| 00:04.00 | Teln1100A | Whats the difference between PSTN line and Line1 on Linksys SPA3102 and how do I control the logic of when I dial out on Phone port to go out through Voip |
| 00:17.45 | EclecticRob | where are asterisk sound files saved usually? |
| 00:20.43 | lesouvage | Is there a problem with the Asterisk forum. I just find out that my posts over the last years show up under another name. |
| 00:23.08 | lesouvage | I think there is a problem: My posts are from before the date this member has joined the asterisk forum. It' not really a problem but kind of strange to discover. |
| 00:29.47 | *** join/#asterisk apardo (n=apardo@87.223.171.17) |
| 00:36.25 | *** join/#asterisk hellop (n=hellop@cpe-66-91-197-100.hawaii.res.rr.com) |
| 00:42.53 | lesouvage | eclecticrob: /var/spool/asterisk/sounds or on a debian system /usr/share/asterisk/sounds |
| 00:43.07 | EclecticRob | I found it, thanks though :) |
| 00:54.05 | *** join/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 00:57.44 | dug | everytime I restart asterisk (1.4) I get a message telling me I have setup asterisk successfully on the first call, is there a way to disable this? |
| 01:00.23 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 01:04.01 | russellb | dug: you probably want to erase everything from extensions.conf and start over |
| 01:17.40 | *** join/#asterisk Corydon76-dig (i=ten@pdpc/supporter/sustaining/Corydon76-home) |
| 01:17.40 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 01:28.34 | dug | russellb: so I deleted extensions.ael and extensions.conf is basically blank, if I set all routes to the IVR it works, now how do I define it so that if I define the specific zap/3-1 to go to the IVR, when I do that it says no such context, what would be a simple context for an incoming line? |
| 01:30.13 | *** join/#asterisk apardo (n=apardo@80.174.32.86.dyn.user.ono.com) |
| 01:33.42 | jdg | dug: seems that context defined in zapata.conf for channel 3 does not exist in extensions.conf |
| 01:36.21 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
| 01:37.42 | *** join/#asterisk kaihanari (n=kaihanar@CPE001478ec4f47-CM0011e6c7e1cf.cpe.net.cable.rogers.com) |
| 01:38.37 | *** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu) |
| 01:38.46 | kaihanari | anyone know any good to-pstn providers other than teliax that offer cheap toll-free numbers like teliax? teliax would be ideal but they have an issue with my credit card, namely, it doesnt like what i put in the zip feild so im looking for someone similar |
| 01:42.52 | *** join/#asterisk Greenbox (n=12243@c-68-59-20-153.hsd1.sc.comcast.net) |
| 02:08.09 | *** join/#asterisk etfonhomey_ (n=chatzill@mobile-166-214-179-099.mycingular.net) |
| 02:09.49 | dug | for some reason on every other call my system picks up like a fax, then I call back and my system answers with the IVR |
| 02:12.52 | *** join/#asterisk asdx (n=asdx@adsl-158-87.click.com.py) |
| 02:23.14 | *** join/#asterisk tuxd00d (n=tuxinato@c-76-27-72-211.hsd1.ut.comcast.net) |
| 02:26.23 | *** join/#asterisk coppice (n=chatzill@140.196.17.210.dyn.pacific.net.hk) |
| 02:27.47 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 02:28.45 | *** join/#asterisk ManxPower (n=manxpowe@52.sub-75-200-255.myvzw.com) |
| 02:29.42 | *** join/#asterisk slakware (n=attila@201.19.133.161) |
| 02:30.04 | *** join/#asterisk Yourname` (n=IM@unaffiliated/yourname/x-837320) |
| 02:30.09 | slakware | How would one go about applying patches found on the bugs.digium.com site? |
| 02:34.12 | jdg | download your_patch, change to asterisk sources directory, then: patch -p1 < your_patch |
| 02:35.06 | *** join/#asterisk apardo (n=apardo@80.174.32.86.dyn.user.ono.com) |
| 02:35.30 | slakware | thanks |
| 02:36.41 | EclecticRob | Do you have to do anything to make Asterisk use SIP? I followed the same process for adding a SIP user as I did for an IAX user but it doesn't seem to work for the SIP user... the server doesn't seem to respond to the connection requests or anything... no error log no nothing |
| 02:44.50 | *** join/#asterisk apardo (n=apardo@80.174.32.86.dyn.user.ono.com) |
| 02:45.10 | *** part/#asterisk apardo (n=apardo@80.174.32.86.dyn.user.ono.com) |
| 02:50.56 | jdg | Is chan_sip loaded ? |
| 02:54.18 | ManxPower | load chan_sip.so |
| 02:55.26 | kaihanari | is there any way to amplify a channel's outgoing sound (but not incoming) in a call? my voip provider seems to drop the volume. |
| 02:55.29 | CCFL_Man2 | this cisco works like a mini pbx actually |
| 02:55.35 | kaihanari | but only on outgoing sound |
| 02:56.01 | CCFL_Man2 | kaihanari: speak louder :P |
| 02:56.19 | EclecticRob | hmm... chan_sip is already loaded... I managed to get debugging turned on and I can see that it is not authenticating properly but I haven't been able to figure out why yet |
| 02:56.58 | CCFL_Man2 | EclecticRob: check the usernames and passwords |
| 02:59.21 | CCFL_Man2 | We do not support pulse dialing on any of our cards. |
| 02:59.35 | CCFL_Man2 | then fu carrier access |
| 03:01.44 | CCFL_Man2 | russellb: you there? |
| 03:02.20 | CCFL_Man2 | russellb: how do i calibrate a WE302 dial to pulse at the right speed? |
| 03:02.52 | EclecticRob | CCFL, all seems to be well |
| 03:03.05 | EclecticRob | is there any way to enable debug of authentication for SIP? |
| 03:03.29 | kaihanari | CCFL_Man2, nah. its fine ext-ext. just outgoing sound only |
| 03:04.17 | CCFL_Man2 | EclecticRob: doesn't it give a reason for not authorizing? |
| 03:04.18 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 03:04.24 | EclecticRob | not that I can tell |
| 03:04.54 | CCFL_Man2 | kaihanari: i honestly don't know |
| 03:05.15 | CCFL_Man2 | EclecticRob: make sure the console is fully verbose |
| 03:05.23 | CCFL_Man2 | i thing -vvvv |
| 03:05.35 | etfonhomey_ | core set verbose 10 |
| 03:08.32 | EclecticRob | it is fully verbose |
| 03:08.53 | EclecticRob | I enabled logging and it was showing the packets and headers but I couldn't see any messages that really told me why it wasn't working |
| 03:14.42 | *** join/#asterisk dlynes (n=dlynes@d154-20-9-152.bchsia.telus.net) |
| 03:21.31 | CCFL_Man2 | if i define a dial peer, is anything else required for incomming sip? |
| 03:22.27 | *** join/#asterisk dug (n=chatzill@adsl-71-131-39-119.dsl.sntc01.pacbell.net) |
| 03:25.51 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:29.08 | sparq | Hmm... If I have outbound calling through a SIP peer working, how would people suggest I diagnose non-working inbound calls? |
| 03:29.38 | sparq | I'm pretty sure I've got the right ports forwarded to my Asterisk box |
| 03:30.02 | sparq | (5060/udp and 10000-20000/udp) |
| 03:31.16 | *** join/#asterisk apardo (n=apardo@87.223.171.17) |
| 03:31.33 | *** part/#asterisk apardo (n=apardo@87.223.171.17) |
| 03:33.52 | CCFL_Man2 | sparq: diagnose it on the other end? |
| 03:34.25 | Sweeper | sparq: tcpdump at the edge of the network to see if the requests are actually coming in |
| 03:35.04 | CCFL_Man2 | Sweeper: i fixed the telnet problem, telnet padding needed to be on |
| 03:35.04 | Nugget | telnet is eeeeeeevil! |
| 03:35.30 | CCFL_Man2 | telnet |
| 03:35.30 | Sweeper | CCFL_Man2: coolz |
| 03:35.58 | CCFL_Man2 | Sweeper: i'm a bit dissapointed it doesn't support ssh |
| 03:35.59 | Sweeper | I <3 console servers, too bad I don't have anything that needs them any more :( |
| 03:36.03 | sparq | Aaargh. I think I need a filter. |
| 03:36.47 | sparq | My DSL line is under a permenant ping flood. |
| 03:37.16 | CCFL_Man2 | Sweeper: i have a server in my closet, i use old phone wiring through out the house to get a serial console in one room, i can use the two terminal servers |
| 03:38.04 | Sweeper | sparq: call your isp and report abuse |
| 03:38.07 | CCFL_Man2 | Sweeper: and something i also never knew, serial tunneling is done with a telnet server on one lantronix box and a client on the other |
| 03:38.36 | Sweeper | sparq: also, read the man page, and set the host ip address to the provider's ip |
| 03:38.51 | Sweeper | CCFL_Man2: I despise serial tunneling :v |
| 03:38.58 | EclecticRob | well, it appears I had a problem with NAT :P |
| 03:39.33 | CCFL_Man2 | Sweeper: how come? with software or no flow control it's ok |
| 03:39.43 | asdx | why is telnet evil, not encrypted, vulnerabilities? |
| 03:39.56 | asdx | (just curious) |
| 03:40.05 | CCFL_Man2 | asdx: it's old and still used |
| 03:40.55 | asdx | heh |
| 03:41.20 | CCFL_Man2 | it's a serial line over tcp/ip |
| 03:41.24 | Sweeper | asdx: lack en encryption makes it a bad thing to be used on unsecure networks |
| 03:41.24 | asdx | ssh > telnet :p |
| 03:41.27 | Sweeper | *of |
| 03:41.28 | CCFL_Man2 | basics of basics |
| 03:41.31 | Sweeper | ssh IS nice |
| 03:41.47 | Sweeper | but the fact is, telco hardware likes to stick with what it knows works |
| 03:41.50 | asdx | Sweeper: yeh |
| 03:41.53 | Sweeper | also, telnet has a much lower overhead |
| 03:41.53 | CCFL_Man2 | my terminal servers don't support ssh |
| 03:43.38 | CCFL_Man2 | so i can't use em to access my shell over a wan |
| 03:45.19 | Sweeper | sure you can ;) |
| 03:45.30 | sparq | Sweeper: Meh. I've tried reporting it, but the flood just comes from somewhere else. I think they are just infected Windows boxes. |
| 03:45.32 | Sweeper | you just need proper ssh foo |
| 03:45.34 | CCFL_Man2 | but, i was able to use an old old cisco mc3810 with sip and interface this adit 600 channel bank to sip |
| 03:45.36 | *** join/#asterisk MdeP (n=mdep@99-93-22-190.adsl.tie.cl) |
| 03:46.16 | sparq | oooo. "tcpdump -i ath0 not port 80" is very handy. |
| 03:46.16 | Sweeper | CCFL_Man2: you mean, adit -> cisco --SIP--> something? |
| 03:46.37 | Sweeper | sparq: wifi for your main internets? :v |
| 03:47.02 | CCFL_Man2 | Sweeper: i can only do it if i can telnet into a box from the lantronix and ssh from the box i telneted into |
| 03:47.09 | CCFL_Man2 | Sweeper: exactly |
| 03:47.29 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 03:48.04 | sparq | Sweeper: Sad, but true. |
| 03:48.44 | sparq | Sweeper: the conduit crossing the back yard was wrecked by a tortoise. |
| 03:48.55 | Sweeper | sparq: amazing |
| 03:48.58 | sparq | Sweeper: Until I fix it, WiFi it is. |
| 03:49.19 | CCFL_Man2 | Sweeper: you use an adit 600 before? |
| 03:49.24 | Sweeper | CCFL_Man2: oh yea |
| 03:49.31 | Sweeper | I like thems :) |
| 03:49.37 | CCFL_Man2 | me too |
| 03:49.54 | CCFL_Man2 | i got an oild one though, has the original 4g fxs cards |
| 03:49.59 | CCFL_Man2 | old |
| 03:50.26 | CCFL_Man2 | i got the latest firmware from someone in here |
| 03:50.53 | CCFL_Man2 | and it supports pulse dialing |
| 03:50.56 | CCFL_Man2 | :P |
| 03:52.04 | CCFL_Man2 | it honestly works great |
| 03:52.39 | CCFL_Man2 | but i think the newer fxs cards would work better, i'm not sure |
| 03:52.48 | *** join/#asterisk bmg505 (n=leon@196.209.178.253) |
| 03:53.01 | CCFL_Man2 | i think i might need to adjust input gain |
| 03:54.47 | *** join/#asterisk tuxd00d (n=tuxinato@c-76-27-72-211.hsd1.ut.comcast.net) |
| 03:55.43 | sparq | Sweeper: Hmm... when I place an inbound call, I see a REGISTER packet, then a 200 OK packet, then an INVITE packet, then a 404 Not Found packet, and then an ACK packet. |
| 03:56.06 | sparq | I guess Asterisk is getting the call, but doesn't know what to do with it? |
| 03:57.10 | Sweeper | sounds about right |
| 03:57.23 | Sweeper | btw, you can do a "set dip debug" |
| 03:57.38 | Sweeper | in asterisk |
| 03:57.40 | Sweeper | and see those |
| 03:57.53 | sparq | you mean sip debug? |
| 03:58.04 | Sweeper | maybe |
| 03:58.11 | Sweeper | I forget which is deprected |
| 03:59.28 | sparq | eek. That was a lot of output. |
| 04:00.53 | sparq | Allright, so it's definitely getting the INVITE from my SIP peer |
| 04:04.06 | EclecticRob | man, this is one tricky piece of software :P |
| 04:04.31 | sparq | Sweeper: Is there a simple way of routing inbound calls to a voice menu? |
| 04:05.31 | *** join/#asterisk shido6 (n=shido6@74-130-227-15.dhcp.insightbb.com) |
| 04:06.41 | Sweeper | sparq: just use exten=>_X.,1,Playback('hello') |
| 04:07.02 | Sweeper | in whatever context your peer or register line is pointed at |
| 04:07.17 | *** join/#asterisk slakware (n=attila@201.19.173.140) |
| 04:07.19 | slakware | I am using odbc driver with a mssql database. all is working well. however i am unable to do pattern matching within the realtime extensions database. the sp_execute stored procedure being executed is as follows: exec sp_execute 534,'\_%','outgoing','1'. I saw the bug note on bugs page, however i'm running 1.4.11, which has the fix incorporated... |
| 04:07.56 | slakware | i'm running Asterisk SVN-branch-1.4-r81832 |
| 04:09.11 | Sweeper | sparq: when you're ready to test, you can set sip debug off, and then you'll be able to see the asterisk output |
| 04:09.20 | Sweeper | also make sure you're doing asterisk -rvvvv |
| 04:09.28 | Sweeper | or set verbose 4 |
| 04:11.40 | sparq | Sweeper: Hmm... It still just drops right into my BroadVoice mailbox. |
| 04:12.10 | Sweeper | sparq: turn sip debug off, see if you're still sending the 404 |
| 04:15.07 | sparq | Sweeper: Yep, still sending the 404. |
| 04:15.40 | Sweeper | sparq: pastebin sip.conf and extensions.conf |
| 04:17.15 | *** join/#asterisk WizardWlf (n=shawn@wsip-70-167-225-171.om.om.cox.net) |
| 04:23.02 | sparq | Sweeper: http://vort.org/media/data/sip.conf http://vort.org/media/data/extensions.conf |
| 04:25.48 | Sweeper | sparq: in sip.conf, under sip.broadvoice.com, set context=default |
| 04:25.58 | Sweeper | and do a extensions reload after that, in the * cli |
| 04:26.10 | sparq | Hmm... |
| 04:29.25 | sparq | Sweeper: It still appears to be doing the same thing. |
| 04:29.59 | Sweeper | ok |
