| 00:04.07 | *** join/#asterisk ruben23 (n=AGENT@122.55.48.242) |
| 00:07.12 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 00:08.41 | *** join/#asterisk Superbartt (n=bart@213.10.33.201) |
| 00:09.38 | *** join/#asterisk securevoip (n=securevo@c-76-123-20-170.hsd1.va.comcast.net) |
| 00:17.51 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
| 00:21.43 | *** join/#asterisk Lucas_ (n=lucasb@office.telifon.com) |
| 00:26.17 | *** join/#asterisk JMAFOU (n=crypt@c-68-58-30-239.hsd1.in.comcast.net) |
| 00:30.33 | VaGoNeTaS | i need to use mp3 |
| 00:30.41 | VaGoNeTaS | on the music on hold |
| 00:30.43 | VaGoNeTaS | but is not working |
| 00:30.50 | VaGoNeTaS | asterisk 1.4.21.1 |
| 00:36.24 | ruben23 | is this process would help me to register a remote SIP form my asterisk behind NAT..? http://www.voip-info.org/wiki/view/port+forwarding |
| 00:37.46 | Superbartt | basically yes ruben23 |
| 00:38.20 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 00:39.36 | ruben23 | Superbartt: i just point my SIP softphones to the public IP of my Nat Box..then the box would do the forwarding to my local asterisk..>? |
| 00:41.05 | Superbartt | uhmm... could you make a little sketch of your setup? |
| 00:41.34 | Superbartt | by the way, the site has a better nat guide, covering about all situations |
| 00:41.55 | Superbartt | http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 00:42.39 | Superbartt | good luck, time for me to get some sleep |
| 00:42.44 | VaGoNeTaS | [May 6 20:44:37] WARNING[4669]: format_wav.c:148 check_header: Not in mono 2 |
| 00:42.44 | VaGoNeTaS | [May 6 20:44:37] WARNING[4669]: file.c:376 fn_wrapper: Unable to open format wav |
| 00:42.44 | VaGoNeTaS | [May 6 20:44:37] WARNING[4669]: res_musiconhold.c:250 ast_moh_files_next: Unable to open file '/var/lib/asterisk/moh/fpm-calm-river': No such file or directory |
| 00:42.45 | VaGoNeTaS | <PROTECTED> |
| 00:43.04 | VaGoNeTaS | why does it says that the file doesnt exists if it does? |
| 00:43.50 | VaGoNeTaS | what do i have to do with the wav file |
| 00:44.00 | VaGoNeTaS | need to convert it with some speacial program or some? |
| 00:44.11 | VaGoNeTaS | does it have to be like in a mono way or some? |
| 00:44.18 | [TK]D-Fender | ruben23: for the 500th time : |
| 00:44.20 | [TK]D-Fender | ~sipnat |
| 00:44.20 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 00:44.21 | [TK]D-Fender | ^^^^^^^^^^^^^ |
| 00:44.48 | VaGoNeTaS | TK do you know something about it? |
| 00:44.57 | [TK]D-Fender | VaGoNeTaS: If you need to use MP3, then why are you messing with WAV's? |
| 00:44.59 | ruben23 | Superbartt: do you have email..? |
| 00:45.05 | ruben23 | ill email it to you.. |
| 00:45.10 | VaGoNeTaS | coz mp3 didnt worked |
| 00:45.19 | VaGoNeTaS | so im trying with a wav file |
| 00:45.33 | [TK]D-Fender | VaGoNeTaS: Do you hve format_mp3.so? |
| 00:45.45 | VaGoNeTaS | i dont think so |
| 00:45.59 | [TK]D-Fender | VaGoNeTaS: Did you install asterisk-addons? |
| 00:46.08 | VaGoNeTaS | no didnt |
| 00:46.18 | VaGoNeTaS | and i cant affort right now |
| 00:46.22 | [TK]D-Fender | VaGoNeTaS: Guess what... Asterisk does not support MP3 without it <- |
| 00:46.23 | VaGoNeTaS | so im trying with the wav |
| 00:46.44 | [TK]D-Fender | VaGoNeTaS: And what do you mean "can't afford"? |
| 00:47.11 | VaGoNeTaS | well, i can understand that * doesnt support mp3 without it |
| 00:47.41 | VaGoNeTaS | i dont wanna take the risk recompiling the * again |
| 00:47.48 | VaGoNeTaS | but at this time |
| 00:48.00 | VaGoNeTaS | it DOES support WAV file |
| 00:48.27 | VaGoNeTaS | the thing is... |
| 00:48.48 | VaGoNeTaS | i belive that it has to be in some kind of compressing way |
| 00:48.50 | [TK]D-Fender | VaGoNeTaS: You don't need to recompile * |
| 00:48.51 | VaGoNeTaS | like 64 bits or some |
| 00:48.55 | VaGoNeTaS | no? |
| 00:49.02 | VaGoNeTaS | so what do i have to do to add the * addons? |
| 00:49.09 | [TK]D-Fender | VaGoNeTaS: *'s supported formats are well documented on the WIKI. |
| 00:49.23 | [TK]D-Fender | VaGoNeTaS: download the tarball, and foow the instructions. |
| 00:49.26 | [TK]D-Fender | follow* |
| 00:49.32 | VaGoNeTaS | what tarball |
| 00:49.36 | VaGoNeTaS | *'s addons? |
| 00:50.43 | [TK]D-Fender | VaGoNeTaS: www.asterisk.org |
| 00:53.02 | VaGoNeTaS | ok |
| 00:53.04 | VaGoNeTaS | and |
| 00:53.18 | VaGoNeTaS | what about the wav files that * support already? |
| 00:53.28 | VaGoNeTaS | is 64 bits or some? |
| 00:57.19 | *** join/#asterisk werdan7 (n=w7@freenode/staff/wikimedia.werdan7) |
| 00:58.08 | *** join/#asterisk riddlebox (n=user@75-132-225-75.dhcp.stls.mo.charter.com) |
| 00:58.56 | *** part/#asterisk ruben23 (n=AGENT@122.55.48.242) |
| 01:00.08 | [TK]D-Fender | VaGoNeTaS: ? |
| 01:00.29 | [TK]D-Fender | VaGoNeTaS: 16 bit 8 khz, mono |
| 01:03.02 | *** join/#asterisk saftsack (n=saftsack@p579246D9.dip.t-dialin.net) |
| 01:03.49 | saftsack | hi, i have choppy sound in app_meetme (by hearing the file you are the only person) what could be the reason? dahdi_dummy is loaded without any errors. |
| 01:03.54 | VaGoNeTaS | <PROTECTED> |
| 01:03.54 | VaGoNeTaS | <PROTECTED> |
| 01:03.57 | VaGoNeTaS | TK |
| 01:04.04 | *** join/#asterisk VoipForces (n=kvirc@mail.net-forces.com) |
| 01:04.05 | VaGoNeTaS | that's on the make menuconfig |
| 01:04.09 | VaGoNeTaS | of * addons |
| 01:04.21 | VaGoNeTaS | i dont see anything related to format_mp3 |
| 01:04.46 | [TK]D-Fender | VaGoNeTaS: JUST DO IT |
| 01:04.52 | VoipForces | Quick question for chan_dahdi, can I have channel groups that overlaps channels. i.e. group=1 has channels 1,2,3 and group=2 has channels 2.3.4 |
| 01:05.27 | jaytee | VoipForces, nope |
| 01:05.30 | [TK]D-Fender | VoipForces: yes |
| 01:05.35 | jaytee | nope |
| 01:05.37 | [TK]D-Fender | YES |
| 01:05.41 | jaytee | NO |
| 01:05.43 | VoipForces | LOL I like that, fight it off |
| 01:06.00 | saftsack | cpu load is low, kernel is 2.6.25. no errors from *. any ideas for the choppy sound? |
| 01:06.30 | VoipForces | saftsack: voip, analog telephony, digital telephony? |
| 01:06.40 | saftsack | sip |
| 01:06.50 | VoipForces | jaytee: you say no by experience? |
| 01:07.00 | [TK]D-Fender | VoipForces: http://www.voip-info.org/wiki/view/Asterisk+config+zapata.conf |
| 01:07.02 | jaytee | no, I just say no |
| 01:07.02 | saftsack | 100mbit lan connection to the server. it is in the same room |
| 01:07.09 | [TK]D-Fender | VoipForces: "group: Allows you to group together a number of channels so that the Dial command will treat the group as a single channel. When Dial tries to make a call on a Zap group, the Zap channel module will use the first available (i.e. non-busy) channel in the group for the call. Multiple group memberships may be specified with commas, and to signify no group membership, the portion... |
| 01:07.11 | [TK]D-Fender | ...after the equals sign may be omitted. Group numbers can range from 0 to 31. The default value is an empty string, i.e. no groups." |
| 01:07.19 | *** join/#asterisk Kumbang (n=unknown@125.163.83.153) |
| 01:07.24 | saftsack | everything else works. just the meetme app is broken |
| 01:07.26 | [TK]D-Fender | jaytee: Only because Nancy Reagan told you to! |
| 01:08.00 | jaytee | [TK]D-Fender, I've never seen an example of overlapping channels in groups before in zapata.conf |
| 01:08.05 | [TK]D-Fender | adds some conditioner to jaytee's brain-washing |
| 01:08.11 | VoipForces | [TK]D-Fender: By that definition, i would be tempted to say that if channel 2 in group 1 is busy, group 2 would not know about it. |
| 01:08.19 | [TK]D-Fender | jaytee: Just look at that shine! |
| 01:08.53 | [TK]D-Fender | VoipForces: If the channel is busy, then the channel is busy |
| 01:09.11 | [TK]D-Fender | VoipForces: Stop with the crazy-talk |
| 01:09.42 | [TK]D-Fender | VoipForces: Groups dont know anything other than the list of channels that are part of it. * checks them sequentially just the same. |
| 01:09.59 | VoipForces | [TK]D-Fender: I know it's crazy, the customer wants it that way... tried to fight it off, but it's a freaking applition8 pages long of visio algorythms |
| 01:10.57 | [TK]D-Fender | VoipForces: No, I'm saying your claim about a channel in 1 group not being known as busy in anothergroup 1 |
| 01:11.45 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 01:11.58 | VoipForces | [TK]D-Fender: Well, that was just a worry that if L2 of G2 was busy asterisk would attempt to use it anyway when trying to dial on G1 |
| 01:12.27 | VaGoNeTaS | yes i say it on format intepreters |
| 01:12.29 | [TK]D-Fender | VoipForces: If the channel is busy, then the channel is busy <- |
| 01:12.33 | VaGoNeTaS | u dont have to scream xD |
| 01:12.38 | VaGoNeTaS | hehehe |
| 01:12.54 | VaGoNeTaS | well, after that, it'll be available or i have to restart asterisk? |
| 01:12.56 | VaGoNeTaS | i mean |
| 01:13.03 | VaGoNeTaS | i have to ' module load format_mp3.so ' |
| 01:13.08 | VaGoNeTaS | or restart asterisk 1st? |
| 01:13.08 | VoipForces | [TK]D-Fender: Okok, I get it, I'll set it up and if it breaks I'll come back to hunt you |
| 01:13.13 | [TK]D-Fender | VaGoNeTaS: You can just manually load it if the system is live |
| 01:13.36 | VaGoNeTaS | reportes*CLI> module load format_mp3.so |
| 01:13.36 | VaGoNeTaS | <PROTECTED> |
| 01:13.36 | VaGoNeTaS | <PROTECTED> |
| 01:13.39 | VaGoNeTaS | yeah |
| 01:13.43 | VaGoNeTaS | now what |
| 01:13.47 | VaGoNeTaS | i have to add |
| 01:13.53 | [TK]D-Fender | VoipForces: You're local, and I'm well trained and armed. :) |
| 01:13.59 | VaGoNeTaS | something to the musiconhold.conf ? |
| 01:14.05 | [TK]D-Fender | VaGoNeTaS: Now * supports MP3. |
| 01:14.10 | VaGoNeTaS | yep |
| 01:14.10 | [TK]D-Fender | VaGoNeTaS: Have fun |
| 01:14.11 | VaGoNeTaS | but |
| 01:14.14 | VaGoNeTaS | hahahaha |
| 01:14.16 | VaGoNeTaS | xd |
| 01:15.09 | VoipForces | [TK]D-Fender: I'l bring my 9 years old, which is a blue belt LOL |
| 01:15.26 | VaGoNeTaS | IT WORKED! |
| 01:15.35 | VaGoNeTaS | thank you so mux buddy |
| 01:15.41 | VaGoNeTaS | much* |
| 01:15.44 | VaGoNeTaS | i've just call myself |
| 01:15.45 | [TK]D-Fender | VoipForces: Blue belt in what? |
| 01:15.47 | VaGoNeTaS | put me on hold |
| 01:15.53 | VaGoNeTaS | and mp3 just played |
| 01:15.53 | VaGoNeTaS | xD |
| 01:15.54 | saftsack | all in all: kernel 2.6.26 (sorry about the 25 above), dahdi_dummy loaded without errors, one sip phone in lan, low cpu consumption, telephony and everything works fine. just when i hear the message "you are the only person" at the sart of chan_meetme the sound is very very choppy! |
| 01:16.13 | [TK]D-Fender | saftsack: what about other sound files? |
| 01:16.23 | VoipForces | [TK]D-Fender: Karate, |
| 01:16.37 | saftsack | i will test now. but from this file the alaw and the gsm version had the same issues |
| 01:16.49 | [TK]D-Fender | VoipForces: LOL... *peasant arts* |
| 01:16.54 | [TK]D-Fender | VoipForces: http://en.wikipedia.org/wiki/Tenshin_Shoden_Katori_Shinto-ryu |
| 01:18.04 | VoipForces | [TK]D-Fender: LOL. He just did his first Tameshiwari, was something to see. A guy (adult) did 3 ceent slabs with this foot. |
