machine and then copy the installation over to the other server |
| 21:25.56 | jameswf | (yum|apt-get) install bison-devel |
| 21:26.19 | fujin | uhgh |
| 21:26.23 | fujin | compile on a different machine? |
| 21:26.25 | slowshutt | tried does not work |
| 21:26.26 | fujin | that's a bad idea -_- |
| 21:26.39 | jameswf | no if the kernel is the same... |
| 21:26.50 | jameswf | what do you think RPM's are |
| 21:26.53 | De_Mon | a bad idea is loading a compiler ona production machine with internet acccess |
| 21:26.55 | fujin | kernel isn't really the problem, it's the rest of the toolchain!~ |
| 21:27.20 | De_Mon | one bad apache page and your box is ownzored |
| 21:27.27 | jameswf | most apps are built on other machines.... |
| 21:27.41 | slowshutt | i just want to follow the instructions but can't seem to find bison-devel if i google it i get some info on fedora 4 i am running centos |
| 21:28.22 | jameswf | unless you use a source distro like gentoo |
| 21:28.23 | De_Mon | slowshutt so your looking something like this page? |
| 21:28.24 | De_Mon | http://www.asteriskguru.com/tutorials/asterisk_installation_compilation_centos.html |
| 21:28.42 | slowshutt | yes De_Mon |
| 21:29.30 | De_Mon | that rpm -q bison-devel command doesnt work? |
| 21:29.41 | De_Mon | I dont use centos so... |
| 21:30.05 | slowshutt | nope all the other ones work |
| 21:30.23 | *** join/#asterisk km- (n=pgrace@aeneas.fierymoon.com) |
| 21:30.37 | slowshutt | i am a suse guy, but one client wants asterisk on this sever for testing |
| 21:30.49 | jameswf | suse... ick |
| 21:30.49 | De_Mon | so the next paragraph down says "if yum is unable to find one or more of these packages, try your luck with PBone (whatever that is) |
| 21:30.56 | De_Mon | this is a question for #centos really |
| 21:31.10 | slowshutt | k here i go |
| 21:31.24 | jameswf | dag is pretty good |
| 21:32.53 | robl^ | hrmmmm |
| 21:33.49 | robl^ | is it possible to emulate a Nortel Meridian XMSG key on Asterisk with a SIP phone that supports BLF? |
| 21:34.09 | *** part/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 21:34.52 | [TK]D-Fender | robl^: Which does what exactly? |
| 21:37.12 | [TK]D-Fender | ok, heading home, BBIAB |
| 21:37.24 | robl^ | [TK]D-Fender: sorta a secondary Message Waiting Indicator. Say a phone has 2 distinct SIP registrations. Line 1 (primary line) would use the phone's built-in MWI for a VM. Line 2 (secondary extension), would have an XMSG key to show if VM was waiting for the second line. It would light-up BLF-ish. |
| 21:37.31 | robl^ | now he leaves. ;-) |
| 21:37.35 | *** join/#asterisk gammacoder (n=chatzill@64.132.192.41) |
| 21:38.53 | *** part/#asterisk km- (n=pgrace@aeneas.fierymoon.com) |
| 21:40.24 | *** join/#asterisk angom (n=angom@200.38.31.239.dsl.dyn.telnor.net) |
| 21:46.02 | peanut- | anyone ever buy something from voipsupply.com and have them tack on random charges to your card? |
| 21:46.12 | Qwell | peanut-: call them on it |
| 21:46.17 | Qwell | they screw up - a lot |
| 21:46.28 | Qwell | they sent my friend a second TE220B the other day. |
| 21:46.54 | peanut- | I did call them, they're working on it |
| 21:47.11 | peanut- | instead of a $62 charge, I have a $62 $5 and $57 charge |
| 21:47.17 | jameswf | peanut-: were you charged for a te220b lol |
| 21:47.37 | peanut- | last time I ordered from them, someone else got my card number and put a $5 charge on it |
| 21:48.04 | jameswf | $5 is for the sales persons starbucks |
| 21:48.36 | robl^ | $5 for starbucks?!?!?1 I can't get out of there for less than $5.50 |