| 04:30.25 | Sweeper | now, PASTEBIN the output from the cli, with ssip debug turned on, as well as your new sip.conf |
| 04:30.41 | sparq | heh |
| 04:42.30 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:43.12 | *** join/#asterisk r0d3nt (i=nobody@punk.valuetel.net) |
| 04:43.40 | sparq | Sweeper: http://vort.org/media/data/sip.debug |
| 04:43.47 | sparq | Sweeper: you can reload sip.conf |
| 04:48.14 | sparq | Sweeper: Any clues? |
| 04:53.54 | sparq | I really need to find a good book on this stuff. |
| 04:59.06 | sparq | Sweeper: The 404's don't seem to show up in the debugging logs. I just see them in tcpdump./ |
| 05:06.04 | *** join/#asterisk Chicago (n=Chicago@c-24-12-127-34.hsd1.in.comcast.net) |
| 05:13.55 | *** join/#asterisk apardo (n=apardo@243.144.217.87.dynamic.jazztel.es) |
| 05:16.14 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 05:21.08 | *** join/#asterisk tsurko (n=tsurko@213.91.216.130) |
| 05:23.55 | *** join/#asterisk sergey (n=sergey@gw4-130.iks.ru) |
| 05:32.54 | sparq | Holy hell, the jitter is aweful. |
| 05:35.36 | Nugget | Try decaffeinated. |
| 05:36.32 | *** join/#asterisk BillBinko (n=BillBink@rrcs-24-73-75-102.se.biz.rr.com) |
| 05:37.33 | BillBinko | Hello everyone. |
| 05:38.03 | CCFL_Man2 | Nugget: LOL |
| 05:39.34 | BillBinko | I'm having a problem with my first cut at a PHPAGI app. get_data() works the first time, but afterwords, no audio is sent down the SIP stream . (Asterisk 1.2.22, TrixBox) |
| 05:39.51 | BillBinko | Any pointers? |
| 05:41.06 | *** join/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 05:41.09 | CCFL_Man2 | ok, i connected the serial line to my two terminal servers and serial tunneling seem to be working good |
| 05:55.19 | BillBinko | I'm going back to beating on PHPAGI. If anyone's got any ideas or experience in * stopping to send audio after a successful get_data() call, please ping me |
| 06:04.21 | *** join/#asterisk grndslm (n=grndslm@24-116-87-97.cpe.cableone.net) |
| 06:07.14 | EclecticRob | dang... once you get it (mostly) working, Asterisk is pretty awesome |
| 06:10.55 | BillBinko | Until you hit a snag and you're up till zero-dark-thirty ;-) |
| 06:11.21 | EclecticRob | I can imagine... it is really complex software |
| 06:11.36 | EclecticRob | I still have a few issues but I have some stuff working... which is better than this morning |
| 06:17.41 | CCFL_Man2 | Sweeper: can serial tunneling be done with different baudrates on both ends? |
| 06:26.40 | *** part/#asterisk jmls (n=jmls@62.49.235.130) |
| 06:29.33 | EclecticRob | Does Asterisk come with a module or system to do call forwarding where it prompts the caller for their name and presents it to the end point where they can hear the name and then accept or reject the call? |
| 06:32.39 | ectospasm | EclecticRob: I think it can, but I can't think of how off the top of my head |
| 06:37.57 | *** join/#asterisk catch23 (n=catch23@69.60.124.109) |
| 06:39.26 | catch23 | anyone know of a hardware gateway like the sipura 3000 that supports iax2 & has fxo/fxs ports? the closest thing i could find was the digium iaxy, but it doesn't have a fxo port |
| 06:40.27 | *** join/#asterisk BillBinko (n=BillBink@rrcs-24-73-75-102.se.biz.rr.com) |
| 06:40.45 | BillBinko | Grumble - soft phone went nuts and killed my box |
| 06:41.13 | BillBinko | Not to be a pest, but does anyone have an idea as to why I can only run GET DATA once before I look all audio ? |
| 06:49.26 | *** join/#asterisk r0d3nt (i=nobody@punk.valuetel.net) |
| 06:55.52 | BillBinko | Weird... it works fine on the 3CX softphone, but not at all on the ExpressTalk |
| 06:56.09 | BillBinko | Not critical as long as it works with a real trunk (testing tomorrow) |
| 06:59.59 | sparq | Does anyone know why Asterisk would return a 404 message as a response to an INVITE message from a peer? |
| 07:01.14 | sparq | Is it missing a default extension? Not attached to the right context? Out of weasels? |
| 07:01.32 | *** join/#asterisk af_ (n=getsmart@81-174-47-128.dynamic.ngi.it) |
| 07:02.08 | BillBinko | no idea (sorry) |
| 07:03.37 | Corydon76-dig | 404 generally means the extension does not exist |
| 07:06.02 | sparq | Corydon76-dig: Hmm. I suppose I need to add a default extension to the context then, yes? |
| 07:06.37 | Corydon76-dig | Uh, why would you want a default extension? |
| 07:11.21 | sparq | Corydon76-dig: unless I can automatically select an extension... |
| 07:11.54 | sparq | Whatever it takes to get Asterisk to pick up the phone and do the echo test |
| 07:13.07 | Corydon76-dig | Pick one |
| 07:14.25 | sparq | Corydon76-dig: Well, yes. I just can't figure out how. |
| 07:15.01 | Corydon76-dig | exten => 1234,1,Echo() |
| 07:16.21 | sparq | Corydon76-dig: Hmm... I tried exten => 1,1,Answer() |
| 07:16.29 | sparq | is there something special about 1234? |
| 07:16.56 | ectospasm | Well, you gotta pick up the phone and dial 1234... |
| 07:17.26 | ectospasm | if you want it to drop into the echo test when you pick up, I dunno... you may be able to do it with the s exten |
| 07:17.56 | *** join/#asterisk kissand (n=kissand@ppp115-124.dsl.hol.gr) |
| 07:18.10 | kissand | hello people |
| 07:18.34 | sparq | ectospasm: It never gets that far. Asterisk kicks out a 404, and my peer goes right into its voicemail. |
| 07:18.53 | kissand | i have a BN4S0 and spa-921 sip phone. dtmf does not work while grandstream sip phone dtmf works. any ideas? |
| 07:18.57 | ectospasm | sparq: sounds like a registration problem |
| 07:19.14 | sparq | ectospasm: could be, though outbound calls work fine. |
| 07:19.35 | ectospasm | sparq: what happens on the CLI when you try to dial in? |
| 07:19.48 | CCFL_Man2 | you can serial tunnel with one end at a different speed than the other |
| 07:20.01 | CCFL_Man2 | finally accessing the shell at 115200 |
| 07:20.30 | CCFL_Man2 | some stuff seems to mess up though |
| 07:21.20 | sparq | ectospasm: I see the INVITE, and then the connection closes. I had to find the 404 message with tcpdump. |
| 07:22.00 | ectospasm | sip debug should show you that |
| 07:22.04 | ectospasm | or wait |
| 07:22.53 | ectospasm | sparq: in 1.4 it'll be "sip set debug" |
| 07:23.03 | sparq | Yep. That is ineed what I see with debug. |
| 07:23.25 | sparq | ectospasm: http://vort.org/media/data/sip.debug |
| 07:24.04 | sparq | sorry, my web server thinks it's a binary file. |
| 07:26.23 | ectospasm | sparq: I got it... what phone are you using? |
| 07:26.29 | sparq | Too bad the SIP protocol's version of 404 doesn't actually say what it was that wasn't found. |
| 07:26.59 | ectospasm | sparq: usually it means the user isn't registered |
| 07:27.10 | ectospasm | sparq: what does "sip show peers" show? |
| 07:27.13 | sparq | ectospasm: I'm calling from my cell phone to my BroadVoice number. |
| 07:27.21 | ectospasm | I meant the SIP phone |
| 07:27.45 | ectospasm | I mean, is it a softphone, an ATA, or a hard phone? |
| 07:28.10 | ectospasm | sparq: You may want to set qualify=yes in the sip user/friend section of sip.conf |
| 07:28.12 | sparq | it's a softphone, but right now I'm just trying to get the echotest to work. |
| 07:28.36 | sparq | I figure I should take it one problem at a time ^_^ |
| 07:29.23 | ectospasm | can you do the echo test from the softphone? |
| 07:29.24 | sparq | If I can get Asterisk to do *anything* with inbound calls, then I can get my terminals working |
| 07:29.31 | sparq | yep |
| 07:30.01 | kaldemar | sparq: the number you dialed was not found. |
| 07:30.20 | kaldemar | look at this line: "Looking for xxxxxxxxxx in from-broadvoice " |
| 07:30.35 | sparq | oh, yes. |
| 07:30.52 | sparq | I fixed that. Let me upload a new debug file. |
| 07:31.45 | kaldemar | don't remove the other cli output. |
| 07:33.35 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 07:35.51 | sparq | ectospasm: http://vort.org/media/data/sip.debug.txt |
| 07:36.07 | sparq | kaldemar: ^^ |
| 07:37.02 | ectospasm | sparq: are you trying to dial the softphone? |
| 07:37.18 | sparq | nope |
| 07:37.31 | sparq | I'm calling from a cell phone through my VoIP carrier |
| 07:37.58 | sparq | calling from the softphone to the echo test or the outside world works fine |
| 07:38.17 | ectospasm | sparq: does sip.conf have a peer section for your SIP provider? |
| 07:38.25 | sparq | yep |
| 07:38.42 | sparq | http://vort.org/media/data/sip.conf |
| 07:40.46 | kaldemar | looks to me like the number you're trying to reach is not found in from-broadvoice. |
| 07:41.21 | ectospasm | that'll be in extensions.conf |
| 07:41.31 | kaldemar | based on the information you're giving, you seem to have a dialplan problem rather than a SIP problem. |
| 07:41.50 | kissand | i have a BN4S0 and spa-921 sip phone. dtmf does not work while grandstream sip phone dtmf works. any ideas? |
| 07:42.26 | sparq | kaldemar: which file do you see from-broadvoice in? |
| 07:42.31 | kaldemar | kissand: have you tried different dtmfmodes? |
| 07:42.36 | kaldemar | sparq: extensions.conf |
| 07:42.41 | kissand | all off them :> |
| 07:43.03 | kissand | the same setup with zapata works. with beronet not |
| 07:43.03 | ectospasm | kissand: dtmfmode has to match in the phones, too |
| 07:43.09 | sparq | Drat it. I deleted that context. |
| 07:43.13 | kissand | ector i know, done that |
| 07:43.27 | sparq | everything should be going to [home] |
| 07:43.29 | kissand | the same setup with grandteam works |
| 07:43.42 | kissand | grandstream sip phone works, with linksys no. |
| 07:49.34 | pkunkra | You know, I wonder if there are artists that actually specialize in making music-on-hold tunes. |
| 07:50.15 | pkunkra | or do they just filter out the vocals from a particularly boring piece of music and use that instead? |
| 07:50.21 | kissand | parov stellar MOH tune :> |
| 07:51.52 | pkunkra | hmmm |
| 07:51.55 | pkunkra | http://www.amazon.com/Rough-Cuts-Parov-Stelar/dp/B0005FAINY |
| 07:51.57 | pkunkra | looks boring. |
| 07:54.16 | EclecticRob | does anyone know how to make new ParkAndAnnounce() templates? |
| 07:54.26 | sparq | kaldemar: So, from the softphone, I can dial the extension for the echotest, and it works fine. How the heck am I supposed to dial an extension from outside? |
| 07:55.32 | BillBinko | I am very new to this, but I think the trick is to get it working with the softphone using "simulate external call" (7777 on Trixbox). Then when you call in, you should be ok |
| 07:57.17 | kaldemar | sparq: your outside calls are now landing in context from-broadvoice, you have defined that in sip.conf. you need to have the echo test number reachable from from-broadvoice. |
| 07:57.52 | kaldemar | sparq: if the numer is in home context, you can for example include it by putting "include => home" in from-broadvoice. |
| 08:02.10 | sparq | kaldemar: like so? (you can refresh the files) |
| 08:05.47 | *** join/#asterisk Arno[Slack] (i=100@gre92-1-81-57-177-108.fbx.proxad.net) |
| 08:05.48 | sparq | kaldemar: I apologize for getting ahead of the config files on my web server. It's a pain to expunge the passwords. |
| 08:09.56 | kaldemar | sparq: that should do it, but only if the length of the number is 1. |
| 08:10.19 | kaldemar | if something longer is coming to asterisk, you'll get a 404 again. |
| 08:10.25 | sparq | kaldemar: Hmm. |
| 08:10.31 | kaldemar | except for 101 of course. |
| 08:11.47 | sparq | Man, I haven't felt this dumb since quantum field theory. |
| 08:12.05 | sparq | ectospasm: Thanks for your help |
| 08:13.53 | *** join/#asterisk Arno[Slack] (i=100@gre92-1-81-57-177-108.fbx.proxad.net) |
| 08:14.47 | sparq | kaldemar: In the SIP message header, the To: value is "Russell Neches"<sip:1@my.ip.addr>" |
| 08:15.05 | sparq | kaldemar: that woud seem to suggest it's looking for extension 1, yes? |
| 08:17.17 | EclecticRob | for some reason, I get disconnected almost everytime I call a cell phone just before I would be transfered to voicemail for the cell phone... anyone else experience this? |
| 08:18.12 | sparq | kaldemar: Aaaaand.... it still throws a 404. |
| 08:20.14 | kaldemar | sparq: look at the INVITE line in stead. what number does it have? |
| 08:21.18 | sparq | kaldemar: Ah ha! |
| 08:21.30 | sparq | It had my freaking phone number in it. |
| 08:21.43 | sparq | That's useless. |
| 08:21.55 | *** join/#asterisk apardo (n=apardo@243.144.217.87.dynamic.jazztel.es) |
| 08:24.41 | sparq | SWEET! |
| 08:25.00 | sparq | kaldemar: Thank you. |
| 08:25.26 | kaldemar | you're welcome. |
| 08:26.50 | sparq | Now I get to learn how to make it do something *usefull*. |
| 08:54.04 | *** part/#asterisk BillBinko (n=BillBink@rrcs-24-73-75-102.se.biz.rr.com) |
| 09:01.44 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.129.156) |
| 09:38.45 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.129.156) |
| 09:42.13 | *** join/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 09:43.41 | hi365 | after installing asterisk from source if i type 'service asterisk status' i get 'asterisk: unrecognized service' ? |
| 09:45.32 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 09:53.10 | hi365 | after installing asterisk from source if i type 'service asterisk status' i get 'asterisk: unrecognized service' ? |
| 09:55.36 | mvanbaak | hi365: you will have to install the init script |
| 09:55.50 | hi365 | mvanbaak: install = copy? |
| 09:56.09 | mvanbaak | it's in the sourcedir under contrib/init.d |
| 09:56.12 | mvanbaak | yes |
| 09:56.18 | mvanbaak | copy it and make it executable |
| 09:56.53 | hi365 | cool. thanks |
| 09:57.51 | *** part/#asterisk dseeb_ (n=dcb@CPE-124-179-242-169.vic.bigpond.net.au) |
| 10:05.06 | hi365 | what do i need to do to get the uniqid field in mysql?? ive added the line "#define MYSQL_LOGUNIQUEID" to cdr_addon_mysql.c, but i still dont see the uniqeID |