| 01:18.46 | [TK]D-Fender | VoipForces: Yes.... brick & board breaking is a nifty "trick". |
| 01:20.07 | *** join/#asterisk chendy (n=chatzill@58.251.100.254) |
| 01:20.12 | [TK]D-Fender | VoipForces: Rest assured its applicability to anything else is worthless. Oh... and let ME bring *1* cement block, you can bring his bandages :) |
| 01:21.16 | VoipForces | [TK]D-Fender: If the channels were previously defined in an other group, do you think I need te specify the switch type again when I use it in an other group? |
| 01:21.29 | saftsack | [TK]D-Fender: normal sound files also doesnt work |
| 01:21.30 | [TK]D-Fender | VoipForces: But rest assured that if your child is ever attacked by an angry mob of bricks, they'll never see it coming ;) |
| 01:21.36 | [TK]D-Fender | saftsack: ... |
| 01:21.38 | [TK]D-Fender | ~gsmbug |
| 01:21.39 | infobot | [~gsmbug] there is a bug compiling Asterisk with GCC 4.2 where optimization errors cause GSM transcoding to distort heavily. Compile with GCC 4.1 or lower, or follow this patch : http://bugs.digium.com/view.php?id=11243 Also please read : http://forums.digium.com/viewtopic.php?p=116731&sid=3c65e49ec840380ed20f1c8426382b39 |
| 01:21.40 | [TK]D-Fender | ^^^^^ |
| 01:22.22 | saftsack | gcc (Debian 4.3.3-5) 4.3.3 also for this version? ;) |
| 01:22.24 | [TK]D-Fender | VoipForces: Settings carry between channel definitions unless overridden |
| 01:22.34 | VoipForces | [TK]D-Fender: LOL, well, he likes it and gives him assurance, beside not a lot of angry mobs of angry bricks in St-Remi, Quebec LOL |
| 01:22.47 | VoipForces | [TK]D-Fender: True, forgot about it |
| 01:23.26 | [TK]D-Fender | VoipForces: Out of curiosity, do you know the name of his branch or Karate, and his Sensei? |
| 01:24.10 | VoipForces | [TK]D-Fender: Sensei Claudine Verville Black 7eme Dan, Ate Waza Kan |
| 01:26.02 | [TK]D-Fender | VoipForces: I should adjust myself then : Bastard offshoot of a peasent art :) |
| 01:26.21 | [TK]D-Fender | VoipForces: At least it isn't Kyukushin :) |
| 01:26.46 | [TK]D-Fender | has no time for "do" |
| 01:27.17 | VoipForces | [TK]D-Fender: Well, I know shit about it other then watching him so. |
| 01:27.28 | [TK]D-Fender | finishes his exam for 3rd Kyu tomorrow |
| 01:27.44 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-243298fe55d7dcf5) |
| 01:28.28 | [TK]D-Fender | VoipForces: Hopefully they follow the primary teachings of some other school |
| 01:28.34 | saftsack | [TK]D-Fender: -- <SIP/patton-0a0c3de8> Playing 'hello-world.alaw' (language 'en') same error with this file. so this can't be a gsm bug, or? |
| 01:29.00 | [TK]D-Fender | saftsack: is * transcoding? |
| 01:29.21 | saftsack | no! there is just alaw allowed on this channel |
| 01:29.33 | [TK]D-Fender | saftsack: and you have that file in .alaw? |
| 01:29.39 | KyleK | can someone recommend an iax softphone for windows? |
| 01:29.40 | *** join/#asterisk lowlevel (n=Stuart@lowlevel.ca) |
| 01:29.47 | [TK]D-Fender | saftsack: Test with a softphone to rule out a problem with your gateway |
| 01:29.48 | saftsack | -- <SIP/patton-0a0c3de8> Playing 'hello-world.alaw' (language 'en') |
| 01:29.58 | [TK]D-Fender | KyleK: zoiper |
| 01:30.00 | [TK]D-Fender | ~zoiper |
| 01:30.01 | infobot | [~zoiper] Zoiper (Formerly known as Idefisk) is a free SIP and IAX soft-phone for Windows, MacOSX, and Linux that can be found at http://www.zoiper.com |
| 01:30.09 | KyleK | thx |
| 01:30.21 | saftsack | [TK]D-Fender: i had this issue with a phone in the office too. now im out so i have to check it over the gateway. |
| 01:30.29 | VoipForces | Talking of zoiper, anyone implemented the new web version? |
| 01:30.51 | [TK]D-Fender | saftsack: I mean NOT the Patton |
| 01:31.00 | [TK]D-Fender | saftsack: Remote SIP is fine |
| 01:31.08 | [TK]D-Fender | saftsack: just prove it isn't the PATTON. |
| 01:31.20 | [TK]D-Fender | saftsack: Next, are you running a VM on that box? |
| 01:31.35 | saftsack | yes i know. but directly with a sip phone in the same room, same problem without the patton |
| 01:31.46 | saftsack | no vm on this box. |
| 01:31.52 | KyleK | so with iax i wouldn't have to worry about nat, as long as I can reach my *? |
| 01:32.05 | saftsack | but it is a precompiled debian kernel, may they did some shit with the timer while compiling the kernel? |
| 01:32.38 | saftsack | but on the other hand for playback dummy isnt used ... |
| 01:33.08 | [TK]D-Fender | saftsack: Debian often uses 250hz timers which screw stuff up.. others like tzafrir_laptop may be able to better advise you on this. |
| 01:33.37 | [TK]D-Fender | KyleK: Why is it again that you haven't been able to follow the guides you've been given? |
| 01:34.06 | KyleK | what guides? the nat one? |
| 01:34.08 | [TK]D-Fender | saftsack: Run a zttest and check your timer accuracy |
| 01:34.12 | [TK]D-Fender | KyleK: Yes |
| 01:34.28 | saftsack | [TK]D-Fender: thx ;) i will try a selfcompiled kernel now. will bring the infos if it works in a few hours. compiling on an alix is :/ ^^ |
| 01:34.55 | [TK]D-Fender | VoipForces: http://idefix.www5.50megs.com/atew.htm |
| 01:35.01 | [TK]D-Fender | VoipForces: TABARNAC! |
| 01:35.48 | [TK]D-Fender | VoipForces: its a DERIVITIVE of Kyukushin! Then they make a reference to a name with "samurai" in it. HERESY |
| 01:37.25 | VoipForces | [TK]D-Fender: Yup that's them |
| 01:37.27 | KyleK | I did follow it, I forwarded ports 10000 to 20000 on my crappy home router for rtp and added a nat=yes and canreinvite=nonat |
| 01:37.50 | [TK]D-Fender | KyleK: Pastebin it all up... |
| 01:38.37 | [TK]D-Fender | KyleK: and http://www.imagebin.ca your route forwarding screen-shot |
| 01:38.58 | KyleK | not sure what the point is, that stuff works |
| 01:39.14 | [TK]D-Fender | KyleK: Oh? Then why soil yourself with IAX? |
| 01:39.55 | KyleK | I'm at a friends house right now behind a router, and I dont have access/permission to do any routing, so I figured that IAX would make sense |
| 01:40.12 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-72-128-66-243.wi.res.rr.com) |
| 01:42.20 | [TK]D-Fender | KyleK: So you want to connect to your home * server? |
| 01:42.20 | VoipForces | [TK]D-Fender: Do you think thatdahdi group change/addition need a comolete asterisk restart? Doing a simple reload of chan_dahdi does not seem to work... |
| 01:42.20 | KyleK | yup |
| 01:42.20 | [TK]D-Fender | KyleK: Clients don't need forwarding, only * |
| 01:42.20 | [TK]D-Fender | KyleK: Which you'd has seen if you read the guide... |
| 01:42.20 | KyleK | ohh, i read that * doesn't support stun, and I assumed stun was for both ways... |
| 01:42.33 | [TK]D-Fender | KyleK: It does support it, and doesn't need it. |
| 01:42.54 | [TK]D-Fender | VoipForces: feel free to show me. |
| 01:43.22 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-72-128-66-243.wi.res.rr.com) |
| 01:43.38 | VoipForces | [TK]D-Fender: The chan_dandh file? |
| 01:43.52 | VoipForces | VoipForces: chan_dahdi.conf file? |
| 01:43.59 | [TK]D-Fender | VoipForces: Your configs, your failed call attempt, etc |
| 01:45.29 | VoipForces | http://pastebin.com/m51dfae93 |
| 01:45.48 | VoipForces | [TK]D-Fender: http://pastebin.com/m51dfae93 |
| 01:46.47 | [TK]D-Fender | VoipForces: Schmuck, you are definig a channel MULTIPLE TIMES. |
| 01:46.56 | [TK]D-Fender | grabs his katana... |
| 01:47.17 | VoipForces | [TK]D-Fender: That's what I was saying at the beninningLOL |
| 01:47.38 | *** join/#asterisk Deeewayne (n=dwayne@c-76-29-245-9.hsd1.al.comcast.net) |
| 01:47.38 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 01:47.45 | [TK]D-Fender | VoipForces: A channel can belong to multiple groups buy your breakup is horribly wrong :) |
| 01:47.45 | VoipForces | VoipForces: My original question: "Quick question for chan_dahdi, can I have channel groups that overlaps channels. i.e. group=1 has channels 1,2,3 and group=2 has channels 2.3.4" |
| 01:48.15 | VoipForces | [TK]D-Fender: ok, explain... |
| 01:48.40 | [TK]D-Fender | VoipForces: Who is supposed to be in group 1? |
| 01:48.51 | [TK]D-Fender | VoipForces: you did 25 twice |
| 01:49.23 | VoipForces | g1 and g2 are used by an inbount application/script |
| 01:49.39 | VoipForces | g3 and g4 are use by an outbound script. |
| 01:49.44 | [TK]D-Fender | VoipForces: .... |
| 01:49.53 | [TK]D-Fender | 2 x 25 in group 1 |
| 01:49.54 | VoipForces | [TK]D-Fender: I know it's hell... |
| 01:50.03 | [TK]D-Fender | how can it be there TWICE? |
| 01:50.05 | VoipForces | [TK]D-Fender: Ah fuc^&* |
| 01:50.12 | [TK]D-Fender | VoipForces: typo'd for what? |
| 01:50.16 | VoipForces | [TK]D-Fender: copy/paste error |
| 01:50.35 | VoipForces | [TK]D-Fender: Still I should be able to address g4 right? |
| 01:50.36 | [TK]D-Fender | VoipForces: The buffer doesn't corrup in midstream you know.. |
| 01:50.51 | [TK]D-Fender | VoipForces: No... let me fix this mess up |
| 01:50.54 | VoipForces | [TK]D-Fender: :-P |
| 01:54.10 | *** join/#asterisk ingenius (n=alektro@host90.190-230-73.telecom.net.ar) |
| 01:56.06 | [TK]D-Fender | VoipForces: http://pastebin.com/m757af8ba |
| 01:56.59 | VoipForces | [TK]D-Fender: Oh, now that's a way I did not think about |
| 01:57.49 | VoipForces | [TK]D-Fender: Let me give that a try right now. |
| 01:58.33 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
| 01:58.36 | [TK]D-Fender | VoipForces: I passed you the instruction page..... and copied the line right out of it... |
| 01:59.36 | VoipForces | [TK]D-Fender: I just did not read it carefull enough, must be my french brain operating at this hour |
| 02:00.07 | VoipForces | [TK]D-Fender: Having spent all day doing mysql and getting a named pipe script to work to log queue log directly in mysql |
| 02:02.05 | VoipForces | [TK]D-Fender: Still getting a Unable to request channel DAHDI/g4/ |
| 02:02.28 | [TK]D-Fender | VoipForces: "A Torontonian goes to Newfoundland and asks a Newfie neurosurgeon to turn him into a Newfie. Doctor turns to him and says he'll have to remove 75% of his brain. Guy agrees and wakes up 3 days after wards. The doctor tells him there was an accident and he removed 95% instead. The man, still drowsy turned to him and say 'C'est rien, c'est pas grave.'" |
| 02:02.43 | [TK]D-Fender | :p |
| 02:03.00 | VoipForces | [TK]D-Fender: Yes accessing channels directly works |
| 02:03.16 | VoipForces | [TK]D-Fender: Man been a while since I read a newfie joke |
| 02:03.21 | [TK]D-Fender | voidahdi show channel 29 |
| 02:05.48 | VoipForces | [TK]D-Fender: http://pastebin.com/m6b807308 |
| 02:07.10 | VoipForces | [TK]D-Fender: I think I'll gave to restert for the group change to be active |
| 02:14.44 | *** join/#asterisk etfonhomey (n=etfonhom@74-131-80-191.dhcp.insightbb.com) |
| 02:17.42 | VoipForces | Good night, I'm going to bed early for a change. Thanks [TK]D-Fender for the help. |
| 02:19.31 | [TK]D-Fender | VoipForces: Alrighty |
| 02:20.13 | VoipForces | [TK]D-Fender: Can not restart the server as it's sending faxes through the PRI right now anyway |
| 02:20.34 | [TK]D-Fender | VoipForces: Yet you have been supposedly restarting DAHDI... |
| 02:20.47 | [TK]D-Fender | VoipForces: Get some sleep and try tomorrow |