| 21:49.11 | jameswf | could buy from attacom.... oh wait |
| 21:49.20 | peanut- | so nobody else has seen randomass charges from voipsupply? |
| 21:50.53 | robl^ | I've never purchased from voipsupply. I have used Attacom (no more) and Telephony Depot. No trouble with either. I've purchased from Telephony Depot at least 15 times, never a problem. |
| 21:54.53 | peanut- | it's freaking weird |
| 21:58.05 | draygon | peanut-, call your bank and do a chargeback. |
| 21:58.51 | tzanger | peanut-: I've bought tons of stuff from them... never a problem |
| 21:59.10 | *** join/#asterisk ginvent (n=ginvent@76.203.119.54) |
| 22:00.01 | ginvent | Newbie question. I installed via svn and my version is 1.4 r88116. Is that the latest like 1.4.13? |
| 22:00.30 | ginvent | Or should I download and build 1.4.13? |
| 22:00.42 | *** part/#asterisk grtrfgthy (n=dddddd@123-2-65-142.static.dsl.dodo.com.au) |
| 22:01.01 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 22:03.29 | ginvent | Hmm, not sure how to interpret if I have the latest version. |
| 22:04.09 | ginvent | svn should have gotten the latest... but how do I know? |
| 22:07.00 | file | ginvent: if you are using the 1.4 branch you are using a version *newer* than 1.4.13, the branch keeps getting fixes... and will eventually be tagged as 1.4.14 and life will go on |
| 22:07.45 | *** join/#asterisk frenzy (n=frenzy@unaffiliated/frenzy) |
| 22:08.21 | Shaun2222 | dont do a charge-back, thats going to cost them a fee... try to work it out with them first, if they dont want to fix it then charge them back. |
| 22:08.27 | ginvent | Ah ok... thanks... |
| 22:08.43 | ginvent | Anyone use sipphone? I keep getting Packet2Packet bridging SIP/250-0822fd78 and SIP/sipphone-out-082444d0 and no audio |
| 22:08.51 | Shaun2222 | i've used them before but i think i used paypal each time i bought somthing, cant remember |
| 22:09.00 | file | ginvent: are you behind NAT? |
| 22:09.08 | ginvent | Not that I know of. |
| 22:09.30 | file | k, rtp debug will show you packets going/coming |
| 22:09.36 | ginvent | My server is behind a linksys router... should not be nat. |
| 22:10.31 | peanut- | I'd have to cancel my debit card if I said it was an unauthorized charge |
| 22:10.50 | ginvent | Hmmm... |
| 22:10.51 | peanut- | and I just did that because of the last time I dealt with them and someone other company charged $5 to my card |
| 22:10.58 | ginvent | Sent RTP P2P packet to 198.65.166.131:43714 (type 00, len 000160) |
| 22:11.00 | *** join/#asterisk Alowishus (n=jpenix@fwsdo.projectdesign.com) |
| 22:11.15 | ginvent | Does that mean it is not routing properly? |
| 22:11.47 | file | nope, it means it sent an RTP packet to 198.65.166.131 port 43714 |
| 22:12.18 | ginvent | and then to the local 202... which is the phone. |
| 22:12.33 | ginvent | why is it going direct to phone? |
| 22:12.49 | file | if you are seeing rtp debug on the screen, it means it isn't... it's going through your Asterisk |
| 22:12.58 | ginvent | Ah... duh... sorry |
| 22:13.01 | slowshutt | looks like bison and bison-devel is one and the same thing |
| 22:13.03 | file | and if you see packets going back and forth, it means audio is flowing through Asterisk at least |
| 22:13.19 | *** join/#asterisk mitcheloc (n=mitchel@adsl-68-120-227-239.dsl.irvnca.pacbell.net) |
| 22:13.30 | ginvent | weird... so why don't I hear anything... teliax is fine... zaptel is fine... sipphone no audio. |
| 22:13.40 | ginvent | It's registered... |