| 10:08.03 | *** join/#asterisk Arno[Slack] (i=100@gre92-1-81-57-177-108.fbx.proxad.net) |
| 10:29.38 | *** join/#asterisk henkoegema (n=henkoege@d54C552E4.access.telenet.be) |
| 10:29.57 | henkoegema | <PROTECTED> |
| 10:34.12 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
| 10:34.37 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
| 10:42.18 | *** join/#asterisk shtoom (n=shtoom@59.93.115.109) |
| 10:43.17 | shtoom | Asterisk ended with exit status 127 |
| 10:43.17 | shtoom | Asterisk died with code 127. |
| 10:43.17 | shtoom | cat: __ASTERISK_VARRUN_DIR__/asterisk.pid: No such file or directory |
| 10:43.17 | shtoom | Hi I've just installed aterisk 1.4 when I tried to start it thru init script its producing the above error , any help? |
| 10:45.04 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 11:14.10 | *** join/#asterisk keulin (n=cray@ifth-pdcr2.infotheme.fr) |
| 11:15.03 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 11:16.28 | *** join/#asterisk keulin (n=cray@ifth-pdcr2.infotheme.fr) |
| 11:40.33 | *** join/#asterisk m0t3jl (n=m0t3jl@ip103.galance.net) |
| 11:41.23 | m0t3jl | Hi, its not really an asterisk question, but maybe it is :-): How can I forward calls on XLite SoftVoIP phone? (It doesnt look like it has a button for it) |
| 11:44.55 | *** join/#asterisk yannj_fr (n=yannj_fr@ALagny-152-1-15-203.w83-112.abo.wanadoo.fr) |
| 11:57.10 | *** join/#asterisk LakeSolon (n=blake@12-202-202-168.client.mchsi.com) |
| 12:15.11 | *** join/#asterisk puzzled (n=patrick@53536BB3.cable.casema.nl) |
| 12:15.40 | puzzled | hi |
| 12:16.37 | *** join/#asterisk riddlebox (n=victoria@75-132-225-75.dhcp.stls.mo.charter.com) |
| 12:16.53 | *** join/#asterisk Woifi1988 (n=anon@M1591P025.adsl.highway.telekom.at) |
| 12:16.55 | Woifi1988 | hi |
| 12:17.08 | *** join/#asterisk Qapf (n=Qapf@mail.oldworlddoor.com) |
| 12:19.38 | Woifi1988 | how does voicemail notificaton work with asterisk? do i need a postfix server? |
| 12:21.13 | m0t3jl | Woifi1988, if you have VM capable phone then the phone itself shows the user that they have a VM |
| 12:23.44 | Woifi1988 | m0t3jl: is x-lie a vm capable phone? |
| 12:23.51 | *** join/#asterisk WizardWlf (n=shawn@wsip-70-167-225-171.om.om.cox.net) |
| 12:24.06 | *** join/#asterisk davixx (n=davixx@82.248.85.33) |
| 12:24.07 | m0t3jl | Woifi1988, yes ... at least I think it is |
| 12:24.21 | m0t3jl | Woifi1988, when you have a message it appears as a message icon |
| 12:24.58 | Woifi1988 | and when i want to use e-mail notofication i need a postif server? |
| 12:26.06 | Woifi1988 | postfix |
| 12:26.53 | *** join/#asterisk davixx (n=davixx@82.248.85.33) |
| 12:27.07 | m0t3jl | Woifi1988, I believe that you have to be able to send emails, but I dont think that you really need to have postfix, I think that if you tell asterisk to use a certain smtp server then it should be also possible |
| 12:27.58 | *** join/#asterisk keulin (n=cray@ifth-pdcr2.infotheme.fr) |
| 12:28.15 | Woifi1988 | m0t3jl: but i can't find any option for that! there is only the option to enter a mail adress but i can't find an option to enter a smtp server! |
| 12:29.07 | m0t3jl | Woifi1988, thats strange ... So I suppose that it must be using its own server or something... I really dont know |
| 12:29.29 | Woifi1988 | m0t3jl: okay thanks |
| 12:29.46 | m0t3jl | But I would like to know :-) |
| 12:30.07 | m0t3jl | Since when the Asterisk is to receive faxes you can also put an email adress to do so ... |
| 12:32.42 | riddlebox | m0t3jl, faxing will not work if you have an ATA device converting sip to analog will it? |
| 12:32.45 | *** join/#asterisk DEac- (n=deac@Platin.DenKn.de) |
| 12:32.48 | DEac- | moin |
| 12:33.40 | DEac- | i want to call a function in extension.conf, which does a sql-fetch, but without odbc. is this possible? |
| 12:33.44 | m0t3jl | riddlebox, dunno, I did not test faxing |
| 12:34.02 | DEac- | and without agi |
| 12:34.29 | riddlebox | ahh I just figured since I saw you talking about it, ohh well I have a smart switch in front of the line so it doesnt really matter |
| 12:36.14 | m0t3jl | riddlebox, well, we have one PST line dedicated to fax machine, so we dont care about faxes |
| 12:40.07 | riddlebox | has anyone seen Druid? it seems like it is using asterisk with their own gui |
| 12:42.37 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 12:43.19 | *** join/#asterisk potsboy (n=chrisg@vc-196-207-32-228.3g.vodacom.co.za) |
| 12:45.04 | riddlebox | I keep sending links of different companies using Asterisk, and also any vendor that we deal with that has an Asterisk appliance to my boss hopefully we will soon start to sell asterisk solutions |
| 12:56.17 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 13:05.42 | *** join/#asterisk af_ (n=getsmart@81-174-47-128.dynamic.ngi.it) |
| 13:07.02 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 13:07.49 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:09.28 | *** part/#asterisk potsboy (n=chrisg@vc-196-207-32-228.3g.vodacom.co.za) |
| 13:25.08 | *** join/#asterisk zapp-branigan (n=zapp-bra@9.218.216.87.static.jazztel.es) |
| 13:52.28 | riddlebox | do you guys recomend using the Asterisk Gui provided from Digium, or just edit the conf files by hand? |
| 13:52.47 | mvanbaak | I do edit the confs by hand |
| 13:53.42 | riddlebox | I edit them by hand now, but I think my extensions.conf is pretty ugly hrmm I wonder if there is an article somewhere that will show you good ways of doing it |
| 13:56.29 | *** join/#asterisk matt_ (n=matt@2001:770:168:1:220:edff:feb4:7c9d) |
| 14:28.16 | *** join/#asterisk ToTo (n=ToTo@87.9.208.53) |
| 14:31.48 | *** join/#asterisk MukulJain (n=jainmuku@cm69.omega97.maxonline.com.sg) |
| 14:33.42 | MukulJain | Hi |
| 14:33.50 | MukulJain | I am having problem with Asterisk BLF |
| 14:34.03 | MukulJain | I am using GXP-2000 Phones, |
| 14:34.06 | CCFL_Man2 | russellb: you there? |
| 14:34.37 | shtoom | Asterisk ended with exit status 127 |
| 14:34.37 | shtoom | Asterisk died with code 127. |
| 14:34.37 | shtoom | cat: __ASTERISK_VARRUN_DIR__/asterisk.pid: No such file or directory |
| 14:34.37 | shtoom | Hi I've just installed aterisk 1.4 when I tried to start it thru init script its producing the above error , any help? |
| 14:34.37 | shtoom | Â |
| 14:35.29 | MukulJain | shtoom : Do you have /var/run/asterisk directory in your system ? |
| 14:35.54 | shtoom | MukulJain: Ya I've created it manually |
| 14:36.01 | MukulJain | what permission ? |
| 14:36.19 | MukulJain | Ensure that it has permission to the Asterisk user and group for RWX |
| 14:36.27 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177582660.dsl.bell.ca) |
| 14:37.48 | shtoom | here is ls out put : drwxr-xr-x 2 root root 40 2007-09-09 16:05 asterisk |
| 14:38.27 | MukulJain | okay for me it's like |
| 14:38.41 | MukulJain | drwxrwx--- 2 asterisk asterisk 4096 Sep 7 11:05 asterisk |
| 14:38.54 | tzafrir | shtoom, the pid file is likely to be generated in the varrundir, if one is defined in asterisk.conf |
| 14:39.18 | shtoom | ya its getting generated there |
| 14:39.19 | MukulJain | Do you have asterisk user created in your system ? |
| 14:39.24 | MukulJain | aah Okay. |
| 14:39.46 | shtoom | if i start asterisk with asterisk -vvvvvvvvvvvc |
| 14:39.50 | shtoom | its running ok |
| 14:40.08 | tzafrir | so it is probably an error from safe_asterisk |
| 14:40.27 | shtoom | I am getting this error when i run it with /etc/init.d/asterisk start |
| 14:40.33 | tzafrir | I think it was recently adapted to have those directories modified |
| 14:40.59 | shtoom | tzafrir:I am running asterisk 1.4 |
| 14:41.14 | shtoom | on ubuntu fiesty |
| 14:41.29 | shtoom | first it gave me Bad fd error |
| 14:41.31 | MukulJain | 1.4.11 : I am having no erros, I had to copy the new init script after make install |
| 14:41.41 | shtoom | I fixed it after googling a but |
| 14:41.48 | shtoom | *bit |
| 14:42.34 | shtoom | but this one started to show up now |
| 14:42.47 | tzafrir | MukulJain, what linux distribution do you use? |
| 14:43.05 | MukulJain | CentOS 4.5 |
| 14:43.24 | shtoom | MukulJain: then theres is the difference i guess |
| 14:43.27 | tzafrir | shtoom, don't use safe_asterisk :-) |
| 14:43.34 | tzafrir | s/:-)/:-(/ |
| 14:44.01 | tzafrir | MukulJain, did you run 'make config'? |
| 14:44.17 | MukulJain | I run configure followed by make install |
| 14:44.25 | shtoom | actually Bad fd error is specific to unbuntu |
| 14:44.41 | MukulJain | and then I had to copy the file to overwrite my script at /etc/rc.d/init.d/asterisk |
| 14:45.04 | MukulJain | shtoom : In that case I really dont have clue, Didnt got chance to use Ubuntu so far :( Sorry Bro |
| 14:45.52 | shtoom | Mukuljain:Thats ok pal , Let me try make config |
| 14:47.27 | shtoom | Hmm that doesn't help |
| 14:48.45 | MukulJain | Anyone having GXV-3000 GS phones ? |
| 14:48.52 | MukulJain | I just got mine working on : Not bad |
| 14:52.36 | tzafrir | shtoom, get the init.d script from the asterisk deb |
| 14:53.04 | tzafrir | specifically, ubuntu likes to delete /var/run at startup, so the init.d script needs to recreate it |
| 14:53.23 | *** join/#asterisk rkioko (n=rkioko@41.206.48.74) |
| 14:53.31 | tzafrir | I think that this is already supported in the Debian package's init.d script as well. |
| 14:53.57 | tzafrir | And as a bonus you'll get an init.d script that does not use safe_asterisk by default |
| 14:54.34 | MukulJain | Out of curosity" What is the diffence in Asterisk and safe_asterisk ?? |
| 14:56.05 | *** join/#asterisk KDan (n=KDan@87-194-122-30.bethere.co.uk) |
| 14:56.24 | tzafrir | safe_asterisk is a wrapper script that is intended to restart asterisk if it ever gets killed |
| 14:56.49 | tzafrir | asterisk (/usr/sbin/asterisk) can run just fine as a daemon |
| 14:57.03 | tzafrir | safe_asterisk, however, runs it in a console |
| 14:57.11 | tzafrir | It gets quite a few other things wrong |
| 14:58.30 | MukulJain | Thanks tzafrir :) |
| 14:58.40 | MukulJain | How do u put red line ? on this chat ? |
| 14:58.41 | shtoom | tzafrir:I've copied contrib/init.d/rc.debian.asterisk to /etc/init.d/asterisk |
| 14:58.44 | MukulJain | Sorry new to IRC ?? |
| 14:59.14 | shtoom | but it seems like it is still calling /usr/sbin/safe_asterisk |
| 15:00.09 | MukulJain | hi |
| 15:01.02 | *** join/#asterisk Ebola (n=Ebola@host86-143-7-120.range86-143.btcentralplus.com) |
| 15:02.01 | MukulJain | Hi I am having problem with Asterisk BLF : Not working with GXP 2000 |
| 15:05.36 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 15:07.36 | *** join/#asterisk famicon (i=pastry@c51447ddc.cable.wanadoo.nl) |
| 15:09.28 | *** join/#asterisk MukulJain (n=jainmuku@cm69.omega97.maxonline.com.sg) |
| 15:15.15 | MukulJain | Hi |
| 15:15.27 | MukulJain | Anyone using BLF ?? using Grandstream 2000 ? |
| 15:17.53 | mvanbaak | yeah |
| 15:18.05 | MukulJain | Hi MvanBaak |
| 15:18.13 | MukulJain | I am having problem in using BLF |
| 15:18.18 | mvanbaak | what is it ? |
| 15:18.20 | MukulJain | Are u able to use them well |
| 15:18.29 | MukulJain | Okay, my scenario is |
| 15:18.34 | mvanbaak | yeah, they work great |
| 15:18.35 | MukulJain | I am having 3 extensions |
| 15:18.43 | MukulJain | all using gxp-3000 |
| 15:18.50 | mvanbaak | 2000 |
| 15:18.50 | MukulJain | and 2 Zaptel Lines |
| 15:18.54 | MukulJain | sorry 2000 yeah |
| 15:19.02 | MukulJain | I am able to monitor Zaptel, no problem |
| 15:19.13 | MukulJain | When they are busy light turn red, else they are green |
| 15:19.19 | MukulJain | so great no issue |
| 15:19.23 | mvanbaak | indeed |
| 15:19.34 | MukulJain | but for the other GS phones, whether they are busy or free. the lights are always green |
| 15:19.47 | mvanbaak | did you setup the hints correctly ? |
| 15:19.49 | shtoom | hooray ! I've crude coded the paths in there in safe_asterisk script it started working |
| 15:19.49 | shtoom | #ASTSBINDIR=__ASTERISK_SBIN_DIR__ |
| 15:19.50 | shtoom | #ASTPIDFILE=__ASTERISK_VARRUN_DIR__/asterisk.pid |
| 15:19.50 | shtoom | ASTSBINDIR=/usr/sbin/ |
| 15:19.50 | shtoom | ASTPIDFILE=/var/run/asterisk/asterisk.pid |
| 15:20.18 | MukulJain | hints are setup, but the status are always idle |
| 15:20.34 | MukulJain | when I type show hints at asterisk |
| 15:21.23 | mvanbaak | they show up ? |
| 15:21.44 | MukulJain | yes |
| 15:22.18 | mvanbaak | can you post some configs to a pastebin ? |
| 15:22.35 | mvanbaak | the sip.conf entry for a phone, the dialplan for the phones and hints |
| 15:22.44 | mvanbaak | ~pb |
| 15:22.45 | 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 |
| 15:24.32 | MukulJain | kams@ext-local-custom : Zap/4 State:Idle Watchers 0 |
| 15:24.32 | MukulJain | <PROTECTED> |
| 15:24.32 | MukulJain | <PROTECTED> |
| 15:24.32 | MukulJain | <PROTECTED> |
| 15:24.32 | MukulJain | <PROTECTED> |
| 15:24.33 | MukulJain | <PROTECTED> |
| 15:24.46 | MukulJain | Right now my phones are off |
| 15:24.53 | MukulJain | but when they are on the Status turns to IDLE |
| 15:25.12 | MukulJain | now when I call from 200 to 201, The status still show IDLE even if they are on call |
| 15:25.14 | mvanbaak | pastbin please |
| 15:25.17 | MukulJain | sorry |
| 15:25.28 | MukulJain | will do now |
| 15:26.06 | MukulJain | Jusdt do dthat |
| 15:26.32 | MukulJain | how to send the link for u to see ? |
| 15:27.11 | mvanbaak | just paste the url here |
| 15:27.52 | MukulJain | http://pastebin.com/m2da047b5 |
| 15:28.40 | mvanbaak | can you also paste the relevant info for a phone in sip.conf and extensions.conf ? |
| 15:31.35 | *** join/#asterisk tsurko (n=tsurko@78.90.100.214) |
| 15:31.45 | MukulJain | done |
| 15:32.05 | MukulJain | http://pastebin.com/d6dc98eb8 |
| 15:33.10 | *** join/#asterisk af_ (n=getsmart@81-174-47-128.dynamic.ngi.it) |