| 02:21.02 | VoipForces | [TK]D-Fender: I did wort case is that it corrupted 23 faxes lol |
| 02:24.29 | etfonhomey | [TK]D-Fender, on incoming FXO lines, is a hunt group the same exact thing as having rollover setup on the line? I see the terms used interchangeably all over the place. |
| 02:24.59 | [TK]D-Fender | etfonhomey: If the telco told you the term, then yes. |
| 02:26.12 | etfonhomey | [TK]D-Fender, I'm having an issue where the hunt groups I have setup on my phone system are lasting longer than the telco's timeout on their huntgroup. Is it common for a telco to be able to do a hunt group for busy only? |
| 02:26.28 | securevoip | having a problem getting video working from one x-lite client (exten 103) to another x-lite client (exten 104); http://pastebin.com/m721aff08 |
| 02:26.31 | securevoip | any ideas? |
| 02:26.39 | [TK]D-Fender | etfonhomey: Do not mix an ASTERSIK zap group with telco hunting |
| 02:30.17 | etfonhomey | [TK]D-Fender, I've got 3 FXO lines coming into my system at veterinary clinic. I want the call to come into reception and ring for 4 rings (~16 seconds), then ring in the office manager's office for 2 rings (~8 seconds), then go to an AA I have setup. |
| 02:31.01 | [TK]D-Fender | etfonhomey: So far has nothing to do with a telco hunt group |
| 02:32.15 | etfonhomey | [TK]D-Fender, Outgoing calls use the same FXO lines and if the reception is on the phone occupying one of the ports, when a second call comes in, I want the call to rollover to a free analog line. |
| 02:34.13 | [TK]D-Fender | etfonhomey: thats the telco's job, not yours |
| 02:36.32 | etfonhomey | [TK]D-Fender, exactly. Via a hunt group, no? |
| 02:36.44 | [TK]D-Fender | etfonhomey: Yes. |
| 02:41.19 | etfonhomey | [TK]D-Fender, but back to my main question is there such a service with the telco's that only rolls over if busy? One that if the line is not busy it will ring it perpetually with no timeout? |
| 02:42.09 | *** join/#asterisk salzh (n=Administ@202.106.126.231) |
| 02:45.35 | [TK]D-Fender | etfonhomey: That is the norm |
| 02:45.59 | [TK]D-Fender | etfonhomey: You can ask for a split "Busy" / "No-Answer" transfer option usually |
| 02:46.32 | *** join/#asterisk Kisu (n=k@ip70-179-88-179.dc.dc.cox.net) |
| 02:47.16 | *** join/#asterisk Kisu (n=k@ip70-179-88-179.dc.dc.cox.net) |
| 02:48.36 | *** join/#asterisk Kisu (n=k@daniel1117.broker.freenet6.net) |
| 02:51.10 | *** join/#asterisk paulius (n=paulius@unaffiliated/paulius) |
| 02:51.12 | paulius | Hey guys. Quick question about the SPA-3102. I was under the impression that I could connect my whole house's POTS wiring to it to allow my analog phones to work. But according to the manual, it says to only connect a single phone device. |
| 02:51.18 | paulius | Who's right and why? |
| 02:51.31 | jql | ... |
| 02:51.53 | paulius | ,,, |
| 02:52.00 | jql | Vonage says it's okay to use with theirs... PAP2Ts I believe |
| 02:52.32 | securevoip | do you know the REN (http://en.wikipedia.org/wiki/Ringer_equivalence_number)? |
| 02:52.53 | jql | if the internet worked better, I might be able to come up with infos |
| 02:53.28 | paulius | Is this aimed at me? |
| 02:53.34 | KyleK | paulius: the telco has access to the REN stuff |
| 02:53.51 | paulius | KyleK: Well what's that? |
| 02:54.16 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 02:54.26 | KyleK | the amount of power all the phones that are set to ring use matters |
| 02:54.44 | paulius | okay. |
| 02:54.56 | paulius | I don't have any old school phones. |
| 02:55.08 | KyleK | also if multiple phones pick up at the same time, the levels there might matter, if its all newer phones you should be fine |
| 02:55.23 | paulius | Uniden cordless phone |
| 02:55.30 | jql | vonage talks some big talk about their adapters... http://www.vonage.com/support.php?article=649 *5* phones |
| 02:55.44 | etfonhomey | [TK]D-Fender, these analog lines are from Comcast and if no one answers for 6 rings it starts ringing on another line in the group. I'm going to see if I can have them turn the timeout off or at least raise it to a large value. |
| 02:55.52 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 02:56.15 | KyleK | paulius: the base counts as one, the handsets hooked to it dont count |
| 02:56.24 | paulius | obviously. |
| 02:56.34 | KyleK | yup |
| 02:56.35 | paulius | I only have a cordless phone and then this other phone. |
| 02:56.40 | paulius | Don't think that it should be a concern. |
| 02:56.49 | paulius | so how much REN can the linksys thingie supply? |
| 02:56.56 | [TK]D-Fender | etfonhomey: etfonhomey they can. Its a question of if they will. |
| 02:57.01 | kc8pxy | REN? |
| 02:57.08 | [TK]D-Fender | paul7 is the norm |
| 02:57.10 | *** join/#asterisk securevoip (n=securevo@c-76-123-20-170.hsd1.va.comcast.net) |
| 02:57.21 | paulius | oh so that's a lot. |
| 02:57.26 | [TK]D-Fender | paulius: 7 is the norm |
| 02:57.35 | paulius | Like the wikipedia states, these digital phones shouldn't use more than 0.1 |
| 02:57.52 | paulius | And then I got another phone which the ringer is powered by the actual line, but its ringer is usually off. |
| 02:58.00 | [TK]D-Fender | paulius: Depends on the actual device load. powered phones don't use the voltage to power a mechanical bell so they flag in lower |
| 02:58.25 | paulius | And btw, I have full access to my demarc (NID) so I know how to plug it all in. |
| 02:58.42 | paulius | [TK]D-Fender: Okay but why are we discussing about this? |
| 03:00.00 | paulius | I don't think I'll ever come close to the limit. |
| 03:02.35 | [TK]D-Fender | paulius: Sorry for adding an enlightening and useful piece of knowledge to the conversation. Please remind me never to escape the minimalistac frame of reference of your questions again! :) |
| 03:02.51 | *** part/#asterisk etfonhomey (n=etfonhom@74-131-80-191.dhcp.insightbb.com) |
| 03:03.06 | paulius | [TK]D-Fender: No, it's neat that you do. But I'm just curious about the conclusion of this whole thing... Will it work or will it not and can it cause some bad things to happen |
| 03:03.09 | [TK]D-Fender | paulius: ... oh and I was writing mine before your 2 lines came in. |
| 03:03.27 | [TK]D-Fender | paulius: From what you've described you could run double that |
| 03:03.34 | paulius | yay |
| 03:04.26 | *** join/#asterisk sah-work (n=Bawbatos@adsl-76-193-217-250.dsl.pltn13.sbcglobal.net) |
| 03:05.24 | securevoip | anyone out there have sip video working in asterisk? |
| 03:05.29 | jql | a 1amp ata within 20-meters >>> money-conserving telco 1km away |
| 03:05.43 | paulius | [TK]D-Fender: So the short answer is that it should work, right? |
| 03:06.50 | [TK]D-Fender | paulius: how many ways do you want me to say "yes"? |
| 03:06.58 | paulius | okay okay fine |
| 03:06.59 | paulius | thanks |
| 03:07.16 | [TK]D-Fender | paulius: No, "okay" twice is still only 1 way! |
| 03:07.16 | jql | don't electocute yourself. good luck. :) |
| 03:07.29 | paulius | jql: Yeah don't worry. |
| 03:07.40 | jql | phone lines taste funny... |
| 03:07.44 | [TK]D-Fender | securevoip: I have |
| 03:08.43 | securevoip | securevoip: having a problem getting video working from one x-lite client (exten 103) to another x-lite client (exten 104); http://pastebin.com/m721aff08 |
| 03:09.41 | paulius | [TK]D-Fender: I'm honestly so excited for this whole vo-ip revolution in my house |
| 03:10.59 | [TK]D-Fender | securevoip: sip.conf masking only passwords please |
| 03:11.19 | Kumbang | hi guys, does asterisk-gui work for asterisk-1.6 |
| 03:11.43 | eppigy | we do not use asterisk-gui |
| 03:15.14 | securevoip | [TK]D-Fender: http://pastebin.com/m66b6d202 |
| 03:15.50 | [TK]D-Fender | Kumbang: Yes |
| 03:16.57 | [TK]D-Fender | securevoip: Looks OK, check your clients |
| 03:17.07 | [TK]D-Fender | securevoip: Test on both sides |
| 03:17.15 | paulius | thanks for your help |
| 03:17.20 | *** join/#asterisk trnzmeta (n=bleh@secure27.lnk.telstra.net) |
| 03:19.46 | securevoip | [TK]D-Fender: so far, I have tested GXV3000, Aethra Maia, 2 X-Lite, etc. Everything says "No matching video codec"??? |
| 03:20.19 | securevoip | [TK]D-Fender: Do you see any problem with using allow=all for each SIP device? |
| 03:21.15 | [TK]D-Fender | securevoip: test the 2 devices direct to each other |
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| 03:27.48 | *** part/#asterisk kb3ien (n=kb3ien@216.152.227.62) |
| 03:29.36 | Kumbang | i think | character doesn't work in asterisk-1.6, tried Goto(default|6000|1) |
| 03:30.20 | Kumbang | pbx_extension_helper: No such label 'default|6000|1' |
| 03:32.08 | RypPn | try a comma |
| 03:32.37 | RypPn | , = | in 1.6 iirc |
| 03:34.11 | [TK]D-Fender | Kumbang: As is well documented in the CHANGES |
| 03:36.07 | eppigy | HAHAAHAHA CHANGE LOGS |
| 03:36.22 | eppigy | RIDICULOUS |
| 03:36.52 | *** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) |
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| 04:20.13 | securevoip | [TK]D-Fender: works great if I take asterisk out of the picture... |
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| 05:28.54 | hardwire | does a little dance |
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| 05:35.40 | noisewaterphd | so, I've worked mainly with just Polycom SIP phones thusfar, but I've got an opportunity to pick up some Aastra and a few Grandstream phones for a really good price. Anyone have opinions on those two brands. How are they better/worse than Polycom stuff? |
| 05:39.23 | *** join/#asterisk viq (n=viq@unaffiliated/viq) |
| 05:52.57 | drmessano | ~grandstream |
| 05:52.58 | infobot | rumour has it, grandstream is the Yugo of VoIP hardware. Run. Run away now. |
| 05:53.04 | drmessano | ~grandsuck |
| 05:53.08 | drmessano | hmmm |
| 05:53.22 | drmessano | ~cisco |
| 05:53.23 | infobot | cisco makes the routers that move the pr0n across the internet at a very high rate of exchange. a company full of patent protected punks!, or <reply>Cisco phones are expensive crap which should be avoided with extreme prejudice |
| 05:54.27 | noisewaterphd | so no grandstream, I'd heard before they are built quite cheap. How about Aastra, any dirt on them? |
| 05:54.56 | dpryo | ~linksys |
| 05:54.57 | infobot | i heard linksys is a tool of satan |
| 05:55.01 | dpryo | :o |
| 05:55.13 | dpryo | SPA942 isn't that bad :) |
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| 07:09.32 | Erol_ | hi, do i need to seperate a voip network from the main network by something like VLAN? |
| 07:11.25 | henk | not necessarily |
| 07:16.55 | creativx | we arent |
| 07:17.25 | creativx | how saturated is your network today matters |
| 07:18.28 | henk | creativx: vlans don't help with a saturated network... |
| 07:24.36 | creativx | ofcourse not |
| 07:25.02 | creativx | but QOS can |
| 07:25.08 | creativx | depending on your infrastructure |
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| 07:53.56 | *** join/#asterisk proxium (n=proxium@196.203.51.238) |
| 07:57.38 | proxium | gm everyone, I want to upgrade my Asterisk Server from 1.4.22 to 1.6.0.9 and I have freePBX, any advice please to cleany make all turning with no problem? |