| 22:15.13 | slowshutt | i have the error no 'modules.conf'found,no modules will be loaded help please |
| 22:18.03 | slowshutt | cd / |
| 22:18.04 | [TK]D-Fender | if you have installed * for the first time you should do "make samples". |
| 22:18.12 | slowshutt | sorry wrong mashine |
| 22:20.25 | slowshutt | thx [TK]D-Fender |
| 22:20.30 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 22:20.31 | *** mode/#asterisk [+o anthm] by ChanServ |
| 22:20.31 | slowshutt | asterisk up and running |
| 22:20.44 | slowshutt | \./ |
| 22:21.00 | *** join/#asterisk unixdog (n=unixdog@adsl-69-234-184-228.dsl.irvnca.pacbell.net) |
| 22:21.09 | unixdog | never seen this before need input |
| 22:21.13 | unixdog | [Nov 7 06:08:30] WARNING[2764]: pbx.c:5878 pbx_builtin_setvar: Ignoring entry '0' with no = (and not last 'options' entry) |
| 22:21.15 | De_Mon | how can I get background music, and intermitant anouncements prompting user to press a key? |
| 22:21.45 | *** join/#asterisk xBull (n=sabalo@190.3.89.131) |
| 22:22.05 | De_Mon | unixdog nothing before that error like maybe what config it was looking at? |
| 22:22.15 | ginvent | Argh, I get one way audio on sipphone... I know I have seen that before... but where. |
| 22:22.23 | De_Mon | nat |
| 22:22.23 | ginvent | My audio goes out, but not in. |
| 22:22.39 | ginvent | Yes, but how do I check/set that? |
| 22:23.02 | De_Mon | sip.conf |
| 22:23.15 | *** join/#asterisk Innatech (n=it@remote.eb-lawyers.com) |
| 22:24.45 | ginvent | I am lost... how do I have to have nat setup? |
| 22:25.07 | *** part/#asterisk robl^ (n=robl@pdpc/supporter/monthlybyte/robl) |
| 22:25.22 | Innatech | It depends on what you want to do. |
| 22:25.25 | TJNII | ~sipnat |
| 22:25.32 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 22:25.54 | ginvent | How do I know if I am behind nat? It's just a linksys router. |
| 22:26.10 | TJNII | ginvent: If you don't know you probably are |
| 22:27.04 | Innatech | if you have a 192.168.x.x IP and you can see the Internet, you're behind NAT> |
| 22:27.11 | tzafrir_home | cpina, here? |
| 22:27.19 | Innatech | (or any of the other private IP ranges) |
| 22:29.00 | ginvent | Ah... ok... |
| 22:29.20 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 22:29.30 | unixdog | http://www.pastebin.ca/765284 |
| 22:29.36 | unixdog | thats the full dial plan |
| 22:29.56 | unixdog | for 1 route |
| 22:30.11 | unixdog | simple mexico rout |
| 22:30.12 | unixdog | e |
| 22:31.13 | [TK]D-Fender | unixdog, exten => s,n,Set($["${OUTMAXCHANS_${DIAL_TRUNK}}foo" = "foo"]) <- WTF |
| 22:31.46 | *** join/#asterisk Seldon75 (n=chatzill@69.77.161.3) |
| 22:31.49 | Seldon75 | hello, |
| 22:31.51 | [TK]D-Fender | unixdog, and next time also show your codes EXECUTION so we don't have to GUESS where the error occured |
| 22:31.52 | Seldon75 | how are we all? |
| 22:33.08 | Seldon75 | I have an issue with MeetMe holding lines open. If I call into a MeetMe teleconf both internally and externally and then hang up both lines Asterisk still has the following channels occupied |
| 22:33.21 | Seldon75 | asterisk*CLI> Zap/6-1 333@incoming:1 Up MeetMe(|MIxp) |
| 22:33.23 | Seldon75 | asterisk*CLI> Zap/pseudo-57250690 s@from-internal:1 Rsrvd (None) |
| 22:33.26 | unixdog | it sets how many channels they get |
| 22:33.34 | unixdog | we linkit them to 100 channels |
| 22:33.46 | unixdog | but I guess I missed something |
| 22:33.52 | *** part/#asterisk zerohalo (n=zeroHalo@pool-71-255-162-167.bstnma.east.verizon.net) |