| 15:33.32 | MukulJain | Has it to do with Call Waiting ? |
| 15:33.50 | MukulJain | Because lights are on, and my Watcher count increases as IP phones are turned on |
| 15:34.00 | MukulJain | But the status on Asterisk is always IDLE |
| 15:34.20 | MukulJain | whereas it shld change to something like :Busy isnt it ? |
| 15:35.15 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 15:36.21 | mvanbaak | what is the hint for the sip phone ? |
| 15:36.28 | mvanbaak | you only pasted the hint for the zap line |
| 15:37.27 | MukulJain | hint is same as extension nos for the IP Phones |
| 15:37.46 | MukulJain | I am using Trix : which auto create hint for each extension |
| 15:37.51 | MukulJain | a sec let me show |
| 15:38.04 | mvanbaak | ah, trixbox |
| 15:38.07 | mvanbaak | that's evil ;) |
| 15:38.21 | MukulJain | ;) u dont seems to like that |
| 15:38.30 | mvanbaak | no, I dont like it |
| 15:38.37 | mvanbaak | most people here dont like it |
| 15:39.37 | MukulJain | I know, I am new to ASterisk ! learning it still |
| 15:39.44 | MukulJain | to not to depend on that evil ; |
| 15:39.52 | MukulJain | http://pastebin.com/d306aa747 |
| 15:40.26 | MukulJain | That's how hints are auto created for each extension |
| 15:40.34 | MukulJain | for the ZAP, it's not done so I had to do that myself |
| 15:40.40 | mvanbaak | it looks sane indeed |
| 15:40.49 | mvanbaak | you can try with call-limit |
| 15:40.51 | mvanbaak | set it to 1 |
| 15:41.08 | MukulJain | oh but in that case I shall loose call waiting isnt it ? |
| 15:41.12 | Corydon76-dig | Problem with a GUI is that it attracts users who want to be spoonfed... and the last thing we want to do is to dumb it down |
| 15:42.12 | MukulJain | I have option of having 4 calls into my phone |
| 15:42.42 | MukulJain | with Call-limit set to 1, I shall loose functionality of getting second or 3rd call on my phone when I am already on one call ? |
| 15:43.22 | mvanbaak | I think so |
| 15:43.27 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 15:43.33 | mvanbaak | never used it before |
| 15:43.36 | Corydon76-dig | You can talk out of the four corners of your mouth all at once? |
| 15:43.46 | mvanbaak | lol |
| 15:44.01 | Corydon76-dig | Most people I know can only hold a single conversation at once. Even the really smart ones. |
| 15:44.04 | mvanbaak | meh, gnome isn't all that evil |
| 15:44.22 | mvanbaak | Most people already have trouble with a single conversation |
| 15:44.23 | mvanbaak | ;) |
| 15:44.26 | MukulJain | I understand, but then It's useful as u know second call coming in |
| 15:44.34 | MukulJain | u can put first on hold and take second one |
| 15:44.53 | MukulJain | or atleast flash of callerID on screen tells who's on next call |
| 15:44.58 | Corydon76-dig | Then you want 2-line appearance |
| 15:45.24 | MukulJain | GXP 2000 support 4 lines, |
| 15:45.25 | Corydon76-dig | I dunno, my customers are all important enough that I never put any of them on hold to talk to another |
| 15:45.38 | mvanbaak | we use queues for that |
| 15:45.57 | JT | eww, GXP2000 |
| 15:46.05 | MukulJain | Okay, so you mean only way is to limit call to 1 ? |
| 15:46.14 | MukulJain | GXP2K has 4 lines, |
| 15:46.21 | Corydon76-dig | The secretary-general of the UN could call, and if I'm on a call with a customer, he's going to have to wait until I'm done |
| 15:46.30 | mvanbaak | Corydon76-dig: indeed |
| 15:46.34 | MukulJain | Well that is debatable ! |
| 15:46.56 | Corydon76-dig | I would have said president of the US, but I'll put him on hold forever |
| 15:47.17 | MukulJain | in my case its sometimes not Customers, it can be emergency department in our hospital while doc is on call. |
| 15:47.19 | mvanbaak | Corydon76-dig: zapateler will get him. I bet he's not sending callerid |
| 15:48.03 | JT | wait... emergency department of a hospital, and you're using GRANDSTREAMS?! |
| 15:48.06 | MukulJain | So somehow functionality is imptt to have second call coming on phone ! whether to take or not take or redirect to voicemail is on scenario |
| 15:48.33 | Corydon76-dig | If I heard "Please hold for the POTUS", he'd get congestion from the phone company |
| 15:48.53 | MukulJain | So question is : "Can we do this without call limit " ? |
| 15:49.59 | *** join/#asterisk ManxPower (n=manxpowe@74.sub-70-221-199.myvzw.com) |
| 15:54.48 | WilliamK | JT, please say it ain't so |
| 15:55.29 | *** join/#asterisk mohsen (n=chatzill@81.31.160.140) |
| 15:55.56 | WilliamK | I'd expect at least Cisco or Polycom |
| 15:56.01 | JT | WilliamK: heh |
| 15:56.06 | JT | preferably polycom |
| 15:56.23 | *** join/#asterisk pepo-- (n=pepOSX@201.210.227.45) |
| 15:57.13 | MukulJain | Friends the question is not about recommendation on Phones that I need |
| 15:57.20 | MukulJain | the problem is about BLF not working |
| 15:57.29 | JT | whatever, the phones are shite |
| 15:57.31 | MukulJain | Later, I would ask your opinion on phones |
| 15:57.32 | JT | ~gs |
| 15:57.32 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 15:57.37 | JT | ~phones |
| 15:57.38 | jbot | phones is, like, http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. |
| 16:03.46 | WilliamK | MukulJain, I think the problem overall is how your phone/adapter is presenting the calls.... I've used other phones and I know it works fine |
| 16:03.54 | *** join/#asterisk wunderkin (n=wunderki@ip72-223-86-126.ph.ph.cox.net) |
| 16:04.19 | WilliamK | thus why we're telling you the phone itself has issues |
| 16:04.41 | MukulJain | Okay, so you mean that GXP-2000 has issue with the BLF, |
| 16:04.58 | MukulJain | So no one here is using them with BLF I assume :( |
| 16:06.29 | WilliamK | only phones & adapters I'm using myself are Cisco, SNOM, Sipura ATA and I just got an Aastra 9133i |
| 16:06.43 | MukulJain | ic |
| 16:07.27 | WilliamK | and obviously, I'm doing alot of testing for my end-user customers now :) |
| 16:08.31 | *** join/#asterisk t3rror (n=t3rror@adsl-065-005-255-180.sip.owb.bellsouth.net) |
| 16:10.00 | MukulJain | Thanks WilliamK, I am using these GXP for testing |
| 16:10.21 | MukulJain | BLF is working with ZAP lines, but phone status are not changing for the GXP |
| 16:10.29 | MukulJain | I assume that's because it can accept multiple calls |
| 16:11.48 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 16:12.43 | *** join/#asterisk jfg (n=jfg@dyn-83-157-144-166.ppp.tiscali.fr) |
| 16:13.49 | tzafrir | MukulJain, what do you see in 'sip show subscriptions' ? |
| 16:13.52 | CCFL_Man2 | WilliamK: most cisco sip loads give terrible functionality |
| 16:14.02 | tzafrir | (hmmm, is this the right direction?) |
| 16:14.25 | mvanbaak | CCFL_Man2: that's why I use chan_skinny |
| 16:14.49 | CCFL_Man2 | mvanbaak: skinny support really that good in asterisk? |
| 16:15.04 | jfg | hi |
| 16:15.04 | mvanbaak | CCFL_Man2: I really like it yeah. |
| 16:15.19 | mvanbaak | works very stable here |
| 16:15.26 | CCFL_Man2 | mvanbaak: you can always run ccm linux on another box :P |
| 16:15.38 | mvanbaak | CCFL_Man2: no, I want asterisk |
| 16:15.48 | MukulJain | a sec |
| 16:16.14 | jfg | i read on voip-info.org that asterisk cannot resgister on a remove server using MGCP, and i'd like to know if there is someone working on it ? |
| 16:16.14 | mvanbaak | I did patch 1.4 chan_skinny with the hint/voicemail stuff from trunk though |
| 16:16.16 | MukulJain | right now there are no entries |
| 16:16.25 | jfg | *remote |
| 16:16.41 | CCFL_Man2 | mvanbaak: i don't blame you, ccm sucks |
| 16:16.49 | mvanbaak | jfg: I dont think so |
| 16:16.53 | mvanbaak | CCFL_Man2: :) |
| 16:17.14 | jfg | mvanbaak: ok |
| 16:17.15 | jfg | thanks |
| 16:17.29 | mvanbaak | jfg: you can do a search on bugs.digium.org |
| 16:17.34 | mvanbaak | ehm |
| 16:17.34 | t3rror | where could a man find some information on how to improve call quality |
| 16:17.36 | mvanbaak | bugs.digium.com |
| 16:17.41 | jfg | ok |
| 16:17.45 | mvanbaak | but I cant recall a ticket for mgcp there |
| 16:17.48 | t3rror | i have a situatuion where incoming audio sounds great, but outgoing audio is garbled |
| 16:17.50 | russellb | there are no patches for that currently |
| 16:18.04 | russellb | in theory, it wouldn't be hard to add, it is just not something that is asked for very often |
| 16:18.07 | t3rror | not garbled, but choppy |
| 16:18.27 | mvanbaak | t3rror: bandwidth trouble ? |
| 16:18.27 | jfg | russellb: you're talking about mgcp ? |
| 16:18.32 | russellb | jfg: yes |
| 16:18.32 | CCFL_Man2 | mvanbaak: plus, i'm not sure how in the hell you'd ever get trunk lines into ccm |
| 16:18.34 | jfg | ok |
| 16:18.51 | mvanbaak | CCFL_Man2: no idea. I like my asterisk setup |
| 16:19.03 | mvanbaak | I only have a couple of skinny phones, but they work great |
| 16:19.08 | CCFL_Man2 | mvanbaak: you use a T1 card? |
| 16:19.14 | mvanbaak | nope |
| 16:19.16 | mvanbaak | pure voip |
| 16:19.17 | t3rror | mvanbaak: shouldn't be > one call and I have 378k upload |
| 16:19.21 | jfg | i think it's not too hard, but i'm new to asterisk and this functionality interest me a lot, so i'd like to work on it, but i need to learn more about asterisk before |
| 16:19.22 | CCFL_Man2 | ahh |
| 16:19.48 | mvanbaak | CCFL_Man2: I get my calls from 2 IAX2 providers |
| 16:20.41 | t3rror | mvanbaak: the server is a 400mhz P3 w/ 256 MB RAM |
| 16:21.32 | mvanbaak | should be plenty |
| 16:21.39 | CCFL_Man2 | mvanbaak: ahh |
| 16:21.49 | t3rror | mvanbaak : so it is on the low-end but i thought it would be enough for one call. i am just looking for information on how to start troubleshooting it |
| 16:23.17 | CCFL_Man2 | russellb: were you the one who collects the WE phones? |
| 16:33.03 | russellb | CCFL_Man2: nope |
| 16:34.27 | ManxPower | Strom_C / Strom_M might be the one. |
| 16:34.48 | outtolunc | looks more stylish |
| 16:35.24 | t3rror | why do you have to purchase a license to use g729? |
| 16:35.33 | russellb | patents |
| 16:35.36 | ManxPower | t3rror: patent and license issues |
| 16:35.55 | t3rror | is there no way to get a trial of it? |
| 16:36.13 | t3rror | currently i am using g711 and i am getting choppy outgoing audio |
| 16:36.20 | ManxPower | t3rror: if you cannot afford $10 for a 1 channel license then you have no business using Asterisk |
| 16:36.21 | CCFL_Man2 | g711u 4lyf3! |
| 16:36.24 | t3rror | i want to test g729 to see if it fixes my problem |
| 16:36.25 | tzafrir | t3rror, use gsm or speex |
| 16:36.38 | ManxPower | choppy sound is seldom a codec issue. |
| 16:37.00 | t3rror | i don't think it is a codec issue, i think it is a bandwidth issue |
| 16:37.05 | russellb | it is if you don't have the bandwidth for it :) |
| 16:37.24 | t3rror | i don't have a problem paying for the license if it solves my problem |
| 16:37.40 | t3rror | i just don't want to buy something that i might not need |
| 16:37.41 | ManxPower | t3rror: ulaw takes 80kilobits. A modem is 56kilobits. Is your internet service THAT slow. |
| 16:37.47 | tzafrir | t3rror, why not use gsm, ilbc or speex? |
| 16:37.53 | CCFL_Man2 | if you get a pri from a telco the codec they use is g711u, in the US anyway |
| 16:38.04 | tzafrir | or maybe even g726 would prove to be good enough? |
| 16:38.11 | t3rror | my upload is 378k or something |
| 16:38.17 | CCFL_Man2 | vonage used g723 |
| 16:38.18 | WilliamK | CCFL_Man2, the SIP ver I'm using on the 7940 right now is the 7.4 release... thinking about testing the 8.x release |
| 16:38.36 | tzafrir | t3rror, 37kB or 37kb? |
| 16:38.45 | ManxPower | t3rror: you should be able to run FOUR ulaw calls over 384k |
| 16:38.57 | CCFL_Man2 | WilliamK: i'd upgrade, my 7912 is at 8.0.1 right now, the latest |
| 16:39.02 | t3rror | ok, so maybe it is the horsepower of the server |
| 16:39.13 | t3rror | 400mhz P3 with 256MB RAM |
| 16:39.21 | t3rror | running gentoo with minimal services |
| 16:39.22 | ManxPower | t3rror: it could be a thousand things. |
| 16:39.25 | tzafrir | well, any compressed codec would be worse that g711 |
| 16:39.30 | tzafrir | regarding horsepower |
| 16:39.32 | ManxPower | t3rror: you won't be able to run G729 on that server |
| 16:39.32 | WilliamK | reason why I didn't upgrade initially was customer deployment timetable, and I wanted to make sure it worked right |
| 16:39.37 | MukulJain | G711 is ulaw right ?? |
| 16:39.42 | WilliamK | now I have 3 spares here, so I can play with those |
| 16:39.50 | Corydon76-dig | 711 is both ulaw AND alaw |
| 16:39.51 | tzafrir | t3rror, but there may still be a bandwidth issue, only not in the link from you to your ISP |
| 16:40.06 | CCFL_Man2 | WilliamK: you don't trust cisco!? what a suprise! :P |
| 16:40.17 | t3rror | i have the server connected directly to the router (WRT54GL) |
| 16:40.22 | ManxPower | MukulJain: G711u aka ulaw aka PCMU , G711a aka alaw aka PCMA |
| 16:40.27 | WilliamK | been around the block a time or two with cisco :) |
| 16:40.36 | t3rror | so it should be at least 10MBit connection if not 100 |
| 16:40.50 | Corydon76-dig | Yeah, Cisco really takes you for a ride |
| 16:40.59 | CCFL_Man2 | g711alaw is used in broken countries like europe :P |
| 16:41.13 | CCFL_Man2 | WilliamK: who hasn't |
| 16:41.14 | ManxPower | Corydon76-dig: I experienced my first major Cisco IOS bug the other day. |
| 16:41.15 | *** join/#asterisk JoelSolanki (i=Joel@220.224.119.167) |
| 16:41.31 | MukulJain | Just want to check something, |
| 16:41.32 | t3rror | tzafrir: i can't use those codecs because the ATA (PAP2) doesn't support them |
| 16:41.51 | MukulJain | 2 endpoints using same Codecs, does the RTP traffic still pass thru the ASterisk server ? |
| 16:41.57 | CCFL_Man2 | t3rror: it supports them i think |
| 16:42.04 | ManxPower | Corydon76-dig: the IOS release we wre using is not even available from Cisco anymore. I assume it was just too badly broken. |