| 08:02.14 | lftsy | Hello proxium, I had no problem upgrading it on a Debian lenny server! |
| 08:02.16 | lftsy | Go ahead |
| 08:03.32 | proxium | lftsy: how to uninstall the previous and keep the config so FreePBX can use it? |
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| 08:27.42 | Gestahlt | Hi |
| 08:27.47 | Gestahlt | I have a big problem with asterisk |
| 08:28.03 | Gestahlt | Outbound calls from my ISDN Trunk working wonderful |
| 08:28.12 | Gestahlt | SIP to SIP also |
| 08:28.22 | Gestahlt | but i cant get any ISDN Inbound calls |
| 08:29.20 | Gestahlt | This is the capi debug which i get |
| 08:29.22 | Gestahlt | http://pastebin.com/m21c93fe8 |
| 08:29.47 | Gestahlt | ive got a fritzcard B1 (Active ISDN card). CAPI and such is installed and works fine |
| 08:30.08 | Gestahlt | Im sitting on this problem for weeks now |
| 08:30.23 | Gestahlt | i dont know what to do anymore.. |
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| 08:53.08 | BeeBuu | how can i make 3 members in a meet room can't speak and listen? |
| 08:55.07 | dpryo | You don't want them to listen? |
| 08:55.17 | dpryo | You can mute and kick users |
| 08:56.57 | BeeBuu | dpryo: but i want them back in some time |
| 08:59.22 | dpryo | Shouldn't be that hard to write something that parks the channels, if you know your way around asterisk :) |
| 08:59.41 | KyleK | can people be put on hold in a meet room? |
| 09:00.13 | dpryo | You can divert the channel to where you want |
| 09:01.41 | *** join/#asterisk ceegee (n=christia@mail.cg-networks.de) |
| 09:01.44 | ceegee | hello |
| 09:03.03 | KyleK | yo |
| 09:03.04 | ceegee | we are running asterisknow 1.5, now my boss wants that the external caller number shows up on call transfer, actually the number from the person who answered the call first is displayed on transfer |
| 09:03.34 | ceegee | I have no idea where to change this behavior |
| 09:04.17 | BeeBuu | dpryo: park the channels? which command is? |
| 09:04.33 | KyleK | ceegee: so if i transfer a call to you from conspiracy theory guy you wont know its him? |
| 09:04.59 | ceegee | KyleK: i give you an example |
| 09:06.09 | dpryo | BeeBuu: This might help you: http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Redirect |
| 09:07.20 | ceegee | KyleK: we receive a call from external, for example my colleagues wife, I answer the call, so I see her number in my display. I press hold and I dial the number of my colleague, when he answers my call, he has my number in display, after i hang up he should see the external number of his wife |
| 09:07.43 | BeeBuu | dpryo:redirect? can i tranfer them back? |
| 09:08.35 | ceegee | KyleK: you understand what i mean? |
| 09:09.44 | KyleK | caller id is usually only done once for an incoming call |
| 09:11.55 | ceegee | KyleK: as I see the problem, the caller id must change when I hang up the phone after successfully transfer, right? |
| 09:12.30 | KyleK | well for your colleague he only gets one call |
| 09:12.52 | KyleK | hey im going to transfer your wife onto the line, and then shes on the line |
| 09:15.10 | dpryo | BeeBuu: yes, you can redirect the channels whereever you want |
| 09:16.15 | KyleK | ceegee: i guess forwarding the call to your colleague without talking to him is out of the question? |
| 09:16.16 | BeeBuu | dpryo: thanks. |
| 09:16.29 | ceegee | KyleK: yes it is |
| 09:16.53 | ceegee | KyleK: I know about that, and I know that this works with correct number |
| 09:16.58 | Gestahlt | Can you guys help me with my problem as well? |
| 09:17.18 | ceegee | KyleK: but you know the rule "the boss is always right"? ;-) |
| 09:17.58 | KyleK | well whats the colleague using for a phone? hardware of software? |
| 09:19.18 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
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| 09:21.54 | ceegee | KyleK: snom 320 |
| 09:25.40 | BeeBuu | dpryo: can i call that in feature.conf? |
| 09:27.14 | KyleK | yaynetsplit |
| 09:27.30 | Gestahlt | suffers |
| 09:27.42 | Gestahlt | If i dont get asterisk working today i will just buy a finished pbx |
| 09:27.54 | Gestahlt | im kinda tired of the bullshit |
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| 09:30.53 | KyleK | ceegee: best bet would be to email snom about it, ask for ANY way to change whats on the display during a call, even if its via the web interface |
| 09:31.30 | ceegee | KyleK: ok thanks |
| 09:35.57 | KyleK | I wonder if a reinvite would do it |
| 09:35.58 | *** join/#asterisk matsk (n=matkar@193-44-227-98.customer.telia.com) |
| 09:37.31 | *** join/#asterisk saftsack (n=saftsack@p579246D9.dip.t-dialin.net) |
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| 09:56.29 | defswork | how do you pronounce dahdi ? |
| 10:08.12 | tzafrir_laptop | http://www.russellbryant.net/blog/2008/05/19/zaptel-project-being-renamed-to-dahdi/ http://www.russellbryant.net/dahdi.wav |
| 10:08.34 | tzafrir_laptop | ~dahdi |
| 10:08.35 | infobot | [~dahdi] Digium/Asterisk Hardware Device Interface (DAhdi). The new name of zaptel More info at http://www.asterisk.org/zaptel-to-dahdi , and is pronounced "dah-dee" with a short A, or pronounced like http://www.russellbryant.net/dahdi.wav |
| 10:24.23 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 10:27.01 | *** join/#asterisk casix (n=casix@pbxedifici.adamvozip.es) |
| 10:27.04 | casix | hello |
| 10:29.40 | *** join/#asterisk qdk (n=qdk@81.7.168.130) |
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| 10:34.19 | c4rg | does anyone actually help here? |
| 10:34.24 | *** part/#asterisk BeeBuu (n=beebuu@125.95.21.47) |
| 10:38.52 | mort_gib | Yes, quite a few people help out here |
| 10:40.53 | c4rg | ;-) |
| 10:41.20 | tzafrir_laptop | ~ask |
| 10:41.21 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 10:41.31 | c4rg | anyone having problems with faxing using wanpipe/dahdi/asterisk 1.4? |
| 10:42.02 | tzafrir_laptop | has still not spotted mr. Anyone on the channel |
| 10:42.51 | c4rg | how funny ;-) |
| 10:43.12 | tzafrir_laptop | try a more specific question |
| 10:56.46 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 11:09.01 | *** join/#asterisk smiley- (i=smiley@212.63.204.251) |
| 11:16.30 | orn | I am using Asterisk as an SMG, and when I have an inbound call from the PSTN, the SMG sends it to a SIP proxy. If that SIP proxy sends it back to the SMG with a diversion flag, the Asterisk sends a CANCEL to the proxy, and sends an invite back to the PSTN, keeping the SIP proxy out of the loop. How do I prevent that from happening? Also the Asterisk regards the new diverted call as 0 billsec. |
| 11:17.52 | smiley- | any reason why asterisk is creating about 12 tmp-files /minute in /tmp? files are called ast-ami-FsoIf9 |
| 11:18.03 | smiley- | with the last 6 chars random |
| 11:18.38 | *** join/#asterisk freddyk (n=freddy@host33-4-dynamic.52-79-r.retail.telecomitalia.it) |
| 11:18.55 | freddyk | can anyone help on fixing a problem with sip registrations ? |
| 11:19.18 | freddyk | i user register => user:password@section_name_with_host_options |
| 11:19.22 | freddyk | on asterisk 1.6.1.0 |
| 11:19.31 | freddyk | [May 7 13:19:26] WARNING[17186]: acl.c:376 ast_get_ip_or_srv: Unable to lookup 'trunk_euteliavoip' |
| 11:19.40 | freddyk | this is what i get |
| 11:19.50 | freddyk | it seems asterisk try to resolve section name as an hostname |
| 11:19.58 | freddyk | how can i get out of that ? |
| 11:29.36 | orn | put your sip.conf on pastebin.com |
| 11:32.55 | *** join/#asterisk inckie (i=kimse1@please.press.control-alt-delete.eu) |
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| 12:19.20 | smiley- | any reason why asterisk is creating about 12 tmp-files /minute in /tmp? files are called ast-ami-FsoIf9 |
| 12:19.56 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:28.35 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
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| 12:35.56 | *** join/#asterisk Kernel_Core (n=I@85.133.155.134) |
| 12:35.58 | Kernel_Core | hi all |
| 12:36.26 | Kernel_Core | I configured my TE110P card with the latest Dahdi and asterisk 1.4 |
| 12:36.33 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 12:36.41 | *** join/#asterisk stope (n=nobody@69.60.247.142) |
| 12:36.47 | *** part/#asterisk patrick-- (n=patrick@eos.openroot.de) |
| 12:36.51 | Kernel_Core | when I issue dahdi_cfg , it is Okey and I get 31/31/0 |
| 12:37.06 | Kernel_Core | but when I want to use it in asterisk , I get this error "[May 7 17:04:05] WARNING[28116]: chan_dahdi.c:2789 pri_find_dchan: No D-channels available! Using Primary channel 16 as D-channel anyway! |
| 12:37.07 | Kernel_Core | " |
| 12:39.28 | ltd | did you specify dchan= in the dahdi config? |
| 12:40.39 | Kernel_Core | yea |
| 12:41.30 | Kernel_Core | span=1,1,0,ccs,hdb3 |
| 12:41.31 | Kernel_Core | bchan=1-15 |
| 12:41.31 | Kernel_Core | dchan=16 |
| 12:41.31 | Kernel_Core | bchan=17-31 |
| 12:41.31 | Kernel_Core | echocanceller=oslec,1-15 |
| 12:41.31 | Kernel_Core | echocanceller=oslec,17-31 |
| 12:42.09 | Kernel_Core | ltd: when I run dahdi_tool I see the status of the card is OKey and I see 31/31/0 |
| 12:43.20 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 12:43.37 | *** join/#asterisk ariel_ (i=3fd6eca9@gateway/web/ajax/mibbit.com/x-59a4d32a4ea91694) |
| 12:44.55 | ltd | Kernel_Core: IIRC, that means that there was no signalling present on the defined d-channel. |
| 12:45.34 | ltd | Kernel_Core: Are you sure you don't want a ,crc4 on the end of your span= line? |
| 12:45.48 | ltd | Kernel_Core: not sure where you're located. We generally always use CRC4 here in aus. |
| 12:47.40 | *** join/#asterisk Gopaul (n=Miranda@59.97.120.228) |
| 12:48.23 | Gopaul | I am getting this kind of message in asterisk console, Executing [failed@from-internal:4] Macro("OutgoingSpoolFailed", "dialout-trunk|2|failed||") in new stack |
| 12:48.31 | Gopaul | what exactly it is |
| 12:48.33 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 12:48.46 | Kernel_Core | ltd: I am in Iran and asked from friend , he is a Cisco man , he told me the Iran configuration is the same as aus |
| 12:48.58 | seanbright | Gopaul: what are you using freepbx? |
| 12:49.02 | *** join/#asterisk andresmujica1 (n=andresmu@ubuntu/member/andresmujica) |
| 12:49.07 | Kernel_Core | ltd: but when I set CRC4 , my E1 doesn't go up ! |
| 12:49.55 | ltd | Gopaul: that extension is entered when the outgoing call fails. |
| 12:50.17 | ltd | Gopaul: "failed" |
| 12:50.48 | ltd | Kernel_Core: Hm. I can tell you 100%, with every provider I've used here in AU, CRC4 is required |
| 12:51.13 | ltd | Kernel_Core: But, if your E1 doesn't come up, that would indicate otherwise. |
| 12:52.34 | Kernel_Core | ltd: I enabled CRC4 and I got REC BLU/REC REC ..... |
| 12:52.42 | *** join/#asterisk jks (i=jks@193.189.93.254) |
| 12:54.16 | ltd | Kernel_Core: what alarm does it have? |
| 12:54.28 | Gopaul | seanbright: i am using trixbox |
| 12:54.36 | *** join/#asterisk smps (n=smps@193.170.53.51) |
| 12:54.38 | Kernel_Core | E1-PBX*CLI> pri show span 1 |