| 22:33.55 | Seldon75 | does anyone have any idea why the lines wouldnt close? |
| 22:33.59 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:38.24 | TJNII | Seldon75: I don't know too much about MeetMe, but what little I read led me to believe it might hold a room open for people to come and go whenever. There isn't a setting for that, is there? |
| 22:42.07 | Seldon75 | TJNII: I can understand it holding a 'room' open but surely it shouldnt tie up one of our Zap channels indefinitely |
| 22:42.54 | [TK]D-Fender | Seldon75, if that line is up and it shouldn't be then its a call disconnect detection issue and nothing to dow ith MeetMe. |
| 22:43.44 | penguinFunk_ | Seldon75: if noone is connected you will not be using any Zap channels |
| 22:43.46 | hmmhesays | is there anything I should be looking for in the zaptel configuration to allow for answer supervision? |
| 22:43.52 | *** join/#asterisk macros73_ (n=cs@c-24-3-246-27.hsd1.mn.comcast.net) |
| 22:44.10 | Seldon75 | it seems to only happen with MeetMe |
| 22:44.20 | penguinFunk_ | i use meetme |
| 22:44.40 | penguinFunk_ | if the room is empty all zap and sip channels are empty (if noone is on the phone too of course) |
| 22:45.35 | Seldon75 | ok, any suggestions for diagnosing why the line is still up (before I forcibly close it)? |
| 22:46.11 | tzanger | perhaps dumb question... is there a dialplan var that will tell me which zap channel I am? |
| 22:46.26 | tzanger | i.e. I dial 500 and it says "you're calling from four oh" or osmething if I'm on Zap/40 |
| 22:46.29 | tzanger | ? |
| 22:46.34 | *** join/#asterisk diclophis-work (n=jbardin@65.203.37.58) |
| 22:46.43 | diclophis-work | hello all |
| 22:46.56 | diclophis-work | has anyone ever used AT&Ts "IP Flexible Reach" service with Asterisk? |
| 22:50.02 | Seldon75 | tzanger: you could implement in your dialplan |
| 22:50.11 | Seldon75 | but I dont think there's a dedicated function |
| 22:50.35 | tzanger | Seldon75: how do I implment it is the question |
| 22:50.41 | tzanger | I can do it if I know of a var |
| 22:51.09 | Seldon75 | ah, I see |
| 22:51.17 | Seldon75 | sorry, dont know the var |
| 22:51.22 | Seldon75 | thre should be one though |
| 22:51.35 | Seldon75 | [TK]D-Fender could tell you I'm sure |
| 22:51.55 | *** join/#asterisk TrentCreek (n=tjones14@cpe-70-117-207-168.rgv.res.rr.com) |
| 22:52.30 | [TK]D-Fender | tzanger, ${CHANNEL} <- |
| 22:55.30 | tzanger | [TK]D-Fender: yeah... gotta chop off the Zap/ |
| 22:55.36 | tzanger | ${CHANNEL:-4}? |
| 22:55.48 | [TK]D-Fender | tzanger, "show fuunction CUT" <- :) |
| 22:55.54 | [TK]D-Fender | tzanger, "show function CUT" <- :) |
| 22:56.57 | tzanger | hmm |
| 22:57.45 | De_Mon | actually I switched to CUT |
| 23:00.32 | *** join/#asterisk jailshell (i=silas@otaku.freeshell.ORG) |
| 23:01.54 | tzanger | exten => 500,n,Set(A=${CUT(${CHANNEL},/,2)}) |
| 23:01.54 | tzanger | exten => 500,n,Set(B=${CUT(${B},-,1)}) |
| 23:01.54 | tzanger | exten => 500,n,SayDigits(${B}) |
| 23:01.59 | tzanger | what the hell am I doing wrong? |
| 23:02.07 | tzanger | Zap/5-1 for instance |
| 23:02.18 | tzanger | field 2 would be 5-1 |
| 23:02.23 | De_Mon | I've got a .call file that goes to a remote queue. Asterisk loops... |
| 23:02.23 | De_Mon | exten => _dial-.,3,Set(SIP_USER=${CUT(CHANNEL,/,2)}) |
| 23:02.31 | tzanger | oh |
| 23:02.36 | De_Mon | you have too many ${}'s for your channel |
| 23:03.16 | tzanger | yeah I thought you gave it the text to cut |
| 23:03.24 | [TK]D-Fender | tzanger, you pass cut a VARIABLE, not the RESULT of a variable. |