| 16:42.27 | tzafrir | t3rror, try g726 |
| 16:42.28 | ManxPower | CCFL_Man2: Linksys supports alaw, ulaw, g726, g729 and I believe G723.1 |
| 16:42.36 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 16:42.50 | CCFL_Man2 | ManxPower: ahh, vonage uses g723 |
| 16:42.58 | tzafrir | t3rror, and see if there's any improvement. Also note that the PAP2 can only use one g729 call at a time |
| 16:43.10 | ManxPower | there was some issue with regards to G726 on the Linksys. Had something to do with the SDP or RTP media type or something like that. |
| 16:43.21 | t3rror | g711u&a g726-16-40 and g723 |
| 16:43.30 | ManxPower | There is a #define in Asterisk to make them work togather. I don't know if Linksys ever fixed their firmware for this issue. |
| 16:43.31 | t3rror | i only want one call at a tiem |
| 16:43.49 | WilliamK | tzafrir, that really sucks... thought Sipura fixed that on the 2002's |
| 16:43.57 | ManxPower | t3rror: asterisk does not suport G723.1 in any useful way for most people |
| 16:44.06 | tzafrir | t3rror, alternatively, try a soft phone . compare its behaviour with ulaw to its behaviour with gsm |
| 16:44.27 | CCFL_Man2 | g711u should be fine over a lan |
| 16:44.31 | tzafrir | t3rror, also, does the phone connect directly, or is proxied through Asterisk? |
| 16:44.42 | ManxPower | the issue with G729 is one of two. G729 takes large amounts of CPU power, perhaps the device is underpowered. Also Linksys has to pay a license for every simul G729 call, so maybe it was a cost saving measure. |
| 16:44.57 | ManxPower | or maybe both 8-) |
| 16:45.01 | tzafrir | t3rror, if it is proxied through asterisk, then use ulaw/alaw on the LAN and a compressed codec on the WAN |
| 16:45.53 | WilliamK | ManxPower, at least they should make 2 different models and let you have the option to pay more |
| 16:46.24 | CCFL_Man2 | ManxPower: i never liked the pap2 anyway |
| 16:47.06 | MukulJain | CCFL_Man2 : What do you suggest for ATA Adapter ? |
| 16:47.09 | ManxPower | CCFL_Man2: Never used it. Always used the SIPura branded models. Liked them quite a bit, but just don't need them for most of my applications |
| 16:47.21 | CCFL_Man2 | ManxPower: ahh |
| 16:47.29 | ManxPower | The SIPura/Linksys boxes are some of the best out there. |
| 16:47.33 | CCFL_Man2 | MukulJain: an adit 600 channel bank |
| 16:47.48 | MukulJain | CCFL_Man2 : For 2 Ports Solution ? |
| 16:48.38 | CCFL_Man2 | MukulJain: only if those 2 lines are required at the same facility as the other lines |
| 16:49.02 | *** part/#asterisk jfg (n=jfg@dyn-83-157-144-166.ppp.tiscali.fr) |
| 16:49.35 | ManxPower | Generally if we need a POTs line we order a POTS line. |
| 16:49.47 | ManxPower | simple, easy, no drama, reliable |
| 16:50.09 | CCFL_Man2 | ManxPower: channel banks make multiple extentions cheap :P |
| 16:50.19 | MukulJain | Question ----> IF I have 2 Remote Phones using G729, and they call each other. Will the RTP will pass thru Asterisk or they will talk end to end with only SIP packets been sent to server ?? |
| 16:50.52 | CCFL_Man2 | everything will go through asterisk |
| 16:52.17 | CCFL_Man2 | ManxPower: most business workers just need a regular phone and caller id box anyway |
| 16:52.28 | MukulJain | IS it possible to have 2 Phones RTP traffic directly and only SIP to go to ASterisk |
| 16:52.35 | WilliamK | anyone have a performance idea of how many calls the quad core 2.4ghz cpu can handle? |
| 16:52.36 | MukulJain | this should be more effective isnt it ? |
| 16:52.48 | MukulJain | Because Bandwidth requirement would be lower ? |
| 16:53.06 | MukulJain | Else all remote phone inter-communications are taking bandwidth on the server side ? |
| 16:53.29 | CCFL_Man2 | you can dial the other extention directly |
| 16:54.28 | tzafrir | WilliamK, highly dependent on the codec |
| 16:54.46 | *** join/#asterisk shido6 (n=shido6@74-130-227-15.dhcp.insightbb.com) |
| 16:54.57 | WilliamK | first idea that comes to mind would be g729 |
| 16:55.03 | WilliamK | I like to go for worst-case |
| 16:55.04 | WilliamK | :) |
| 16:55.22 | CCFL_Man2 | with women too? |
| 16:55.24 | tzafrir | WilliamK, if oyu would be using g711, the bottleneck would likely be the networking stack |
| 16:55.53 | MukulJain | CCFL_Man2 : lol |
| 16:56.28 | tzafrir | WilliamK, if you'd like to go for worst case, try speex. Though maybe it got optimized in the recent year or two (it has gone through massive work). ilbc is also a CPU consumer |
| 16:56.41 | CCFL_Man2 | personally, i've always liked the ones with big asses |
| 16:56.50 | WilliamK | CCFL_Man2: now that's bad, and heck no.... only want the nice ones that can at least qualify to standards :) |
| 16:57.08 | CCFL_Man2 | WilliamK: lol |
| 16:57.14 | MukulJain | CCFL_Man2: What's your worse case ? ;) Lol |
| 16:57.31 | ManxPower | if the codecs for the 2 legs of the call are the same, there is NO NAT involved, you do not have t/T/w/W or other Dial options that makes Asterisk mointor the audio, and you do not have canreinvite=no, then the RTP audio should go directly between the two devices by default. |
| 16:57.52 | CCFL_Man2 | heh, probably liking black guys if she's white |
| 16:57.55 | WilliamK | g729 possible to do say 120calls with the quad core, or limited to say 96? |
| 16:58.21 | ManxPower | WilliamK: nobody knows. |
| 16:58.23 | CCFL_Man2 | ManxPower: ahh, never knew that |
| 16:58.48 | MukulJain | ManxPower : canreinvite=no, what functionality we loose by using this option |
| 16:58.53 | ManxPower | unfortunatly when reinvites happen it is common to lose the first 1/2 second of audio. Doesn't sound like much, but the issue is QUITE annoying. |
| 16:59.13 | MukulJain | Sometime back I read some Cisco paper, which said that RTP streams are end to end whereas IP Phone to Server it's only SIP (Signalling). |
| 16:59.15 | ManxPower | MukulJain: you lose the ability to have the RTP audio go direct between the two endpoints |
| 16:59.27 | WilliamK | Manx, I love being the one testing to find out :) |
| 16:59.48 | CCFL_Man2 | oh yeah, i think rtp audio is peer to peer |
| 16:59.51 | *** join/#asterisk bbryant (n=12243@c-68-59-20-153.hsd1.sc.comcast.net) |
| 17:00.03 | WilliamK | I just didn't wanna really spend the extra $ on the board with the echo cancel on it right off the bat if I didn't need to |
| 17:00.17 | ManxPower | CCFL_Man2: that is the ideal, in the real world RTP audio is frequently NOT peer-to-peer |
| 17:00.19 | MukulJain | I am sorry, I wanted to ask If we have canreinvite=yes, what do we loose in terms of functionality |
| 17:00.31 | ManxPower | WilliamK: echo cancel has to be done at the PSTN/IP interface. |
| 17:00.40 | t3rror | tzafrir: i was able to register the pap2 with teliax directly and there was not an issue with audio, the audio issue has cropped up since the addition of the asterisk server as a proxy |
| 17:00.41 | CCFL_Man2 | ManxPower: ahh |
| 17:00.46 | ManxPower | MukulJain: canreinvite=yes will not work if you have any NAT involved. |
| 17:00.52 | WilliamK | Manx, yep and that was where I was refering to |
| 17:01.01 | MukulJain | oh but most of the times the IP Phones are using NAT |
| 17:01.09 | WilliamK | thinking of getting a quad T1 card and putting it into a quad core box |
| 17:01.11 | MukulJain | We never connect them directly to the Broadbands |
| 17:01.18 | ManxPower | MukulJain: none of the 200 or so phones I manage use NAT |
| 17:01.20 | MukulJain | they are always behing a router |
| 17:01.31 | MukulJain | Are they all using Public IP addreses ? |
| 17:01.36 | tzafrir | t3rror, hmmm, run 'sip show channels' when in a call. What codecs do you use? |
| 17:01.36 | ManxPower | Then again, none of those phones talk to the outside world via IP |
| 17:01.41 | CCFL_Man2 | WilliamK: quad T1 cards are expensive, probably cost as much as the box |
| 17:02.09 | ManxPower | MukulJain: IP phones do not require a connection to the internet. |
| 17:02.18 | MukulJain | Remote IP Phones |
| 17:02.20 | WilliamK | CCFL_Man2, yeah and I'm protesting it highly |
| 17:02.26 | MukulJain | Connecting over Internet back to Asterisk |
| 17:02.30 | ManxPower | none of my customers ever send voice over the internet. |
| 17:02.33 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 17:02.39 | MukulJain | I see, |
| 17:02.40 | ManxPower | MukulJain: in your case then you can't do reinvites. |
| 17:02.52 | MukulJain | I have 2 phones which are remote, |
| 17:02.53 | CCFL_Man2 | WilliamK: a cheap solution i found was to use a cisco router with T1 card |
| 17:02.56 | ManxPower | Polycom Phone <-> Asterisk <-> PSTN PRI |
| 17:03.00 | MukulJain | Let's say Server in Dallas and phones in NY and LA |
| 17:03.04 | CCFL_Man2 | WilliamK: something used |
| 17:03.13 | WilliamK | I'm pulling 2 PRIs in |
| 17:03.17 | WilliamK | to start |
| 17:03.24 | MukulJain | when NY calls LA, I assume today it's like NY ---> DALLAS ---> LA |
| 17:03.32 | MukulJain | Whereas it should be NY-->LA for the RTP |
| 17:03.38 | MukulJain | for the optimized scenario |
| 17:04.20 | CCFL_Man2 | WilliamK: a router or voice gateway with two T1 interfaces |
| 17:04.20 | ManxPower | You, of course, have a WAN between the locations. T-1 or Frame Relay or something else? |
| 17:04.20 | CCFL_Man2 | WilliamK: or, a gateway per T1 |
| 17:04.33 | MukulJain | ManxPower: NY and LA are using Broadband connections |
| 17:04.49 | ManxPower | MukulJain: that word has no meaning. Perhaps you mean cable/dsl |
| 17:04.51 | MukulJain | and they are using Router, IP phones are on private IP's and ofcourse using NAT |
| 17:04.53 | WilliamK | CCFL, my experience is that the quad/dual port board is cheaper than the seperate gw idea |
| 17:04.57 | MukulJain | yes, sorry, Cable / DSL |
| 17:05.04 | CCFL_Man2 | WilliamK: i have an old cisco mc3810, it's perfect for such an application, but the ethernet interface is only 10mbit |
| 17:05.22 | CCFL_Man2 | WilliamK: i'm talking used gateways |
| 17:05.51 | CCFL_Man2 | since they are easier to find than used T1 cards |
| 17:07.04 | CCFL_Man2 | i use mine the opposite way, using sip trunks to terminate to fxs ports with it's T1 interface and a channel bank |
| 17:07.20 | ManxPower | Cisco routers need two things to handle IP/PSTN interface. Three things, actually. The correct IOS feature set, the T-1 card, and the DSP chips All of these are expensive. |
| 17:07.52 | CCFL_Man2 | ManxPower: can you get dsp addons for the routers that support wics? |
| 17:08.03 | ManxPower | it ends up being more expensive than just using a digium or sangoma card. |
| 17:08.14 | Sweeper | well |
| 17:08.17 | Sweeper | depends |
| 17:08.19 | MukulJain | CCFL_Man2 : Yes, you need to purchase DSP (PVDM's Modules) for the Router |
| 17:08.31 | Sweeper | you can get like 8 t1's into a cisco |
| 17:08.32 | MukulJain | CCFL_Man2: Depends how many active channel |
| 17:08.35 | CCFL_Man2 | ahh, thars right |
| 17:08.40 | ManxPower | CCFL_Man2: it depends on the router. The DSP chips go into connectors on the router motherboard, like the RAM or FLASH. |
| 17:08.53 | CCFL_Man2 | ahh, yes |
| 17:09.01 | ManxPower | so if the router does not support the DSP sticks..... |
| 17:09.12 | CCFL_Man2 | my 1721 does not |
| 17:09.16 | ManxPower | When I first started with VoIP I made the mistake of trying to use Cisco for the IP/PSTN stuff. |
| 17:09.24 | ManxPower | That was an expensive mistake. |
| 17:09.24 | MukulJain | ManxPower : Most of the Voice Enabled routers will hv slots for the DSP's |
| 17:09.48 | ManxPower | CCFL_Man2: I had a 1721 and it had DSP connectors, maybe it was the 1721V or something like that. |
| 17:09.58 | MukulJain | ManxPower: Which one is a good PSTN/IP MEdia Gateway, for say 2 ports only ? |
| 17:10.00 | t3rror | gentoo*CLI> sip show channels |
| 17:10.00 | t3rror | Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message |
| 17:10.00 | t3rror | 192.168.1.115 pap1 38537a5b-15 00101/00102 ulaw No Rx: ACK |
| 17:10.00 | t3rror | 1 active SIP channel |
| 17:10.11 | ManxPower | MukulJain: For 2 ports I would use a SIPura |
| 17:10.14 | MukulJain | I want them to be installed over Internet at another country and talking back to Linux |
| 17:10.26 | MukulJain | Okay, I am using Linksys 3102, which is SIPURA earlier I believe |
| 17:10.31 | MukulJain | but having echo issues. |
| 17:10.37 | ManxPower | Linksys and SIPura are pretty much the same. |
| 17:10.43 | ManxPower | MukulJain: echo is always an issue. |
| 17:10.59 | ManxPower | anything with decent echo canceling will be massivly expensive |
| 17:12.07 | CCFL_Man2 | ManxPower: same here, but i used an older access concentrator, only $46, but i needed a $40 prom, 64mb ram, the required IOS, a voice dsp module i got for $15 total, and a $40 T1 DVM, though i could have used the T1 trunk card already in the router |
| 17:12.52 | t3rror | tzafrir: i just tried using g723 and the asterisk server said this: Sep 9 12:11:32 NOTICE[4004]: chan_sip.c:3775 process_sdp: No compatible codecs! |
| 17:13.24 | tzafrir | t3rror, g726, not g723 |
| 17:13.39 | tzafrir | t3rror, but I suspect you're using gsm to your ISP |
| 17:13.42 | ManxPower | t3rror: you cannot use G723 with Asterisk in any useful way |
| 17:13.56 | tzafrir | or something similar. And hence the lower quality |
| 17:14.14 | CCFL_Man2 | ManxPower: but, for being old and only 40Mhz, it works pretty nice, but i'm limited with the choice channels with the VCM6 i got |
| 17:14.37 | t3rror | just tried g726-40 and got the same no compatible codecs message from the asterisk console |
| 17:14.38 | CCFL_Man2 | choice = voice |
| 17:16.12 | CCFL_Man2 | ManxPower: plus the free adit 600 channel bank i got with original software load and original fxs cards, but it seems to work really well, atlease with tollfreegateway |