| 12:54.39 | Kernel_Core | Primary D-channel: 16 |
| 12:54.39 | Kernel_Core | Status: Provisioned, In Alarm, Down, Active |
| 12:54.39 | Kernel_Core | Switchtype: EuroISDN |
| 12:54.39 | Kernel_Core | Type: CPE |
| 12:54.43 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 12:54.50 | smps | hi |
| 12:54.50 | Gopaul | seanbright: sometime the outbound numbers are working and sometimes it is not getting the failed message |
| 12:54.59 | smps | i am experiencing segfaults on chan_local |
| 12:55.04 | Kernel_Core | ltd: I get this error |
| 12:55.17 | smps | anyone could help me ? |
| 12:55.23 | smps | i have backtrace from gdb |
| 12:55.51 | [TK]D-Fender | Gopaul: This is a dialplan issue, and GUI's are not supported in this channel |
| 12:55.59 | ltd | Kernel_Core: Not sure how to help you mate. Your config looks fine there apart from that. |
| 12:56.26 | [TK]D-Fender | Kernel_Core: PB your configs, and is your circuited confirmed active with the telco? |
| 12:56.32 | Gopaul | [TK]D-Fender: I am using AGI via originate action |
| 12:56.49 | ariel_ | Morning folks |
| 12:57.01 | Gopaul | [TK]D-Fender: I am not using any dialplan, only the voip account is registerd apart from that I am using originate action to originate a call from JAVA |
| 12:57.06 | Kernel_Core | ltd: thanks for your help |
| 12:57.20 | Kernel_Core | ltd: is it neccesary to send you my chan_dahdi.conf ?! |
| 12:57.29 | Kernel_Core | ltd: maybe there is an error ! |
| 12:57.31 | [TK]D-Fender | Gopaul: Your originate is doing what you're telling it to, and what you showed us is a line of dialplan executing. |
| 12:57.35 | ltd | Kernel_Core: pastebin it |
| 12:57.40 | [TK]D-Fender | Gopaul: This is not an "error" |
| 12:58.04 | [TK]D-Fender | Kernel_Core: Yes, there could be as you were only showing half of your configs |
| 12:58.31 | Gopaul | [TK]D-Fender: so when this kind of situation happens bcoz i am getting often |
| 12:58.37 | Gopaul | [TK]D-Fender: my scenario is like this |
| 12:58.40 | *** join/#asterisk ddunavant (i=4621b21e@gateway/web/ajax/mibbit.com/x-febc02b8c0b03c6f) |
| 12:59.08 | Gopaul | [TK]D-Fender: I am fetching a number from database and insetering into the originate action as a exten so that the number will get dialed thru voip account |
| 12:59.19 | Kernel_Core | http://pastebin.com/m61681e9a here is my extensions.conf and chan_dahdi.conf |
| 12:59.33 | Kernel_Core | ltd: please check it out |
| 12:59.35 | [TK]D-Fender | Gopaul: What you are seeing is dialplan executing, which is what Callfiles, AMI & CLI Originate fire off. These things "just work", and it IS processing. WHAT it is doing however is based on your dialplan and your GUI owns it, not you |
| 12:59.48 | ddunavant | so, quick question: does anyone know why this statement would simply fail: exten => s,n,Set(IntMenu=${IF($[ ${Path} = 6]?SLIntMenu:IRIntMenu)}) |
| 13:00.18 | Gopaul | [TK]D-Fender: So how to overcome this, I am still confused! |
| 13:00.46 | [TK]D-Fender | Gopaul: If your call is not processing like you want this is a DIALPLAN issue and not supported here |
| 13:00.57 | ltd | Kernel_Core: I don't see anything too out of the ordinary there in your chan_dahdi.conf |
| 13:01.38 | [TK]D-Fender | Kernel_Core: Is this a brand new card? |
| 13:01.53 | Gopaul | [TK]D-Fender: but i am not using any dialplan, the call is originated from JAVA |
| 13:01.55 | Kernel_Core | [TK]D-Fender: LoL ! no ! it is an old TE110P |
| 13:02.07 | ltd | digium or aftermarket? |
| 13:02.21 | Kernel_Core | digium |
| 13:02.31 | Gopaul | [TK]D-Fender: is it could be any originate action timeout problem |
| 13:02.43 | Gopaul | [TK]D-Fender:? |
| 13:02.49 | Kernel_Core | I used this command to call it modprobe wcte11xp t1e1override=0xFF |
| 13:02.49 | Kernel_Core | 0xFF= E1 mode |
| 13:02.49 | ltd | Kernel_Core: has your telco tested your circuit 100%? |
| 13:03.02 | Kernel_Core | ltd: it seems yea |
| 13:03.03 | [TK]D-Fender | Gopaul: you showed us a dialplan line being executed, what launched it is irrelevant |
| 13:03.21 | Kernel_Core | ltd: when your link is green they say it is OKey ! |
| 13:03.28 | [TK]D-Fender | Gopaul: No. |
| 13:03.31 | ltd | Kernel_Core: Not necessarily |
| 13:03.48 | Kernel_Core | ltd: so what can be source of the problem |
| 13:03.59 | Gopaul | [TK]D-Fender: can I pastebin my code? |
| 13:04.15 | [TK]D-Fender | Gopaul: Feel free, and include the COMPLETE output of your attempt |
| 13:04.40 | ltd | Kernel_Core: It might not be your config. It could be something physical or with your telco - since you're not seeing any D-channel activity |
| 13:05.22 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:05.27 | ltd | Kernel_Core: Have them come out with an analyzer and do a test call. |
| 13:05.43 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:05.43 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:05.50 | Gopaul | [TK]D-Fender: http://pastebin.com/m44a091e8 this is my output in my asterisk console |
| 13:05.58 | [TK]D-Fender | Kernel_Core: Funny thought is you are using the right kind of patch cable.... |
| 13:06.17 | ltd | [TK]D-Fender: If that was the case he wouldn't be seeing Layer1 up |
| 13:06.54 | [TK]D-Fender | ltd : guess we're all out of ideas ATM... |
| 13:07.47 | Gopaul | [TK]D-Fender: http://pastebin.com/m6596988e this is my code |
| 13:07.57 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 13:08.09 | Kernel_Core | [TK]D-Fender: if the patch cable is wrong , then is it possible to get the OK status in my E1 ?! |
| 13:08.26 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:08.28 | henk | is it possible to 'extend' a number i got from my sip provider so i can not just call '12345' but also '12345-1' and '12345-2'? |
| 13:08.47 | ltd | Kernel_Core: I'd call your telco and ask them to stick an analyzer on their end and see what they can see. I'd also ask them if CRC4 is a requirement or not. |
| 13:09.13 | Kernel_Core | ltd: I asked before , CRC isnot required |
| 13:09.30 | seanbright | Gopaul: that is just going to make [TK]D-Fender angry. |
| 13:09.59 | Kernel_Core | ltd: I have a cisco man friend , he send me a working E1 conf |
| 13:10.08 | Kernel_Core | controller E1 0 |
| 13:10.09 | Kernel_Core | clock source line primary |
| 13:10.09 | Kernel_Core | pri-group timeslots 1-31 |
| 13:10.12 | seanbright | you have a cisco man friend? |
| 13:10.13 | seanbright | that's hot. |
| 13:10.19 | Gopaul | seanbright: i am just pasting my code |
| 13:10.25 | ltd | rofl |
| 13:10.30 | seanbright | Gopaul: he is not interested in your java code |
| 13:10.32 | Kernel_Core | isdn switch-type primary-net5 |
| 13:10.49 | seanbright | Gopaul: he is saying that he wants asterisk CLI output of a failed called |
| 13:10.52 | Gopaul | seanbright: just for refrence |
| 13:11.08 | seanbright | Gopaul: all of the output, not just the one line you pasted when you first came in. |
| 13:11.09 | Gopaul | seanbright: the asterisk cli output also pasted |
| 13:11.16 | seanbright | ohhh |
| 13:11.18 | seanbright | shuts up |
| 13:11.20 | seanbright | and goes away |
| 13:11.23 | [TK]D-Fender | Gopaul: originateAction.setChannel("SIP:"+extnum+"@5066") <- you don't put ":" after SIP |
| 13:11.39 | Gopaul | [TK]D-Fender: that i have changed |
| 13:11.53 | [TK]D-Fender | Gopaul: And your action on all of these is to dump into the dialplan, and that we do not suppord |
| 13:12.00 | [TK]D-Fender | support |
| 13:12.17 | ltd | Kernel_Core: I really do suggest calling your telco. |
| 13:12.24 | Gopaul | [TK]D-Fender: can you please brief me I am not able to understand |
| 13:12.24 | leifmadsen | seanbright: you are especially snarky today :) |
| 13:12.39 | seanbright | leifmadsen: i'm in a good mood, what can i say? |
| 13:13.06 | [TK]D-Fender | Gopaul: Your Oriiginate dumps the channel into the dialplan. EXTEnsION.CONF. Your's is generated by a gui and is not supported here. |
| 13:13.23 | [TK]D-Fender | seanbright: ... he provided it |
| 13:13.35 | seanbright | [TK]D-Fender: yeah, took me a while to catch up |
| 13:14.28 | Gopaul | [TK]D-Fender: I am not having any GUI kind of thing |
| 13:14.29 | leifmadsen | I kinda wish extensions.conf would be renamed to dialplan.conf |
| 13:14.45 | leifmadsen | (with extensions.conf just being an alias for dialplan.conf) |
| 13:14.47 | [TK]D-Fender | Gopaul: TRIXBOX <--- |
| 13:14.56 | Kernel_Core | ltd [TK]D-Fender: thanks for you help ! |
| 13:15.28 | Gopaul | [TK]D-Fender: but the calls are getting originated by originate action via manager.conf rite? |
| 13:15.30 | [TK]D-Fender | [09:10]<Kernel_Core>isdn switch-type primary-net5 <-- was this what you needed to account for? |
| 13:15.48 | [TK]D-Fender | Gopaul: Yes, they start there and enter the DIALPLAN. |
| 13:16.03 | ltd | primary-net5 is the same as is used in aus. |
| 13:16.27 | Gopaul | [TK]D-Fender: so how should I originate my call? |
| 13:16.29 | seanbright | leifmadsen: open a bug on the bug tracker |
| 13:16.40 | ltd | primary-ts015 for older stuff |
| 13:16.46 | leifmadsen | seanbright: it's a feature request and I'd have to close it without a patch :) |
| 13:16.53 | [TK]D-Fender | Gopaul: Go ask in your GUI support channel. You don't seem to know what you're doing with it |
| 13:16.58 | leifmadsen | unless I assign it to you of course :) |
| 13:17.02 | seanbright | leifmadsen: you're a man of principle and i respect that. |
| 13:17.13 | leifmadsen | seanbright: even I don't get special privileges :) |
| 13:17.33 | seanbright | leifmadsen: ln -s extensions.conf dialplan.conf |
| 13:17.42 | seanbright | now pay me my money |
| 13:18.51 | leifmadsen | :) |
| 13:18.55 | *** part/#asterisk smiley- (i=smiley@212.63.204.251) |
| 13:18.56 | Gopaul | [TK]D-Fender: thanks for your support |
| 13:19.13 | leifmadsen | seanbright: I'll pay you the same amount that I pay all the other issue closers |
| 13:19.24 | leifmadsen | I'll even pay you a bonus of 20% |
| 13:21.47 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 13:23.06 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 13:23.42 | *** join/#asterisk Brixius (n=Brixius@PDN-VBA.OnvoyInc.fw.onvoy.net) |
| 13:23.43 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/professional/sulex) |
| 13:23.45 | Brixius | Hello |
| 13:23.59 | seanbright | leifmadsen: with that and $2 i can buy a cup of coffee! |
| 13:24.31 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 13:24.54 | Brixius | I have a question with an asterisk dialplan modification, I'm trying to migrate from the MySQL command to use res_odbc instead. |
| 13:25.23 | *** join/#asterisk xrmx__ (n=rm@host75-254-dynamic.5-87-r.retail.telecomitalia.it) |
| 13:25.26 | Brixius | However I'm not sure how to modify the following line to use res_odbc. |
| 13:25.27 | Anth8708 | quick question (btw, you guys are awesome, specially Fender): i've setup a hint for an extension, it shows with core show hints. I have a polycom 560 watching (shows on the phone and with core show subscriptions), but then the extension being watched is in use/ringing, etc, the hint isn't updated (core show hints still shows idle). I've added the items to sip.conf [general] as directed in... |