| 23:03.25 | De_Mon | When the .call is picked up by a real person they press 1, now I want to add that local channel to a dynamic queue |
| 23:03.44 | *** join/#asterisk tripps (n=ss@66.60.235.100) |
| 23:03.54 | De_Mon | I need to just write this and see what happens |
| 23:04.41 | De_Mon | no, see if I addqueuemember() its not going to throw the current caller into the queue |
| 23:04.47 | IPetrov2 | exten => 500,n,Set(B=${CUT(${A},-,1)}) |
| 23:04.56 | tzanger | yeah I got that one :-) |
| 23:04.59 | tzanger | thank you guys, it's now working |
| 23:05.00 | IPetrov2 | :) |
| 23:05.22 | De_Mon | its not ${A} its A |
| 23:05.33 | IPetrov2 | hehe |
| 23:06.00 | tzanger | got a 580 port zap system |
| 23:06.07 | tzanger | it's a little daunting to figureo ut which phone is which port |
| 23:06.17 | tzanger | soon it'll be a 1200 port zap system |
| 23:06.25 | De_Mon | label them! |
| 23:06.40 | tzafrir_home | tzanger, this is why we added xpp_blink |
| 23:06.48 | De_Mon | or, better yet, replace it with sip hahaha |
| 23:07.26 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 23:09.19 | De_Mon | so, how do I create an extension that adds a caller into queue as a dynamic member? |
| 23:09.28 | JT | why would you put all the ports on the one machine? |
| 23:10.15 | De_Mon | I think the rub is that its a channel not a technology\resource |
| 23:11.07 | De_Mon | (interface) |
| 23:11.51 | De_Mon | thanks for listening it really helped ;) |
| 23:12.08 | [TK]D-Fender | De_Mon, just AgentLogin them <- |
| 23:13.31 | tzanger | tzafrir_home: xpp_blink? |
| 23:13.38 | De_Mon | hrm, thats an interesting idea. I dumped all my agent code in favor of dynamic queue members |
| 23:13.40 | tzanger | De_Mon: I was going ot make them SIP |
| 23:13.49 | tzanger | but we went TDMoE insteead |
| 23:14.09 | *** join/#asterisk craigk (n=ckowald@58.174.122.198) |
| 23:14.10 | De_Mon | tzanger interesting, didn't know such a thing existed |
| 23:14.20 | tzanger | De_Mon: it doesn't, yet |
| 23:14.36 | tzafrir_home | perldoc xpp/util/xpp_blink |
| 23:14.47 | tzafrir_home | in zaptel |
| 23:15.27 | tzafrir_home | though it is specific to our device |
| 23:16.06 | De_Mon | tzanger how do you hook TDMoE into asterisk? |
| 23:16.24 | tzanger | tzafrir_home: nice! |
| 23:16.29 | tzanger | De_Mon: ztdynamic |
| 23:16.33 | tzanger | ztd-eth specifically |
| 23:16.46 | *** join/#asterisk MaliutaWrk (n=nikolai@fw.hitwise.com) |
| 23:16.47 | tzanger | zaptel for blackfin DSP sucks mightily, so I had to write my own driver |
| 23:17.23 | [TK]D-Fender | De_Mon, other freak idea. park the call and add a local channel as a device that will pickup the parked agent. |
| 23:17.24 | De_Mon | is it a switch and you just hook * up to it and tell it to use ztd-eth? |
| 23:17.28 | *** join/#asterisk tristanbob (n=tristan@oalug/member/tristanbob) |
| 23:17.42 | tzanger | De_Mon: kind of |
| 23:17.53 | tzanger | had some trouble iwth that many channels and switches so they're directly connected right now |
| 23:19.10 | tzanger | they've ordered in some better switches, so we'll see |
| 23:19.15 | tzanger | 1200 channels might require gige though |
| 23:19.29 | De_Mon | [TK]D-Fender I like the agentlogin idea, just not sure if I want to resurect chan_agent for it |
| 23:20.22 | [TK]D-Fender | De_Mon, doesn't use chan_agent IIRC. the active channel simply gets bridged w/o a "dial" per-se |
| 23:20.47 | [TK]D-Fender | De_Mon, note I did not say callbackagentlogin |
| 23:20.56 | De_Mon | agentLogin does tho |