| 17:17.25 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 17:17.53 | tzafrir | t3rror, you're not listening |
| 17:19.22 | CCFL_Man2 | too much earwax in the receiver speaker holes probably |
| 17:22.16 | tzafrir | t3rror, go back to ulaw |
| 17:22.41 | CCFL_Man2 | the telco's use it for a reason |
| 17:23.30 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 17:27.20 | RypPn | is it standard practise for asterisk 1.4 not to load up module chan_sip.so ? |
| 17:28.10 | *** join/#asterisk MukulJain (n=jainmuku@cm69.omega97.maxonline.com.sg) |
| 17:30.15 | [TK]D-Fender | RypPn, definately not. Go look that you have it, then check your modules.conf |
| 17:30.58 | RypPn | [TK]D-Fender: yes, I added it to modules.conf under the globals section and all is well now, I just wondered if that was normal |
| 17:31.10 | RypPn | never had to do it with 1.2 |
| 17:31.38 | [TK]D-Fender | RypPn, check your autoload option |
| 17:31.48 | RypPn | its set to yes |
| 17:32.40 | CCFL_Man2 | anyone know how to calibrate an old western electric dial? |
| 17:32.55 | [TK]D-Fender | RypPn, ok, makes no sense unless chan_sip.so simply failed to load due to a port conflict or soemthing |
| 17:34.12 | RypPn | hmmm, I'll just let it slide I think, my concern was that other modules may need manually loading as well now |
| 17:35.00 | RypPn | 153 loaded, that in the ballpark? |
| 17:39.44 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 17:41.24 | *** join/#asterisk ant (n=ant@12-167-225-221.pf-cvl.net) |
| 17:41.35 | *** join/#asterisk Sci_05 (i=Sci_05@ts.bestserversllc.net) |
| 17:42.16 | *** part/#asterisk ant (n=ant@12-167-225-221.pf-cvl.net) |
| 17:46.15 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 17:46.26 | shmaltz | SeaMonkey is way nicer than Mozilla |
| 17:49.08 | *** join/#asterisk jdg (n=jdg@203.185.177.240) |
| 17:53.53 | *** part/#asterisk cavediver (i=jonas@trimix.eklof.eu) |
| 17:59.54 | GlobeTrotter | hi,, i need to upgrade my machine,, but i want to transfer my g729 codecs to my new box... how do i do that> |
| 18:00.31 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 18:01.02 | jdg | Ask relicense to Digium ? |
| 18:02.01 | GlobeTrotter | so i NEED to contact digium if i want to move the codecs from onx to the other?e o |
| 18:02.11 | GlobeTrotter | from one box to the other |
| 18:04.06 | jdg | yes, I think so, but I may be wrong. |
| 18:04.25 | [TK]D-Fender | GlobeTrotter, Yes, because its tied to your primary NIC's MAC address |
| 18:04.32 | Qwell | s/primary// |
| 18:04.37 | *** join/#asterisk TmBerg (n=TmBerg@pdpc/supporter/basic/TmBerg) |
| 18:04.42 | GlobeTrotter | ok,, thanks |
| 18:04.51 | Qwell | and I think you can do it once without having to call |
| 18:08.14 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@032-454-146.area7.spcsdns.net) |
| 18:09.31 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-79-178-30-210.red.bezeqint.net) |
| 18:09.33 | *** join/#asterisk EclecticRob (n=Eclectic@24-176-222-123.static.lnbh.ca.charter.com) |
| 18:11.45 | t3rror | tzafrir: i was using ulaw, but that is where i have audio problems |
| 18:14.59 | CCFL_Man2 | on the lan? |
| 18:16.37 | [X-tp] | is it possible to get H.323 and video to work in asterisk? |
| 18:28.48 | *** join/#asterisk pepo-- (n=pepOSX@201.210.227.45) |
| 18:29.12 | tzafrir_laptop | t3rror, are you sure you use ulaw to your ISP? |
| 18:29.52 | *** join/#asterisk implicit (n=implicit@210.16.55.38) |
| 18:29.55 | t3rror | tzafrir_laptop: i am using iax2 to teliax and sip to my ata |
| 18:30.00 | swiftkick | hello, I have a question. using Asterisk-GUI to manage a multiple provider, multiple tenant PBX. i've been poking around in extensions.conf fine tuning a few things, voice menus, etc. and have run into an odd problem. |
| 18:30.35 | swiftkick | i have 4 DID trunks each that routes to a ring group. the ring groups all ring the proper phones based on their DID context. |
| 18:30.55 | swiftkick | however, when the ring group times out its supposed to transfer the call to an appropriate VMB. |
| 18:31.22 | swiftkick | for some reason, two of the DID trunk contexts don't seem to "know" about the voicemail boxes. |
| 18:31.45 | swiftkick | i.e. at that point the line exten = s,n,Voicemail(37,b) fails nonzero |
| 18:31.53 | swiftkick | even though there is definitely a voicemail box 37 |
| 18:33.03 | swiftkick | and the caller gets the message "the mailbox you are trying to access does not exist" |
| 18:33.27 | *** join/#asterisk pejo_ (n=pete-joh@triton.dsv.su.se) |
| 18:33.31 | swiftkick | then they are prompted to enter a voicemail box #. Well, at that point, no matter what they enter, they cannot connect. |
| 18:34.26 | swiftkick | i.e. the prompt cannot route to any of the vmb's for any of the "tenants", whereas internally, any one person can directly dial any other tenant's employees' extensions & ring through to their VMB's no prob |
| 18:41.16 | swiftkick | im trying to find some documentation that might explain some subtlety of users.conf or extensions.conf that i've missed, but to no avail. |
| 18:44.18 | *** join/#asterisk mmdk (n=mm@0x555281d0.adsl.cybercity.dk) |
| 18:45.26 | t3rror | what free softphone for windows so you all suggest? |
| 18:46.05 | mmdk | i don't have good experience with softphones |
| 18:46.17 | mmdk | it's always been crap for me when it comes to sound |
| 18:46.20 | t3rror | i just need something to test with internally |
| 18:46.28 | mmdk | but maybe somebody can comment on that |
| 18:46.34 | mmdk | i guess you can try x-lite |
| 18:46.41 | t3rror | i couldnt get it working right |
| 18:46.52 | t3rror | it wouldn't register with the server |
| 18:46.58 | mmdk | hmmm....usually it works fine |
| 18:47.07 | mmdk | look at the log and see what it says |
| 18:47.09 | t3rror | the setup screen kept popping up |
| 18:47.16 | mmdk | or at the asterisk log |
| 18:47.24 | Sweeper | t3rror: eh? the sip client one? |
| 18:47.30 | mmdk | yeah |
| 18:47.33 | Sweeper | sounds like you need to reinstall or something |
| 18:48.01 | *** join/#asterisk pepo-- (n=pepOSX@201.210.227.45) |
| 18:48.21 | t3rror | i will try that |
| 18:50.22 | WilliamK | what are most users using for VoIP billing software? |
| 18:52.58 | *** join/#asterisk mordaunt (n=mordaunt@unaffiliated/mordaunt) |
| 18:55.10 | mmdk | anybody that can help with this ? I have set up asterisk. i can call out on my SIP trunk but i cannot receive calls. i can see that the call reaches asterisk but it does not forward to my client. is there any debug option i can enable to see what asterisk tries to do with that call ? |
| 18:56.04 | mmdk | tried SIP debug but nothing usefull came out of that |
| 18:56.47 | jdg | set |
| 18:56.51 | jdg | core set verbose 3 |
| 18:57.20 | mmdk | ok let me try that....thanx |
| 18:57.29 | t3rror | mmdk: i needed a newer version of xlite |
| 18:57.36 | t3rror | mmdk: i reinstalled and it worked fine |
| 18:57.46 | mmdk | that's great |
| 18:59.03 | mmdk | hmmm.....nothing usefull or i can't the useful thing :( |
| 18:59.34 | *** join/#asterisk JoelSolanki (n=joel@220.224.116.215) |
| 18:59.39 | JoelSolanki | Hi all. |
| 19:00.00 | JoelSolanki | can we have extensions authenticate against unix password file ? |
| 19:00.37 | *** part/#asterisk WizardWlf (n=shawn@wsip-70-167-225-171.om.om.cox.net) |
| 19:02.07 | [TK]D-Fender | JoelSolanki, Extensions don't authenticate anything, they simply "are". Be careful of your wording. |
| 19:04.01 | JoelSolanki | sorry. i mean what i set in sip.conf |
| 19:04.43 | JoelSolanki | when i create a user in sip.conf it authenticates but i want to make that authentication with unix password file |
| 19:04.45 | JoelSolanki | is that possible ? |
| 19:05.42 | *** join/#asterisk rkioko (n=rkioko@41.206.48.74) |
| 19:06.10 | jdg | no, I don't think so |
| 19:06.16 | WilliamK | JoelSolanki, the problem I see with the idea is that the passwords are usually MD5 |
| 19:06.30 | [TK]D-Fender | JoelSolanki, You have sip.conf, and you have real-time DB's. How you want to sync THAT with Unix PW's is YOUR job. There is no tool I've ever heard of to do this already. Fell free to WRITE ONE. |
| 19:06.36 | [TK]D-Fender | feel* |
| 19:06.46 | WilliamK | you'd have to run a touch script or something to extract the data and then break the MD5, and reparse it into a file formatted properly |
| 19:07.57 | WilliamK | too much trouble/pain |
| 19:09.05 | JoelSolanki | hmm agree. |
| 19:09.24 | JoelSolanki | this looks hard. |
| 19:09.32 | JoelSolanki | but let me give one link |
| 19:10.02 | JoelSolanki | http://www-rocq.inria.fr/who/Philippe.Sultan/Asterisk/asterisk_sip_external_authentication.html |
| 19:10.19 | WilliamK | JoelSolanki, not alot of trouble if you're a genuine programmer with an idea/concept, however; if you're not, well... that's where the problems really begin |
| 19:10.30 | swiftkick | ok: this is probably a total noob question, but, in extensions.conf is there a difference between the nomenclature "exten =" and "exten =>"? |
| 19:10.34 | JoelSolanki | yes i understand. |
| 19:10.45 | WilliamK | Radius is one thing, Unix password file is another |
| 19:10.54 | WilliamK | Radius usually isn't stored in MD5 |
| 19:11.12 | JoelSolanki | yes |
| 19:11.33 | JoelSolanki | but check the PAM authentication method in that link |
| 19:11.36 | swiftkick | i have been digging thru google trying to find some really simple answers re: users.conf and extensions.conf and have yet to see that addressed... |
| 19:12.02 | WilliamK | JoelSolanki, I looked |
| 19:12.18 | JoelSolanki | WilliamK: Ok. i m starting to work on it. :) |
| 19:12.31 | WilliamK | you're welcome to try and set it up, I just don't have that amount of time |
| 19:12.32 | WilliamK | :) |
| 19:13.08 | JoelSolanki | hope it works out for me. need create a central billing system so asterisk users authenticate against unix password file. |
| 19:13.44 | JoelSolanki | ok take care. bye |
| 19:14.19 | *** join/#asterisk saftsack (n=oliver@p54A7CFFD.dip.t-dialin.net) |
| 19:14.41 | WilliamK | anyone know if AstBill is supported with * 1.4? |
| 19:14.52 | WilliamK | I keep seeing references to 1.0/1.2 |
| 19:16.51 | *** join/#asterisk BrokenArrow (n=Lp@wikipedia/BrokenArrow) |
| 19:17.26 | *** join/#asterisk wundaboy (n=pat@pool-71-111-176-117.ptldor.dsl-w.verizon.net) |
| 19:17.56 | wundaboy | What provider to people recommend for termination/origination? |
| 19:18.41 | wundaboy | i'm currently using Junction networks and it is ridiculously expensive $.029 |
| 19:21.20 | swiftkick | is there a difference between the nomenclature "exten =" and "exten =>" in extensions.conf ? I cant seem to find this documented anywhere, yet asterisk-gui seems to freely write exten =, whereas all the docs online suggest exten =>. |
| 19:23.06 | swiftkick | e.g. http://www.voip-info.org/wiki/index.php?page=Asterisk+config+extensions.conf makes no mention of "exten =" |
| 19:24.24 | mmdk | can someone enlight my knowledge...what doe insecure=very mean ? |
| 19:25.10 | russellb | swiftkick: there is a difference in the code, in that you can tell what was used, but i don't know of anywhere that actually cares which you use |
| 19:25.44 | swiftkick | russelb: aha... so its just a coding convention, essentially? |
| 19:25.51 | russellb | basically, yeah |
| 19:25.56 | pejo_ | Does asterisk have video-mail-app? |
| 19:26.03 | swiftkick | russelb: thank you :) |
| 19:26.07 | pejo_ | like voice mail, but with video? |
| 19:26.09 | mmdk | anyone ? |
| 19:26.14 | russellb | pejo_: app_voicemail can handle video just fine, or it is supposed to anyway :) |
| 19:26.21 | russellb | mmdk: have you looked at sip.conf.sample ? |
| 19:26.34 | russellb | mmdk: "very" is the same as "port,invite" |
| 19:27.01 | russellb | which means 1) allow matching peer on IP and not port. and 2) do not require authentication for incoming INVITEs |
| 19:27.46 | mmdk | no sorry din't see that but thank you for the info |
| 19:27.56 | russellb | you're welcome |
| 19:28.49 | russellb | pejo_: it doesn't require any additional configuration, either. if you call to leavea voicemail with a video phone, it will save the video as well, and will play it back to you when you call in with a compatible phone to check it |
| 19:31.17 | pejo_ | russellb: great, can each user define their own videos to be played when they can answer |
| 19:31.24 | *** join/#asterisk apardo (n=apardo@202.144.217.87.dynamic.jazztel.es) |
| 19:31.35 | *** part/#asterisk apardo (n=apardo@202.144.217.87.dynamic.jazztel.es) |
| 19:31.36 | mmdk | anyone tried 1videoconference with astersik ? |
| 19:31.48 | pejo_ | can't answer |
| 19:32.18 | jdg | videoconference is not supported, just direct video calls |
| 19:32.44 | mmdk | 1videoconference should work with Asterisk....at least that's what they say |
| 19:33.15 | pejo_ | like a normal answer machine you get to record a message that the machine should playback when no one is their to answer the call, do asterisk have that feature? |
| 19:34.10 | jdg | what is 1videoconference ? |
| 19:35.04 | [X-tp] | jdg: how do you define a videoconference in this case? multiple remote sites or H.323? |
| 19:35.29 | mmdk | look here : http://www.vmukti.com/ |
| 19:35.59 | mmdk | i tried to install and configure it but i can't get it to work. have some problem with the mail it suppose to send when you create a meeting |
| 19:36.10 | mmdk | but take a look. it looks cool if it works |
| 19:36.18 | jdg | for me a conference is app_meetme |
| 19:36.27 | [X-tp] | ok |
| 19:36.56 | jdg | mmdk: so it's not asterisk ! |
| 19:37.10 | mmdk | no....it uses asterisk conference feature....don't really know how |
| 19:37.19 | mmdk | that's why i wanted to try it out |
| 19:37.39 | mmdk | but maybe somebody else would like to try it and has some feedback |