| 13:25.29 | Anth8708 | ...the wiki, any ideas? |
| 13:26.22 | leifmadsen | Brixius: show via pastebin, usually it's quite simple to move from MYSQL() to func_odbc |
| 13:26.34 | Brixius | ok |
| 13:26.55 | Brixius | one second. |
| 13:27.02 | *** join/#asterisk PoWeRKiLL (n=lironech@bzq-84-108-86-122.cablep.bezeqint.net) |
| 13:27.10 | [TK]D-Fender | Anth8708: WIKI is often outdated crap. Pastebin is your friend, so are details like what * ver, etc. |
| 13:27.12 | [TK]D-Fender | ~pb |
| 13:27.13 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines 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 , or , http://bin.cakephp.org/ |
| 13:27.14 | [TK]D-Fender | ^^^^^^ |
| 13:27.26 | Anth8708 | rgr. just a sec |
| 13:28.29 | Brixius | http://pastebin.com/m6e5f039f |
| 13:29.35 | Brixius | I can't seem to find how I would change a fetch command that fetches multiple results to a res_odbc config without doing a seperate query for each item. |
| 13:30.07 | Brixius | er multiple results = multiple columns |
| 13:30.37 | leifmadsen | Brixius: exten => foo,n,Set(ARRAY(col1,col2,col3)=${ODBC_GET_VALUES(${ARG1})}) |
| 13:30.58 | leifmadsen | then each value from the column is in a variable: ${col1}, ${col2}, etc... |
| 13:31.20 | leifmadsen | btw: replace ${ARG1} with ${MACRO_EXTEN} |
| 13:31.37 | Brixius | ahh, ok that makes sense. |
| 13:31.47 | leifmadsen | then it is ${ARG1} in the func_odbc.conf file |
| 13:31.53 | Anth8708 | OK. Hint issue: http://pastebin.com/d5c45e0ad |
| 13:32.02 | leifmadsen | read=select .... from tablename where foo = '${ARG1}' |
| 13:32.24 | leifmadsen | Brixius: or (although I can't remember if 1.4 has it native) you could use HASH() |
| 13:32.27 | Brixius | yep, I can write the query's in res_odbc and move them there, I just wasn't sure about the set statement for multiple columns |
| 13:32.33 | leifmadsen | which basically is similar in concept to ARRAY() |
| 13:33.00 | leifmadsen | Brixius: yep, use ARRAY() is probably your best bet |
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| 13:33.33 | Brixius | thanks, I'll give that a go. |
| 13:33.33 | leifmadsen | coolio |
| 13:33.34 | leifmadsen | you can set the variable names to whatever you want btw |
| 13:33.34 | leifmadsen | so just name the variables the same as your column name if you wish |
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| 13:45.42 | [TK]D-Fender | Anth8708: You know... it'd be nice to see the sip.conf to match... masking only passwords |
| 13:45.58 | Anth8708 | you bet, paste bin coming up |
| 13:46.29 | eppigy | hello |
| 13:46.31 | eppigy | i am dave |
| 13:48.00 | [TK]D-Fender | eppigy: http://tinyurl.com/ytzx8y |
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| 13:53.04 | philippel | question, if I have a hint composed of two devices, one is Idle and the other OnHold, what would you expect the value of the hint to return? |
| 13:54.06 | philippel | and similar question, but now one is OnHold and the other is Busy, now what should it return? |
| 13:54.30 | Anth8708 | [TK]D-Fender: Thanks for looking at this. pb with sip.conf (secrets blanked out) and relevant extensions.conf info as well. http://pastebin.com/d377a9873 |
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| 13:59.21 | eppigy | [TK]D-Fender: lol |
| 14:00.30 | [TK]D-Fender | Anth8708: type=friend <-- limitonpeers works a LOT better when TYPE=PEER |
| 14:02.02 | [TK]D-Fender | Anth8708: You should also have something like "call-limit=99" for your phones |
| 14:03.47 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 14:04.05 | jaytee | hmmm, I've used limitonpeers=yes with call-limit=1 successfully on Linksys ATAs |
| 14:04.14 | jaytee | with type=friend |
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| 14:05.39 | telnettech | question: anyone use 3cDaemon as a syslog server? |
| 14:07.18 | jaytee | nope, never used it. |
| 14:08.00 | jaytee | it's vulnerable to a DoS exploit, but then it runs on Windows so what would you expect? |
| 14:08.17 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 14:08.17 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 14:10.07 | Anth8708 | [TK]D-Fender: so my entries in sip.conf should have type=peer and call-limit=99 instead of 100? or does call-limit need to be added to each separate entry? thanks SO much for looking at this again |
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| 14:10.24 | Magicblaze0071 | I am looking for a easy to configure voip router to give my granddad...any recommendations? I was initially thinking of PAP2T-NA but then he has to buy a router... |
| 14:10.40 | telnettech | It didnt write to a file and I need the captured info desperately |
| 14:11.15 | telnettech | so i was looking to see if someone has used it and if there is a copy/paste command that Im missing |
| 14:11.48 | telnettech | I have 2 days of syslog messages to capture |
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| 14:22.40 | Anth8708 | [TK]D-Fender: Thanks! That was it. Adding the call-limit to the entry and not under general. It's working! |
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| 14:23.43 | Katty | presents file with a muffin |
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| 14:29.20 | jaytee | mornin Katty *hugs* |
| 14:30.35 | Katty | hugs jaytee |
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| 14:45.29 | [TK]D-Fender | Anth8708: You're welcome |
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| 14:46.02 | *** join/#asterisk mindi (n=MinD@xdsl-87-78-142-162.netcologne.de) |
| 14:46.10 | mindi | hello |
| 14:47.15 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 14:47.52 | mindi | im tryin to make a call from the CLI, but all console-commands are missing |
| 14:48.24 | mindi | ive read there could be a problem with chan_oss.so |
| 14:48.50 | tzafrir_laptop | mindi, prefix them with 'console' |
| 14:48.55 | mindi | i did |
| 14:48.56 | tzafrir_laptop | console <tab><tab> |
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| 14:50.05 | mindi | nothing - not even when I enter 'help console' |
| 14:50.11 | *** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com) |
| 14:50.52 | [TK]D-Fender | mindi: "core set verbose 10" |
| 14:51.28 | mindi | got it |
| 14:51.44 | mindi | ? |
| 14:52.05 | [TK]D-Fender | mindi: So try again |
| 14:52.08 | mindi | ... No such command 'console' |
| 14:52.15 | [TK]D-Fender | mindi: pastbin your modules.conf |
| 14:52.21 | [TK]D-Fender | ~pb |
| 14:52.22 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines 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 , or , http://bin.cakephp.org/ |
| 14:52.24 | [TK]D-Fender | ^^^^^ |
| 14:52.42 | [TK]D-Fender | mindi: And please confirm your VERsiON <- |
| 14:53.08 | mindi | i just upgraded to asterisk 1.6 |
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| 14:55.13 | mindi | http://pastebin.com/d7a059375 |
| 14:55.39 | *** part/#asterisk gego (n=rick@b238085.customer.hansenet.de) |
| 14:55.41 | ayeso | I cant seem to find a function/app that will let me read a file either line by line or enitrely into a variable.. does this exist? |
| 14:58.55 | mindi | i dont have the module chan_oss.so |
| 14:59.13 | mindi | I guess this is necessary |
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| 15:00.28 | mindi | can I add it somehow? Ive found the source here: http://svn.dd-wrt.com:8000/dd-wrt/browser/src/router/asterisk/channels/chan_oss.c?rev=11933 |
| 15:00.41 | tzafrir_laptop | mindi, it is probably not loaded by default . try: module load chan_oss.so |
| 15:00.53 | tzafrir_laptop | (you may be missing some other parts of modules.conf in that paste) |
| 15:01.14 | tzafrir_laptop | and chan_oss has not been dropped from asterisk |
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| 15:05.06 | mindi | sry, ill make another post, but chan_oss.so doesnt show up with 'module show' |
| 15:05.29 | *** join/#asterisk horvath (n=horvath@74-51-44-45.telnetcommunications.com) |
| 15:06.28 | horvath | I setup BLF on some Linksys SPA942's. I'm wondering why BLF works fine for incoming calls but somehow I'm missing the hints for the outgoing calls. Is there a additional hint line that should be in the outbound context? |
| 15:07.14 | *** join/#asterisk jksM (i=jks@193.189.93.254) |
| 15:07.18 | mindi | http://pastebin.com/d3a252e6 |
| 15:11.36 | tzafrir_laptop | mindi, again, have you tried: module load chan_oss.so ? |
| 15:14.13 | mindi | yes |
| 15:14.49 | mindi | Command 'module load chan_oss.so' failed. |
| 15:16.27 | tzafrir_laptop | what error? |
| 15:17.04 | mindi | Command 'module load chan_oss.so' failed. |
| 15:23.25 | ayeso | Can I have asterisk run an external script (python) and get a variable back? |
| 15:23.31 | *** join/#asterisk marv[work] (n=timr@router.asteriasgi.com) |
| 15:25.45 | leifmadsen | ayeso: yes -- it's called AGI |
| 15:26.00 | leifmadsen | ~agi |
| 15:26.01 | infobot | hmm... agi is the Asterisk Gateway Interface... similar to CGI for web applications AGI lets you script call control and access databases using your favorite language. AGI wrappers are available for Python (pyst), Perl (astperl?) and other languages, or <reply> See also http://www.voip-info.org/wiki-Asterisk+AGI |
| 15:26.05 | mindi | could this be some problem with AsteriskNow? |
| 15:26.23 | tzafrir_laptop | ayeso, core show function SHELL |
| 15:26.33 | tzafrir_laptop | (available as of 1.6.0, IIRC) |
| 15:26.49 | leifmadsen | mindi: probably not compiled |
| 15:27.01 | leifmadsen | mindi: ls /var/lib/asterisk/modules/chan_oss* |
| 15:27.35 | ayeso | tzafrir_laptop: I dont seem to have that |
| 15:27.52 | ayeso | leifmadsen: ill take a look, thx |
| 15:29.05 | mindi | ls: /var/lib/asterisk/modules/chan_oss*: No such file or directory |
| 15:30.59 | leifmadsen | mindi: can't load a module that isn't compiled |
| 15:32.01 | mindi | so i gotta get the source and make it myself, right? |
| 15:32.37 | mindi | its not available via YUM.... |
| 15:32.44 | carrar | sources is best! |
| 15:32.53 | mindi | where? |
| 15:33.21 | [TK]D-Fender | ayeso: SHELL is in 1.6.0.5 confirmed |
| 15:33.48 | [TK]D-Fender | tzafrir_laptop: Looks like SHELL is a native form of the old 3rd party "backticks" |
| 15:34.07 | carrar | hhah evil backticks |
| 15:34.19 | leifmadsen | mindi: svn co http://svn.digium.com/svn/asterisk/tags/<tagged_version> |
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| 15:35.58 | tzafrir_laptop | [TK]D-Fender, right. but it is a function rather than application. |
| 15:36.17 | freddyk | hi all |
| 15:36.22 | [TK]D-Fender | tzafrir_laptop: Yup, much nicer that way, and I'm sure the "gotcha" he failed on ;) |
| 15:36.43 | freddyk | i have a problems with chan sip over asterisk 1.6.1.0 on multiple registrations, multiple servers |
| 15:36.48 | [TK]D-Fender | [11:27]<ayeso>tzafrir_laptop: I dont seem to have that <-- show us |
| 15:37.01 | freddyk | as it try to get hostname from [sectionname] |
| 15:37.07 | freddyk | can anyone help? |
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| 15:37.49 | ayeso | [TK]D-Fender: I think im going to go the AGI route, but a core show function SHELL returns: No function by that name registered. |