| 23:21.03 | putnopvut | Actually AgentLogin does use chan_agent. |
| 23:21.17 | De_Mon | are you talking about the other crazy idea (that I don't fully understand) |
| 23:21.35 | [TK]D-Fender | Well either way.... |
| 23:21.48 | [TK]D-Fender | De_Mon, go try something :) |
| 23:21.57 | De_Mon | if I can do it without chan_agent I'll try it in a heartbeat |
| 23:22.37 | De_Mon | Oh, i re-read your second crazy idea and it would potentially work without any agents |
| 23:22.56 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 23:22.58 | De_Mon | thats one helluva crazy idea |
| 23:23.23 | De_Mon | add a local channel that picks up a parked call... |
| 23:23.34 | *** join/#asterisk metfan2007 (n=metfan20@189.135.178.159) |
| 23:24.04 | [TK]D-Fender | De_Mon, yeah, you just have to think about all the ways you have it "dial" to pick up an EXISTING channel. |
| 23:24.07 | De_Mon | you deserve a crazy genius award for this |
| 23:24.12 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 23:24.39 | [TK]D-Fender | De_Mon, don't forget the paper-clip, chewing gum, or Swiss Army Knife, ok? ;p |
| 23:24.54 | metfan2007 | Hi, Asterisk cannot find the sound files (gsm) inside digits folder, if I copy the sound file to the parent folder and do a playback(1M) it works ok, if I do a playback(digits/1M) it says that the file is missing.... any idea?? pls! |
| 23:25.14 | De_Mon | metfan2007 check folder permissions |
| 23:25.23 | [TK]D-Fender | metfan2007, pastebin the attempts and folder dumps "ls -l" |
| 23:25.52 | *** join/#asterisk RageMax (n=RageMax@pool-72-77-120-39.pitbpa.east.verizon.net) |
| 23:26.15 | *** join/#asterisk r0d3nt (n=astrutt@foster.stonedcoder.org) |
| 23:27.06 | *** join/#asterisk blq (n=Bl@dslb-088-064-133-223.pools.arcor-ip.net) |
| 23:39.36 | metfan2007 | [TK]D-Fender: http://pastebin.com/m4b95884e |
| 23:39.43 | metfan2007 | De_Mon: http://pastebin.com/m4b95884e |
| 23:40.59 | [TK]D-Fender | metfan2007, and where do I see that FILE? |
| 23:41.18 | metfan2007 | inside digits |
| 23:42.29 | metfan2007 | [TK]-Fender: http://pastebin.com/m1d0bb84 |
| 23:45.31 | *** part/#asterisk Chiko_Medina (n=kristian@vps.oficinasdigitales.com) |
| 23:46.11 | metfan2007 | [TK]D-Fender: http://pastebin.com/m1d0bb84 |
| 23:48.55 | *** join/#asterisk asdx (n=diego@adsl-158-235.click.com.py) |
| 23:54.08 | *** join/#asterisk r0d3nt (i=nobody@wsip-24-234-235-73.lv.lv.cox.net) |
| 23:54.09 | De_Mon | I notice that your language is 'es' instead of 'en' would that change the path at all? |
| 23:54.24 | De_Mon | I haven't delt with non-en ivr's |
| 23:56.20 | *** join/#asterisk linxroute (i=linxrout@117.0.26.118) |
| 23:56.29 | metfan2007 | De_Mon: is spanish PBX |
| 23:57.59 | De_Mon | from just reading about languages, I think asterisk will try sounds/es/silence/1M.gsm? I would expect it to fall back on sounds/silence/1M though... |
| 23:58.12 | Qwell | De_Mon: it should, yes |
| 23:58.38 | dijungal | what would cause degrading audio on a call over time? |
| 23:58.47 | dijungal | through and IAX trunk between two servers |
| 23:58.49 | Qwell | without looking at the pastebin, or the problem - languageprefix=yes will probably help |
| 23:59.10 | De_Mon | sorry I ment digits |
| 23:59.16 | Toerkeium | guys, a cheap & good ip phone? anyone? |
| 23:59.37 | Mw3 | linksys spa921 |
| 23:59.40 | Mw3 | maybe |
| 23:59.52 | linxroute | polycom IP 320 |
| 23:59.55 | dijungal | grandstrea, gxp2000 |
| 23:59.55 | Toerkeium | Mw3: do you know if it supports an analog phone line also? |