| 19:37.57 | *** join/#asterisk JunK-Y (n=junky@modemcable183.17-83-70.mc.videotron.ca) |
| 19:38.26 | swiftkick | where is the proper place to specify users voicemail information? using asterisk-gui to set things up, it doesn't seem to write a thing to voicemail.conf. it does appear to update the entries in users.conf regarding voicemail. |
| 19:42.05 | [TK]D-Fender | swiftkick, a stupid flag in users.conf : hasvoicemail = yes |
| 19:45.37 | *** join/#asterisk cavediver (i=jonas@trimix.eklof.eu) |
| 19:45.39 | cavediver | Hi guys. |
| 19:46.20 | cavediver | I have added a service provider using the gui and even though i run activate it seems to be gone from the gui but it still registers to my provider. |
| 19:46.56 | cavediver | I don't know how to get rid of it. I have now two identical providers, but I see only one in the gui. Has this happened to anyone else |
| 19:48.09 | *** join/#asterisk hugelmopf (n=frank@dslb-088-073-239-101.pools.arcor-ip.net) |
| 19:49.34 | cavediver | Noone? |
| 19:50.49 | ManxPower | cavediver: we don't use or support any GUIs here. |
| 19:51.02 | cavediver | I see. |
| 19:51.13 | ManxPower | I suggest going to the channel for the gui you are using |
| 19:51.25 | cavediver | Ok, it would be the official one. |
| 19:51.28 | ManxPower | current topic is: Asterisk: The Open Source Telephony Application Platform -=- Asterisk 1.4.11 (Aug. 21, 2007) -=- Join #asterisknow or #asterisk-gui for AsteriskNOW and Asterisk-GUI info -=- Join #asterisk-commits to monitor svn changes -=- Join #freepbx for freepbx/#trixbox for trixbox support. |
| 19:51.38 | cavediver | I see. |
| 19:51.58 | *** part/#asterisk cavediver (i=jonas@trimix.eklof.eu) |
| 19:53.35 | *** join/#asterisk zapp-branigan (n=zapp-bra@9.218.216.87.static.jazztel.es) |
| 19:55.16 | *** join/#asterisk rkioko (n=rkioko@41.206.48.74) |
| 20:04.04 | *** join/#asterisk swiftkick (i=x@c-67-167-211-153.hsd1.il.comcast.net) |
| 20:04.17 | swiftkick | ok |
| 20:04.33 | swiftkick | [TK]D-Fender thank you for your answer re: voicemail. |
| 20:05.07 | swiftkick | I do have "hasvoicemail = yes" set properly on the entries in users.conf |
| 20:05.59 | swiftkick | thes problem is, with 4 DID trunks, all 4 route properly to respective ringroups in extensions.conf, and ring multiple SIP extensions correctly. but then when ringroups time out they are supposed to transfer to a voicemail box. 2 of them do so flawlessly. the other two exit nonzero (according to the CLI) and tell the caller: "the mailbox you are trying to reach does not exist." |
| 20:06.10 | swiftkick | then "Please enter the mailbox you'd like to reach" |
| 20:06.42 | swiftkick | no matter what the caller enters at that poiont does not display any entries on the CLI screen |
| 20:07.50 | swiftkick | then no matter what existing extensions or VMB's they dial at that prompt, it repeats and hangs up |
| 20:08.03 | swiftkick | so its as though, none of the VMB system is accessible from the DID_trunk context (?!) |
| 20:08.11 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 20:08.32 | *** join/#asterisk diemaco (n=diemaco@unaffiliated/diemaco) |
| 20:08.53 | [TK]D-Fender | ~pb |
| 20:08.54 | 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 |
| 20:08.58 | [TK]D-Fender | there we go |
| 20:09.14 | [TK]D-Fender | swiftkick, that output doesn't prove WHERE things have been screwed up. |
| 20:09.29 | [TK]D-Fender | swiftkick, the CONFIG is clearly wrong, its jsut a matter of WHERE |
| 20:09.57 | swiftkick | [TK]D-Fender: no doubt. whats a concern is that most oif this config was auto-generated by asterisk-gui. |
| 20:10.04 | swiftkick | you want a pb of extensions.conf ? |
| 20:10.13 | [TK]D-Fender | swiftkick, When that GUI does your thinking for you is the problem. |
| 20:10.27 | [TK]D-Fender | swiftkick, And all related configs. voicemail.conf, users.conf, etc. |
| 20:10.41 | [TK]D-Fender | swiftkick, AND of course the CLI output of a failed attempt. |
| 20:10.54 | wundaboy | does anyone have any provider recommendations for origination/termination? |
| 20:10.57 | swiftkick | [TK]D-Fender: well, it isn't exactly doing my *thinking*. It was used to setup the system, by another party, who is no longer available to repair it. I have enough experience getting under the hood with config files to at least poke around and try and diagnose. |
| 20:11.13 | swiftkick | [TK]D-Fender 'k thank you one moment preparing paste |
| 20:12.43 | *** join/#asterisk RipeR-81 (n=ircap8@190.53.33.3) |
| 20:12.47 | dug | I am getting the following error http://pastebin.com/m4ca2880e, the first time I call the IVR works fine, the second time the system picks up and sounds like a fax? |
| 20:14.43 | *** join/#asterisk ptblank (n=MURDER1@cpe-75-84-218-175.socal.res.rr.com) |
| 20:14.44 | [TK]D-Fender | dug, PB failure |
| 20:14.59 | dug | PB failure? |
| 20:15.28 | [TK]D-Fender | sorry, it took the "," at the end. |
| 20:15.31 | *** join/#asterisk shido6 (n=shido6@74-130-227-15.dhcp.insightbb.com) |
| 20:15.50 | [TK]D-Fender | dug, and please provide REAL CLI output, not jsut that cryptic debug. |
| 20:17.18 | dug | fender a bit new to the cli, how would I do that? |
| 20:17.50 | dug | [TK]D-Fender: new to asterisk in general |
| 20:17.51 | [TK]D-Fender | go to CLI, do "set verbose 10", watch the call. pastbin it |
| 20:18.15 | swiftkick | [TK]D-Fender: http://pastebin.com/d7f237d3a |
| 20:18.23 | swiftkick | is the extensions.conf file... |
| 20:18.32 | swiftkick | one sec for some CLI output |
| 20:20.11 | *** join/#asterisk jdg (n=jdg@203.185.180.50) |
| 20:20.39 | *** join/#asterisk JungleRob (n=jungle@user-24-214-39-183.knology.net) |
| 20:20.40 | dug | [TK]D-Fender: http://pastebin.com/m66183668 <- pickup but no IVR |
| 20:21.57 | [TK]D-Fender | dug, Sure looks lie its waiting for input, and it TIMES OUT pretty blatantly |
| 20:21.58 | *** join/#asterisk Strom_M (n=strom@m7b0f36d0.tmodns.net) |
| 20:22.06 | *** part/#asterisk JungleRob (n=jungle@user-24-214-39-183.knology.net) |
| 20:22.14 | [TK]D-Fender | dug, -- Timeout on Zap/3-1, going to 't' |
| 20:23.46 | dug | [TK]D-Fender: I dont enter anything so that would be correct, but it doesnt play mainmenu as it says it is.. or is it timing out prior to playing? |
| 20:23.49 | swiftkick | [TK]D-Fender here is the pastebin of the CLI failure: http://pastebin.com/m44f863cc |
| 20:24.18 | [TK]D-Fender | dug, -- Executing [s@ivr-2:10] BackGround("Zap/3-1", "custom/mainmenu") in new stack |
| 20:24.32 | [TK]D-Fender | dug, says its playing a recording. Doesn't mean it CONTAINS anything useful. |
| 20:24.59 | [TK]D-Fender | swiftkick, Wheres the Voicemail failure? |
| 20:25.05 | *** join/#asterisk bkruse_home (n=root@69.73.127.92) |
| 20:25.05 | *** join/#asterisk Lucky7 (n=Adam@207.200.28.175) |
| 20:25.09 | dug | [TK]D-Fender: but I just called again an it played the ivr correctly |
| 20:25.17 | swiftkick | <PROTECTED> |
| 20:25.32 | [TK]D-Fender | swiftkick, What does that have to do with anything? |
| 20:25.45 | swiftkick | well, |
| 20:25.54 | Lucky7 | [TK]D-Fender : you use Polycom IP330's right? |
| 20:26.08 | [TK]D-Fender | Lucky7, No, but I've provisioned them. |
| 20:26.50 | Lucky7 | I've done the same thing, I'm having a bit of a problem though, with keeping them registered.... IE: I have 15 phones that all connect to a sales ring-all |
| 20:27.16 | Lucky7 | only about 1/2 of those phones ring at any given time, and if you call the extensions that dont ring, you go directly to voicemail |
| 20:27.22 | Lucky7 | like the phone has been disconnected. |
| 20:27.26 | swiftkick | [TK]D-Fender: the ringroup is set up in extensions.conf like this:http://pastebin.com/m476255f5 |
| 20:27.32 | swiftkick | http://pastebin.com/m476255f5 |
| 20:27.34 | [TK]D-Fender | Lucky7, pardon? What exactly is this "thing" they are "registering" to? Your terminology is dangerously vague |
| 20:27.42 | Lucky7 | Asterisk |
| 20:27.52 | Lucky7 | Think > Providisoned phones |
| 20:27.54 | Lucky7 | thing*( |
| 20:28.00 | Lucky7 | .... crap, stupid fingers |
| 20:28.05 | Lucky7 | ok, lemme start over. |
| 20:28.25 | Lucky7 | I have 25 Polycom IP330's, All Provisioned from a centeralized FTP boot server. |
| 20:28.56 | Lucky7 | 15 of these phones, are on a Group-ringer, (i dial in, hit 2, and all the phones are supposed to ring) |
| 20:28.57 | [TK]D-Fender | swiftkick, do "dialplan show ringroups-custom-3" |
| 20:28.58 | mmdk | how does one debug dialing plans ? |
| 20:29.18 | [TK]D-Fender | mmdk, Another wonderfully vague question.... |
| 20:30.30 | swiftkick | [TK]D-Fender, the caller doesn't hang up at that point |
| 20:30.40 | jdg | mmdk, just watch how the calls are handled in the asterisk console |
| 20:30.43 | swiftkick | [TK]D-Fender, the caller gets a message saying "the mailbox you are trying to reach does not exist" |
| 20:31.00 | swiftkick | [TK]D-Fender, and whats odd is, when the VM works, it shows all messages being played on the CLI |
| 20:31.10 | swiftkick | [TK]D-Fender, in this case it does not. |
| 20:31.28 | [TK]D-Fender | swiftkick, No they don't.... where do you see * doing ANYTHING? |
| 20:31.35 | [TK]D-Fender | swiftkick, Where is the call comgin FROM? |
| 20:31.40 | swiftkick | PSTN |
| 20:32.05 | mmdk | jdg, tried watching it and it gets to a 404 |
| 20:32.07 | [TK]D-Fender | swiftkick, SPECIFIC. |
| 20:32.31 | [TK]D-Fender | swiftkick, because clearly it isn't ASTERISK playing that message you're getting, its your TELCO <------------ |
| 20:32.48 | swiftkick | AHA |
| 20:32.54 | swiftkick | i was thinking this was possible |
| 20:32.54 | [TK]D-Fender | swiftkick, probably because the call isn't "answered" and THEY are timing out. |
| 20:33.04 | swiftkick | hmmmmmmmmmmmmmmmmmmmmmmmmmmm |
| 20:33.34 | swiftkick | [TK]D-Fender : except the ringgroup does work |
| 20:33.50 | swiftkick | [TK]D-Fender : it rings all the phones for the entire period up to the timeout specified in the dialplan |
| 20:34.13 | swiftkick | [TK]D-Fender : i have this problem with 2 of 4 ringroups, with different timeouts |
| 20:34.48 | [TK]D-Fender | swiftkick, Clearly * is not playing these messages. There is nothing more to say. |
| 20:35.41 | swiftkick | [TK]D-Fender sure, but, how is * dropping the call after the timeout period specified in the ringgroup? whereas two identically coded ringroups work 100% correctly |
| 20:36.10 | [TK]D-Fender | swiftkick, I guess you'd have to SHOW these 2 different samples. |
| 20:36.24 | [TK]D-Fender | swiftkick, and if everything were perfect it wold WORK. |
| 20:37.21 | shido6 | mic check |
| 20:38.12 | swiftkick | [TK]D-Fender: http://pastebin.com/d7f237d3a note the context [default] ... look at how the extension is handled at the line exten = 7738716638 , both under [default] and under [DID_bp] |
| 20:39.03 | [TK]D-Fender | swiftkick, No, show me the CLI PROVING whats getting EXECUTED. * doesn't care what your dialplan COULD be doing, it cares about what it IS doing. |
| 20:39.27 | [TK]D-Fender | swiftkick, I could code 100 contexts for WORLD DOMINATION, and if they never get executed, who cares? |
| 20:40.17 | swiftkick | [TK]D-Fender: [default] gets executed no matter what yes? i am asking a different question here: can you explain why someone would code a line like : exten = 7738716638,1,Goto(ringroups-custom-2,s,1) |
| 20:40.46 | swiftkick | where 7738716638 is the number being dialed INTO * from the PSTN |
| 20:40.50 | [TK]D-Fender | swiftkick, no, it DOESN'T. * doesn't care if you call a context [default] or [FRED] |
| 20:41.04 | [TK]D-Fender | swiftkick, [default] has NO SPECIAL MEANING |
| 20:44.04 | swiftkick | [TK]D-Fender: here is a paste of a (neighboring) ringroup working correctly to ring 2x extensions then drop thru to the VM |
| 20:44.05 | swiftkick | http://pastebin.com/d665d25f7 |
| 20:44.08 | swiftkick | from the CLI |
| 20:44.10 | [TK]D-Fender | swiftkick, calls coming in via pretty much anything except an ANALOG LINE, typically target a specific extension. When you register with an ITSP and they send a call to you, they usually send the DID you have with them that was DIALED. If that call is designed to land in on that context and it matches that exten, the call would then begin processing, it woul jump to that other context and begin doing whatever is in there. |
| 20:45.21 | [TK]D-Fender | swiftkick, Executing [s@ringroups-custom-2:2] Dial("SIP/bwas1-vir.atl0.cbeyond.net-08b82448", "SIP/31&SIP/32|30") in new stack |
| 20:45.28 | swiftkick | the problem is that the extension seems to be specified as the DID that was Dialed, coming in on the same trunk. (DID Trunk 3.) This appears to be a hack/kludge/screwup on the part of the ITSP |
| 20:45.46 | [TK]D-Fender | swiftkick, and you showed me [ringroups-custom-3] |
| 20:45.51 | swiftkick | yes |
| 20:45.52 | [TK]D-Fender | swiftkick, these are NOT THE SAME |
| 20:46.03 | swiftkick | i was saying: ringroups-custom-2 *WORKS* |
| 20:46.09 | swiftkick | i was saying: ringroups-custom-3 doesn't. |
| 20:46.36 | [TK]D-Fender | swiftkick, then blame the GUI for creating broken configs, or see where you may have screwed up some choices it offered you. this is not the ITSP's fault |
| 20:47.03 | swiftkick | again, in the original pb link i pasted of extensions.conf, http://pastebin.com/d7f237d3a it shows the exten = line based on a DID incoming telephone number in a given context |
| 20:47.04 | [TK]D-Fender | swiftkick, I highly advise you to ditch that GUI and make your own sane configs |
| 20:48.25 | swiftkick | [TK]D-Fender : the config is 4 providers total, less than 20 extensions total. its not really very hard to sort thru -- except for all the annoying "DEMO" and "DUNDI" contexts that asterisk-gui keeps wanting to put in extensions.conf |
| 20:48.56 | swiftkick | I'm trying to preserve some level of managability by GUI for othe end users not as familiar with ssh and vi as myself. :) :) :) |