| 15:38.41 | [TK]D-Fender | ayeso: What ver exactly are you on? |
| 15:40.08 | *** join/#asterisk mohawk (n=ross@host217-40-110-153.in-addr.btopenworld.com) |
| 15:40.55 | oglynn | i have a pri question call inbound on PRI and i send it out to a second PRI when the original caller hangs up i get what looks like an incoming call for 'h' |
| 15:41.45 | [TK]D-Fender | oglynn: Perfectly normal. |
| 15:43.13 | oglynn | [TK]D-Fender i am trying to get 2BCT working and it seems like i almost end up with a second call afterwards and wanted to make sure this initial "error" was not part of it |
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| 15:45.38 | ayeso | [TK]D-Fender: Asterisk 1.4.23.1 |
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| 15:52.11 | oglynn | <PROTECTED> |
| 15:52.59 | [TK]D-Fender | You're call out another PRI. AFAIK you can't do a 2BCT over 2 different links |
| 15:53.37 | oglynn | [TK]D-Fender sorry for the confusion. I was asking about the initial error in case that was a factor this is over the same PRI |
| 15:53.39 | [TK]D-Fender | ayeso: And you were jsut specifically told that this function was added in 1.6.0 So telling us it's not there isn't a surprise |
| 15:54.27 | *** join/#asterisk Witch_Doc (n=me@69.196.64.50) |
| 15:54.28 | [TK]D-Fender | oglynn: you need to set "transfer=yes" for those channels to enable 2BCT and it needs to be supported by your protocol and carrier |
| 15:54.49 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:54.50 | Vec | When I phone puts a user on hold what SIP msgs are sent to asterisk ? |
| 15:55.00 | Vec | or rather SIP methods |
| 15:55.01 | Witch_Doc | anyone know of a provider that will allow me to get a US DID and call forward to my cell phone without charging long distance for the forwarding? |
| 15:55.27 | [TK]D-Fender | Witch_Doc: Where is your cell located? |
| 15:55.32 | Witch_Doc | 604 |
| 15:55.43 | Witch_Doc | DID i'd like for either 240 or 301 |
| 15:55.53 | Witch_Doc | 604 = vancouver, bc |
| 15:55.58 | Witch_Doc | 240 = washington, dc |
| 15:56.40 | [TK]D-Fender | Witch_Doc: You'd need to have 2 calls involved. 1 in to *, from the 240, and another placing a call to the 604 |
| 15:56.57 | Witch_Doc | hmm |
| 15:57.06 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 15:57.11 | [TK]D-Fender | Witch_Doc: Inbound DID's you can get for maybe $8/mo or so, the outbound is the other factor |
| 15:57.15 | Witch_Doc | aren't there any sip providers that could do it on their end for me? |
| 15:57.42 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 15:57.42 | [TK]D-Fender | Witch_Doc: Never an advertised service. Maybe someone will.... |
| 15:57.43 | Witch_Doc | for example vonage allows forwarding of their DID's on their end |
| 15:57.54 | Witch_Doc | not sure if there are others though |
| 15:58.13 | [TK]D-Fender | Witch_Doc: Noone asks this here, so you're just going to have to ask around yourself |
| 15:58.22 | Witch_Doc | ok thanks |
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| 16:10.58 | oglynn | [TK]D-Fender does the transfer=yes need to be set per group? I see "Requested transfer capability" in mu console |
| 16:12.08 | Witch_Doc | can anyone recommend a good free sip provider? |
| 16:14.50 | Vec | Isn't nothing in life free ? |
| 16:15.00 | Witch_Doc | vec double negative? |
| 16:15.18 | Vec | I guess thats a very negative statement |
| 16:15.25 | Vec | Is nothing in life free ? |
| 16:15.43 | *** join/#asterisk qdk (n=qdk@87.61.141.209) |
| 16:15.59 | Vec | Isn't it true that nothing in life is for free ? << does that make sense ? |
| 16:16.14 | Vec | errr, still double neg |
| 16:19.14 | ayeso | Where do AGI scipts need to live in the filesystem? or can i specify the path when I call it? |
| 16:20.38 | Magicblaze0071 | Anyone knows of a better place to buy a SPA-3102 compared to this one: http://www.ipphone-warehouse.com/ProductDetails.asp?ProductCode=spa3102 |
| 16:20.58 | orn | I am using Asterisk as an SMG, and when I have an inbound call from the PSTN, the SMG sends it to a SIP proxy. If that SIP proxy sends it back to the SMG with a diversion flag, the Asterisk sends a CANCEL to the proxy, and sends an invite back to the PSTN, keeping the SIP proxy out of the loop. How do I prevent that from happening? Also the Asterisk regards the new diverted call as 0 billsec. |
| 16:21.55 | orn | If that is even possible, that is. |
| 16:25.26 | ayeso | found it here if anyone was interested: /var/lib/asterisk/agi-bin |
| 16:28.23 | *** join/#asterisk murdock_ut (n=chatzill@mail.kimballequipment.com) |
| 16:29.00 | *** join/#asterisk prashant_jois (n=prashant@68.148.97.186) |
| 16:30.58 | prashant_jois | I have a question regarding IAX connections. When I create an entry in my iax.conf with host=<IP address> my connection works, however, when I put in a domain name, e.g. host=www.mydomain.com, I get a "cause 3 - no route to destionation" error. |
| 16:31.08 | *** join/#asterisk Firass-z0r (n=asadf@c-67-201-205-34.reshall.wwu.edu) |
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| 16:44.38 | OneZero1010101 | Anyone alive ? |
| 16:46.16 | gordonjcp | is |
| 16:46.32 | gordonjcp | or at least, no-one's told me otherwise |
| 16:46.50 | OneZero1010101 | I'm looking for some info on hardware for integration |
| 16:46.58 | Ziaeon | even if the signs seem to tell you otherwise |
| 16:47.02 | OneZero1010101 | lol |
| 16:47.26 | OneZero1010101 | just looking for someone with a bit of knowledge that can give me a bit of info |
| 16:47.32 | seanbright | ~ask |
| 16:47.33 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 16:47.33 | OneZero1010101 | or point me to where to find it |
| 16:47.51 | OneZero1010101 | Easy nuff |
| 16:47.52 | *** join/#asterisk codefreeze-lap (n=murf@72.21.67.40) |
| 16:47.58 | Ziaeon | I'm here against my will |
| 16:48.01 | OneZero1010101 | LOL |
| 16:49.14 | *** join/#asterisk bmoraca (n=chatzill@66.242.174.254) |
| 16:49.15 | OneZero1010101 | I am needing to put together a voip phone system. Am trying to find out what type of hardware, or what equipment I will need to locate to use to do this. I know I will need phones (given), some sort of call manager, which from my understanding is what Asterisk is, as well as some sort of voice / router / gateway or something. |
| 16:49.37 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 16:49.48 | OneZero1010101 | Am I on the right track, trying figure out what pieces I need to put WITH an asterisk server to make this work |
| 16:49.50 | Qwell | OneZero1010101: Asterisk is also the latter. |
| 16:49.51 | seanbright | first stop would be "Asterisk: The Future of Telephony" |
| 16:49.52 | Qwell | ~book |
| 16:49.53 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 16:49.57 | seanbright | what Qwell said. |
| 16:50.11 | seanbright | or infobot, specifically. |
| 16:50.31 | Qwell | well, I never. |
| 16:50.57 | bmoraca | lol |
| 16:50.59 | OneZero1010101 | Where's the cliffs notes lol |
| 16:51.09 | Qwell | OneZero1010101: That is the cliffs notes. |
| 16:51.22 | OneZero1010101 | LOL |
| 16:51.36 | OneZero1010101 | Errr hrmm |
| 16:52.11 | seanbright | you need a server |
| 16:52.27 | seanbright | which has a cpu, some ram, maybe a hard drive, and a network card if you want to do anything |
| 16:53.02 | OneZero1010101 | Yes, i understand this is software on the server... need switches to connect voip phones to |
| 16:53.14 | bmoraca | OneZero1010101: if you want to start using Asterisk, you need to get a good understanding of how it works. yes, there are "turn-key" distros out there that simplify things, but without understanding what they do and why, you'll be a few bricks short of a house |
| 16:53.21 | seanbright | if you're going pure voip (not isdn pri or fxs/fxo) you'll need the appropriate cards which you can get from digium, sangoma, or other vendor. |
| 16:53.27 | OneZero1010101 | How do I get dial tone into the box |
| 16:53.31 | seanbright | err... |
| 16:53.41 | Qwell | OneZero1010101: by reading the aforementioned book. |
| 16:53.41 | seanbright | s/'re going/'re NOT going/ |
| 16:53.46 | gordonjcp | why does asterisk appear to ignore my rtp port settings in rtp.conf? |
| 16:53.49 | *** join/#asterisk Chex (n=Stefan@bas1-montreal48-1176430935.dsl.bell.ca) |
| 16:53.56 | *** join/#asterisk j0 (n=dan@S0106000c29242337.va.shawcable.net) |
| 16:53.58 | OneZero1010101 | k |
| 16:54.44 | bmoraca | OneZero1010101: if this is for a business, pay someone to do it for you. |
| 16:54.48 | seanbright | like me |
| 16:54.56 | Qwell | like him ^^ |
| 16:54.58 | Qwell | wait, what? |
| 16:55.00 | seanbright | heh |
| 16:55.18 | OneZero1010101 | i'm going thru the book, just trying to wrap my head around it |
| 16:55.45 | *** join/#asterisk jeff (i=jeff@unaffiliated/jeff) |
| 16:56.00 | [TK]D-Fender | OneZero1010101: You'll wat to remove the binding and stich the pages end-to-end then. that will help with the wrapping |
| 16:56.09 | seanbright | ba dum dum |
| 16:56.16 | OneZero1010101 | that would help ;) |
| 16:56.34 | [TK]D-Fender | OneZero1010101: here : |
| 16:56.36 | [TK]D-Fender | ~osmosis |
| 16:56.37 | infobot | [~osmosis] Osmosis is the act of beating yourself on the head repeatedly with THE BOOK, until some measure of absorption has occured ... or at least until your unconsciousness restores peace to the channel ... |
| 16:56.42 | OneZero1010101 | YES! |
| 16:56.50 | *** join/#asterisk Killabeez (i=Killabee@c-24-126-188-122.hsd1.ga.comcast.net) |
| 16:57.28 | j0 | what is it called when you have multiple analog lines (each with different phone numbers) and if the inbound line is busy for 1 number, it will send the call to another line? |
| 16:57.42 | seanbright | rollover |
| 16:57.43 | seanbright | heh |
| 16:57.43 | j0 | here i think they call it "overline", but i google is coming up empty for that |
| 16:57.53 | Killabeez | my company has just stuck me with being responsible over the asterisk system, oh joy.. I know absolutely nothing about this software, I heard its kinda a nightmare? |
| 16:58.03 | Qwell | ~book |
| 16:58.04 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 16:58.06 | Qwell | Killabeez: you too. |
| 16:58.17 | j0 | seanbright: ah.. is that supported in asterisk? |
| 16:58.28 | j0 | i'm just not sure how asterisk can tell what # the call is destined for |
| 16:58.42 | Qwell | j0: it can't tell on analog lines |
| 16:58.54 | seanbright | j0: we have 8 analog lines here, and the telco will rollover from one to the other when they are busy |
| 16:59.04 | seanbright | j0: and as Qwell says, over analog you don't know the dialed number |
| 16:59.16 | j0 | seanbright: do you have more than 1 inbound phone number? |
| 16:59.24 | Qwell | for analog? you'd have to. |
| 16:59.27 | seanbright | j0: yeah, each analog line has it's own number |
| 16:59.40 | Qwell | j0: depending on how many lines we're talking about, I would just say get a PRI |
| 16:59.43 | j0 | i see... :) |
| 16:59.49 | Qwell | even if just a partial. it may even end up being cheaper |