| 20:50.54 | wundaboy | does anyone have any provider recommendations for origination/termination? |
| 20:51.04 | bkruse_home | If the gui does not work |
| 20:51.06 | wundaboy | i'm currently using Junction networks and it is ridiculously expensive $.029 |
| 20:51.08 | bkruse_home | submit a bug, and I will fix it |
| 20:51.24 | bkruse_home | [TK]D-Fender: I understand where you are coming from completely, how long have you been using asterisk? |
| 20:51.43 | *** join/#asterisk pepo-- (n=pepOSX@201.210.227.45) |
| 20:51.59 | bkruse_home | I couldnt do anything but config files, vim + terminal is it. However, the GUI is a great start for a user beginning into asterisk....even has a file editor if you want to get down and dirty :P |
| 20:52.41 | swiftkick | bkruse_home: well the current trunk of the asterisk-GUI has some problems with "service providers" - wants to show them all as IAXTEL even if they are set as custom-voip. i rolled back to an earlier version. havent gotten involved in * and *-gui community enough to feel comfortable enough to submit a bug report |
| 20:53.00 | bkruse_home | swiftkick: #asterisk-gui, i need feedback if I am going to fix it! :D so please just do not complain...but help :] |
| 20:53.26 | bkruse_home | swiftkick: You can never go wrong with submitting bug reports |
| 20:53.38 | bkruse_home | even if its "I have no idea how to fix it, but this doesnt work as it should...." |
| 20:53.43 | bkruse_home | what svn rev are you running? |
| 20:53.48 | bkruse_home | [TK]D-Fender: see where im coming from? |
| 20:54.09 | bkruse_home | Could one of the ops put #asterisk-gui in the title for gui questions, I miss so many of these conversations :[ |
| 20:55.22 | swiftkick | too many windows open - brain keeps trying to use "screen" shortcut ctrl-a D to switch irc windows... heh :) |
| 20:55.36 | *** join/#asterisk apardo (n=apardo@87.223.171.17) |
| 20:55.42 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 20:55.58 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 20:56.09 | jdg | it is already in the title :) |
| 20:56.14 | bkruse_home | jdg: it is? |
| 20:56.32 | *** join/#asterisk yannj_fr (n=yannj_fr@ALagny-152-1-85-227.w86-198.abo.wanadoo.fr) |
| 20:57.30 | jdg | bkruse_home: YES -=- Join #asterisknow or #asterisk-gui for AsteriskNOW and Asterisk-GUI info |
| 20:57.47 | bkruse_home | jdg: I guess its just no one reads the topic these days :P |
| 20:57.59 | bkruse_home | jdg: thanks. I keep wanting to type "gdb" when I am talking to you :P |
| 21:03.29 | swiftkick | [TK]D-Fender: really, the issue seems to be asterisk level, not asterisk-gui level, since i am editting most of these files by hand and just verifying that the GUI picks up the changes correctly. but bkruse has sent me in the right direction with bugs.digium.com, specifically http://bugs.digium.com/view.php?id=10151&nbn=10#bugnotes |
| 21:03.42 | swiftkick | [TK]D-Fender and bkruse_home I thank both of you for your assistance with this. |
| 21:03.51 | swiftkick | it is highly appreciated |
| 21:14.06 | RipeR-81 | hey everyone, anybody knows how to open ports on iptables ? |
| 21:15.49 | RipeR-81 | ? |
| 21:17.21 | mvanbaak | RipeR-81: iptables -I INPUT -d <your_ip> --dport <port> -j ACCEPT |
| 21:18.29 | RipeR-81 | thx |
| 21:23.15 | TmBerg | RipeR-81: Or, iptables -I INPUT -p tcp --dport 65533 -j ACCEPT |
| 21:36.49 | wundaboy | no one has suggestions for voip providers? |
| 21:42.04 | *** join/#asterisk tsurko (n=tsurko@78.90.100.214) |
| 21:44.10 | *** join/#asterisk tsurko (n=tsurko@78.90.100.214) |
| 21:49.51 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 21:51.47 | jdg | wundaboy: I'm happy with teliax |
| 21:57.45 | *** part/#asterisk mitcheloc (n=mitchel@adsl-68-120-230-95.dsl.irvnca.pacbell.net) |
| 22:03.29 | WilliamK | anyone have comments on howto fix the zaptel script? http://www.pastebin.ca/689196 |
| 22:07.44 | mvanbaak | wundaboy: depends on what part of the world |
| 22:09.52 | tzafrir_laptop | WilliamK, which version of zaptel is it? |
| 22:10.15 | tzafrir_laptop | And which distro? |
| 22:10.33 | WilliamK | latest svn |
| 22:10.36 | WilliamK | just updated |
| 22:10.42 | WilliamK | branches 1.4 |
| 22:11.41 | tzafrir_laptop | The "unknown line" lines are probably from the output of lsusb. Ignore them |
| 22:11.47 | WilliamK | I looked in the /etc/default/zaptel file and the modules appear to be correct |
| 22:12.09 | WilliamK | however, zaptel is failing to start still via init script |
| 22:12.10 | tzafrir_laptop | which distro is it? |
| 22:12.21 | WilliamK | CentOS 5 |
| 22:12.42 | tzafrir_laptop | in centos the init.d scripts looks at /etc/sysconfig/zaptel ... |
| 22:13.39 | tzafrir_laptop | grep ^MODULES= /etc/sysconfig/zaptel |
| 22:14.11 | WilliamK | still fails after I copied it to /etc/sysconfig/zaptel |
| 22:14.26 | WilliamK | Loading zaptel hardware modules:Running ztcfg: ZT_SPANCONFIG failed on span 1: Invalid argument (22) |
| 22:14.50 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 22:16.19 | WilliamK | by the way, when I did the initial install, it put it in /etc/default/zaptel by itself |
| 22:16.46 | WilliamK | and the config for zaptel.conf works if I manually do the modprobe |
| 22:18.51 | tzafrir_laptop | WilliamK, with svn? or with 1.4.4? (the copy to the wrong place) |
| 22:19.16 | tzafrir_laptop | do you have more than one card? or is ztdummy loaded? |
| 22:19.29 | WilliamK | 1 wcte12xp |
| 22:19.48 | tzafrir_laptop | what is the span line from zaptel.conf? |
| 22:19.55 | WilliamK | here lemme delete the file and I'll know for sure which one put it in the wrong place |
| 22:20.24 | WilliamK | span=1,1,1,d4,ami |
| 22:20.54 | tzafrir_laptop | take a look at /proc/zaptel/1 . Any chance that this is ztdummy ? |
| 22:22.17 | WilliamK | Span 1: ZTDUMMY/1 "ZTDUMMY/1 1" |
| 22:22.28 | *** join/#asterisk mitcheloc (n=mitchel@adsl-68-120-230-95.dsl.irvnca.pacbell.net) |
| 22:23.15 | WilliamK | re-installing zaptel, I deleted the zaptel file to find out which ver installed it in the wrong place |
| 22:25.15 | WilliamK | had to be 1.4.4 that was wrong |
| 22:25.19 | WilliamK | svn is correct |
| 22:25.32 | WilliamK | also appears I got alot farther after deleting the old file |
| 22:25.39 | WilliamK | however, still failed |
| 22:26.57 | WilliamK | okie, I see part of the problem |
| 22:27.08 | WilliamK | svn doesn't appear to have the module for the wcte12xp |
| 22:27.15 | WilliamK | at least in the zaptel file |
| 22:28.32 | CCFL_Man2 | i won an avocado colored western electric 202 |
| 22:28.37 | CCFL_Man2 | on ebay |
| 22:30.56 | WilliamK | guess that's 1 problem, not sure where else wcte12xp has been forgotten from |
| 22:33.38 | *** join/#asterisk craigk (n=craigk@58.174.114.25) |
| 22:37.06 | CCFL_Man2 | http://cgi.ebay.com/Vtg-Bell-Western-Electric-Bakelite-Desk-Telephone-PINK_W0QQitemZ120159439049QQihZ002QQcategoryZ38037QQssPageNameZWDVWQQrdZ1QQcmdZViewItem |
| 22:37.17 | CCFL_Man2 | a pink 302! |
| 22:37.27 | CCFL_Man2 | how farkin rare is that |
| 22:41.10 | WilliamK | this is fun (not really) but might as well say it is |
| 22:41.27 | WilliamK | so far it's been left out of zaptel.init |
| 22:41.28 | tzafrir_laptop | WilliamK, so you loaded without anything defined. The init.d script decided to load ztdummy, because "you have no hardware" |
| 22:42.02 | tzafrir_laptop | Now you modprobe your module, and it got span no. 2, even though it is configured for span no. 1 |
| 22:42.20 | WilliamK | and the scary thing is, it worked |
| 22:42.24 | tzafrir_laptop | Try: /etc/init.d/zaptel restart |
| 22:42.35 | WilliamK | already tried |
| 22:43.03 | WilliamK | Running ztcfg: ZT_SPANCONFIG failed on span 1: Invalid argument (22) |
| 22:43.05 | tzafrir_laptop | That should work if you have your module in MODULES |
| 22:43.18 | JT | CCFL_Man2: europe is not a country. |
| 22:43.55 | WilliamK | I added it manually so I know it's there |
| 22:43.59 | CCFL_Man2 | JT: pardon |
| 22:45.00 | WilliamK | any other possible places it could be missing from? I tried doing a grep looking for lines for wcte11xp that didn't match what I saw for wcte12xp |
| 22:45.57 | JT | < CCFL_Man2> g711alaw is used in broken countries like europe :P |
| 22:46.18 | JT | ^broken statement |
| 22:47.53 | CCFL_Man2 | ok, it's a union |
| 22:48.06 | tzafrir_laptop | WilliamK, modinfo wcte12xp |
| 22:48.17 | tzafrir_laptop | does it give any output? |
| 22:48.19 | JT | also a continent |
| 22:48.38 | JT | and plenty more countries than european ones use G.711a |
| 22:48.55 | JT | in fact, many more countries use it than that which use G.711Mu |
| 22:49.34 | CCFL_Man2 | JT: more countries use PAL, too |
| 22:49.54 | JT | correct |
| 22:49.58 | JT | because NTSC is rubbish |
| 22:50.04 | CCFL_Man2 | i pledge my loyalty to bell |
| 22:50.19 | JT | Never The Same Colour twice |
| 22:50.23 | CCFL_Man2 | technically, yes |
| 22:50.56 | CCFL_Man2 | but early color tubes produced great color |
| 22:51.03 | JT | tubes? |
| 22:51.31 | WilliamK | yep |
| 22:52.00 | JT | if you're talking about a CRT, who cares if the colour is great if it is wrong? |
| 22:52.02 | CCFL_Man2 | picture tubes |
| 22:52.07 | CCFL_Man2 | lol |
| 22:52.10 | JT | oh, a Cathode Ray Tube |
| 22:52.12 | JT | not a lol |
| 22:52.24 | WilliamK | tzafrir_laptop: do you need the info? |
| 22:52.31 | JT | this problem has nothing to do with the CRT |
| 22:53.03 | JT | it has to do with colour being dependant on the phase angle of the video signal |
| 22:53.08 | tzafrir_laptop | WilliamK, if there wasn't an erro: no |
| 22:53.12 | CCFL_Man2 | JT: right |
| 22:53.16 | JT | which is a complete disaster in an analogue transmission network |
| 22:53.22 | JT | ntsc is a disaster :) |
| 22:53.42 | CCFL_Man2 | JT: thats the reason for the luma delay line |
| 22:53.52 | WilliamK | not at error, all good info |
| 22:53.59 | WilliamK | err at = an |
| 22:54.15 | JT | CCFL_Man2: also the resolution is inferior to PAL |
| 22:54.48 | tzafrir_laptop | well, what do you have under /proc/zaptel ? |
| 22:55.06 | CCFL_Man2 | JT: oh it's a few lines :P |
| 22:55.16 | JT | CCFL_Man2: and a few colours... right |
| 22:55.55 | WilliamK | finally works |
| 22:56.22 | WilliamK | after fixing those files, I did a manual modprobe again, unloaded the modules, and restarted zaptel |
| 22:56.24 | CCFL_Man2 | JT: just a couple |
| 22:56.53 | WilliamK | guess zaptel.init (and/or the modules file just needs to be fixed to include the wcte12xp) |
| 22:56.54 | JT | CCFL_Man2: how many other technical innacuracies will i find in your scrollback? :P |
| 22:57.05 | CCFL_Man2 | JT: digital video and s-video solve that :P |
| 22:57.11 | CCFL_Man2 | JT: quite a bit |
| 22:57.28 | CCFL_Man2 | did you see that super sexy pink 302? |
| 22:57.32 | JT | like peer to peer rtp ;) |
| 22:57.37 | tzafrir_laptop | WilliamK, if you have it in MODULES , then it is already included |
| 22:57.46 | *** join/#asterisk kiscokid (n=Ron_Laut@adsl-75-12-78-154.dsl.pltn13.sbcglobal.net) |
| 22:58.25 | CCFL_Man2 | JT: i know :P |
| 22:59.04 | WilliamK | tzafrir, I'm saying the master copy on svn |
| 22:59.11 | WilliamK | I had to manually edit my files to include it |
| 22:59.15 | WilliamK | wasn't there |
| 23:00.10 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 23:04.18 | tzafrir_laptop | right. |
| 23:07.16 | *** join/#asterisk fujin (n=aj@unaffiliated/fujin) |
| 23:14.59 | *** join/#asterisk EclecticRob (n=Eclectic@24-176-222-123.static.lnbh.ca.charter.com) |
| 23:18.25 | WilliamK | gonna be a long night |
| 23:20.16 | *** join/#asterisk Carlos_Tico (n=grillo_v@c-76-30-253-15.hsd1.tx.comcast.net) |
| 23:20.26 | Carlos_Tico | Hi people |
| 23:20.27 | Carlos_Tico | how are you |
| 23:20.43 | Carlos_Tico | i need a little hand to set up my spa3102 with asterisk |
| 23:22.08 | Carlos_Tico | hi ? |
| 23:27.06 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 23:28.11 | GlobeTrotter | hola,, anyone knows what this means? |
| 23:28.14 | GlobeTrotter | chan_iax2.c: Received trunked frame before first full voice frame |
| 23:29.03 | Carlos_Tico | ummm |
| 23:32.04 | *** join/#asterisk Ebola (n=Ebola@host86-143-7-120.range86-143.btcentralplus.com) |
| 23:32.11 | *** join/#asterisk diemaco (n=diemaco@unaffiliated/diemaco) |
| 23:34.31 | WilliamK | tzafrir, it's back up, did the ls command, nothing |
| 23:34.39 | WilliamK | lsmod shows the modules are loaded though |
| 23:39.23 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 23:46.35 | tzafrir_laptop | WilliamK, nothing under /proc/zaptel ? |
| 23:49.03 | WilliamK | yeah 1 |
| 23:49.40 | tzafrir_laptop | what is it? your card or ztdummy? |
| 23:49.41 | WilliamK | it's loading the driver |
| 23:49.53 | WilliamK | Span 1: WCT1/0 "Wildcard TE12xP Card 0" |
| 23:50.11 | tzafrir_laptop | but ztcfg originally failed? |
| 23:50.18 | WilliamK | yep |
| 23:50.38 | tzafrir_laptop | what error? |
| 23:50.56 | WilliamK | how do I see the trace? |
| 23:51.19 | tzafrir_laptop | It was sent to the console |
| 23:52.29 | tzafrir_laptop | I figure that redirecting it to a file would be: exec 2>/tmp/log |
| 23:52.30 | WilliamK | when I reboot the box it fails, doesn't seem to be failing if I manually start it |
| 23:52.58 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 23:52.59 | tzafrir_laptop | if you just run ztcfg, does it succeed? |
| 23:53.41 | WilliamK | line 0: Unable to open master device '/dev/zap/ctl' |
| 23:54.28 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 23:54.28 | *** mode/#asterisk [+o mog] by ChanServ |
| 23:57.26 | tzafrir_laptop | This is immedietly after loading the module? |
| 23:58.02 | WilliamK | rebooted the box, by default it has zaptel and asterisk set to auto load, and this is the result once it booted |