| 16:59.53 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 17:00.01 | murdock_ut | j0: A lot of telco's call them hunt groups. |
| 17:00.12 | seanbright | j0: but we have a "main" number which is associated with the first analog line. if that number is called and that line is busy, the second analog line is rung, etc, etc. |
| 17:00.21 | seanbright | murdock_ut: yes. that's the term. |
| 17:00.24 | j0 | Qwell: here a pri costs a small fortune... but i see that it's just impossible to know what # was dialed on an analog line |
| 17:00.26 | seanbright | j0: what murdock_ut said. |
| 17:00.32 | murdock_ut | j0: Have your telco setup a hunt group for you. |
| 17:00.41 | Qwell | j0: even a partial PRI? you can often get ~8 lines |
| 17:00.42 | seanbright | j0: how many analog lines do you have? |
| 17:00.56 | jameswf | j0 you can use analog DID |
| 17:01.29 | OneZero1010101 | Is it possible to bring a PRI in on your fiber internet access? |
| 17:01.36 | OneZero1010101 | Or do you need a dedicated PRI circuit |
| 17:01.36 | j0 | jameswf: i'll have to see if they support it |
| 17:01.47 | j0 | Qwell: 6, but they're shared between multiple companies |
| 17:02.12 | bmoraca | OneZero1010101: I can get you a PRI on your fiber internet access :P |
| 17:02.14 | j0 | i'll be getting a pri anyway, but just wanted to know the alternatives for when i look at future setups |
| 17:02.23 | murdock_ut | j0: Your going to want a pri then. |
| 17:02.39 | bmoraca | $30 per channel per month plus $30 per month for the equipment :P |
| 17:02.58 | OneZero1010101 | channel = line? so $30 / line? |
| 17:03.04 | bmoraca | yes |
| 17:03.11 | OneZero1010101 | Who would this be through |
| 17:03.11 | j0 | wow.. 8 channel here is $500/month on a 3 year |
| 17:03.21 | murdock_ut | j0: Then you have your DID's and Channels which will make it easier when you have multiple companies sharing the same lines. |
| 17:03.24 | seanbright | j0: where is "here?" |
| 17:03.29 | bmoraca | it would be through me...but only if you're in northern CA |
| 17:03.34 | j0 | seanbright: an hour out of vancouver, bc |
| 17:03.38 | OneZero1010101 | There in lies the problem then |
| 17:03.42 | OneZero1010101 | Southern IN |
| 17:03.48 | bmoraca | i wasn't being entirely serious |
| 17:03.49 | seanbright | OneZero1010101: he was messing with you |
| 17:03.53 | OneZero1010101 | sitting on 2x50mb fiber circuits |
| 17:03.55 | murdock_ut | j0: What type of internet connection do you have? |
| 17:04.06 | seanbright | OneZero1010101: who provides your internet access? |
| 17:04.07 | j0 | murdock_ut: dsl, not reliable for voip |
| 17:04.12 | bmoraca | i do have the service, but i'm not going to peddle my wares in a chat room :) |
| 17:04.14 | murdock_ut | J0: nope. |
| 17:04.29 | *** join/#asterisk mahlon (i=mahlon@martini.nu) |
| 17:04.37 | OneZero1010101 | We have two upstream, a local provider Indiana Fiber Networks, and Insight Business |
| 17:04.51 | j0 | at some point i may try forward-on-busy to a voip provider |
| 17:04.55 | OneZero1010101 | We are a Municipal WISP |
| 17:05.00 | Killabeez | so my question since the asterisk server I have been put in charge of is currently active and working, I need to document what we have, IP addys, logins? |
| 17:05.14 | seanbright | Killabeez: hire a consultant. |
| 17:06.00 | Killabeez | seanbright sit on a anthill. |
| 17:06.07 | seanbright | ? |
| 17:06.19 | murdock_ut | Killabeez: Tickle Tickle |
| 17:06.23 | Killabeez | heh |
| 17:06.31 | seanbright | IP addys and logins have nothing to do with asterisk |
| 17:07.12 | seanbright | do you have linux system admins? (or whatever flavor of unix you are running) |
| 17:07.55 | Killabeez | seanbright yea im one of the senior admins here, cent0s |
| 17:08.09 | seanbright | alrighty |
| 17:08.17 | seanbright | ifconfig and 'cat /etc/passwd' |
| 17:08.31 | Killabeez | seanbright i was looking at trixbox, is this a decent asterisk build? |
| 17:08.35 | bmoraca | Killabeez: look@lan will be an invaluable resource for you in mapping out a network, I'd imagine. |
| 17:08.39 | Qwell | Killabeez: NO! |
| 17:08.51 | Killabeez | do it by hand? |
| 17:08.57 | seanbright | i'd say if you had a active and working install, don't fuck with it |
| 17:09.03 | Killabeez | seanbright alright |
| 17:09.04 | Qwell | or use something besides trixbox |
| 17:09.18 | jameswf | ~trixbox |
| 17:09.18 | infobot | somebody said trixbox was a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org/. We do not recommend using it. |
| 17:09.20 | Killabeez | digium ? |
| 17:09.22 | seanbright | asteristnow is good |
| 17:09.27 | seanbright | asterisknow, rather. |
| 17:09.29 | seanbright | Qwell made it |
| 17:09.36 | seanbright | with his own two hands |
| 17:09.39 | Qwell | That Qwell guy.. he's a freaking genius. |
| 17:09.42 | gordonjcp | can anyone confirm whether I need to do anything other than set the port values in rtp.conf and restart asterisk, to set which ports I want it to use? |
| 17:09.43 | Killabeez | lol |
| 17:09.52 | jameswf | heard Qwell has 3 hands |
| 17:10.01 | seanbright | jameswf: that's not a hand |
| 17:10.05 | jameswf | eww |
| 17:10.09 | seanbright | and stop shaking it |
| 17:10.11 | seanbright | perv |
| 17:10.13 | gordonjcp | because, I've set up the port range I want in rtp.conf, restarted asterisk, and it sends outgoing RTP wherever the hell it likes |
| 17:10.45 | Killabeez | domain:8088/asterisk/static/config/cfgbasic.html -- this brings up a 'Asterisk Control panel' is this propreitary software with asterisk or a addon gui? |
| 17:11.32 | seanbright | asterisk gui |
| 17:11.38 | seanbright | i think. |
| 17:15.41 | Killabeez | ~book |
| 17:15.42 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 17:17.40 | LeddyHM | are there any limitations to "secret" in sip.conf being only alphanumeric or can their be special characters? |
| 17:18.12 | Killabeez | seanbright would you advise upgrading if I have a working install? [root@phones ~]# asterisk -V |
| 17:18.13 | Killabeez | Asterisk 1.4.12.1 |
| 17:18.41 | Qwell | Killabeez: do you like being vulnerable to exploits? |
| 17:18.54 | Killabeez | Qwell no but I don't like fixing broken stuff either |
| 17:19.08 | Qwell | you'll have plenty to fix if you don't upgrade. |
| 17:19.12 | *** join/#asterisk Paulius (n=Paulius@unaffiliated/paulius) |
| 17:19.24 | Killabeez | Qwell 1.6.1.0? |
| 17:19.32 | Paulius | So [TK]D-Fender and beek are going to want to kill me now. |
| 17:19.43 | Qwell | 1.4.latest would be fine, and easy to do |
| 17:20.00 | [TK]D-Fender | Paulius: What? for buying Cisco? |
| 17:20.08 | Paulius | [TK]D-Fender: Yeah. |
| 17:20.14 | [TK]D-Fender | Paulius: Your lost on functionality and cost |
| 17:20.16 | Paulius | [TK]D-Fender: I just ordered a brand new 7971-GE. |
| 17:20.23 | [TK]D-Fender | Paulius: Very sad |
| 17:20.25 | Paulius | haha |
| 17:20.37 | Paulius | Hey man, even people on the Asterisk forums say that it's an amazing phone. |
| 17:20.45 | Paulius | And that the newest SIP firmware works quite well. |
| 17:20.52 | *** join/#asterisk sah-work (n=Bawbatos@65.119.47.100) |
| 17:20.58 | [TK]D-Fender | Paulius: Forum idiots \o/ |
| 17:21.08 | Paulius | I'll investigate into getting the SmartNet contract to get access to the latest SIP firmware. |
| 17:21.14 | [TK]D-Fender | Paulius: You've been warned. Hapyy Uphil Fight. |
| 17:21.29 | [TK]D-Fender | Qwell: Care to do the honours on that one? ;) |
| 17:21.34 | bmoraca | Paulius: did they finally add the ability to do auto-answer based on your SIP header? |
| 17:21.56 | Paulius | bmoraca: No idea, but I don't want auto-answer anyways. |
| 17:22.00 | Qwell | ~stealing_from_cisco_by_not_understanding_what_smartnet_is |
| 17:22.01 | Qwell | stupid bot |
| 17:22.08 | Qwell | http://www.ntbox.com/cisco-openletter.html |
| 17:22.09 | Paulius | lol what |
| 17:22.14 | Qwell | Paulius: read and enjoy. |
| 17:22.29 | jameswf | ~smartnet |
| 17:22.35 | Paulius | Qwell: Thx |
| 17:23.09 | bmoraca | Paulius: my boss is entirely sold on Cisco. Thinks Cisco IP phones are God's gift to man. he just hasn't used Polycoms yet. Polycom phones are way better than Cisco phones. |
| 17:23.20 | Qwell | infobot: smartnet is <reply> SMARTnet isn't what you think it is. Read http://www.ntbox.com/cisco-openletter.html to get a better understanding of the licensing behind Cisco phones. |
| 17:23.21 | infobot | okay, Qwell |
| 17:23.25 | Qwell | ~smartnet |
| 17:23.26 | infobot | SMARTnet isn't what you think it is. Read http://www.ntbox.com/cisco-openletter.html to get a better understanding of the licensing behind Cisco phones. |
| 17:25.10 | bmoraca | anyone else hear that Manny Ramirez got suspended 50 games? |
| 17:25.12 | bmoraca | gloats |
| 17:25.44 | *** join/#asterisk tdg911 (n=tdg911@adsl-068-212-082-135.sip.msy.bellsouth.net) |
| 17:25.53 | *** join/#asterisk PoWeRKiLL (n=lironech@bzq-84-108-86-122.cablep.bezeqint.net) |
| 17:26.03 | Paulius | But Polycom doesn't have touch screen phones with web browsing. |
| 17:26.56 | [TK]D-Fender | Paulius: You could BUY a computer for the difference in cost. |
| 17:27.03 | Paulius | Oh yeah usre. |
| 17:27.05 | Paulius | *sure |
| 17:27.21 | Paulius | [TK]D-Fender: I think the color Polycom is more expensive than the Cisco actually. |
| 17:27.33 | bmoraca | Paulius: roughly the same |
| 17:27.35 | [TK]D-Fender | Paulius: Think? why start now? :) |
| 17:27.47 | Paulius | bmoraca: The new 7971ge cost $250 |
| 17:27.57 | Paulius | I've searched for the polycom and the color version was like $350 |
| 17:28.09 | [TK]D-Fender | Paulius: And you're about to run them illegally :) |
| 17:28.12 | bmoraca | why do you need a color phone anyway? |
| 17:28.32 | Paulius | bmoraca: I don't know, bling? |
| 17:28.35 | Paulius | [TK]D-Fender: Lab use. |
| 17:28.52 | [TK]D-Fender | Paulius: "Lab use" Nice load of BS :) |
| 17:29.21 | Paulius | Believe me, if I needed phones for actual use for a company or something I'd get the cheapest polycom or SIP phones out there. |
| 17:29.43 | Paulius | I wanted something to play with to learn the basics of vo-ip, CallManager, and Asterisk |
| 17:29.57 | [TK]D-Fender | Paulius: You have a CM? |
| 17:30.03 | beek | [TK]D-Fender: You're wasting your time and breath. You're trying to reason with a guy who jerks off to Cisco product brochures. |
| 17:30.17 | Paulius | lol |
| 17:30.24 | [TK]D-Fender | paulido you? |
| 17:30.28 | [TK]D-Fender | Paulius: do you? |
| 17:30.34 | [TK]D-Fender | (have a CM that is) |
| 17:30.35 | Paulius | Of course no. |
| 17:30.37 | Paulius | Oh |
| 17:30.43 | Paulius | I'll probably buy a router with CM soon,. |
| 17:30.53 | [TK]D-Fender | Paulius: wow.. "probably" |
| 17:30.59 | file | you mean CME |
| 17:31.18 | [TK]D-Fender | Paulius: Stop jerking to the wrong brochures! |
| 17:31.20 | [TK]D-Fender | :p |
| 17:31.22 | Paulius | lol |
| 17:31.34 | Qwell | "lab use" == "use" |
| 17:31.36 | Qwell | read that letter. |
| 17:31.53 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:35.09 | Witch_Doc | anyone here using either sip2sip.info or fonosip.com? |
| 17:35.20 | *** join/#asterisk lucasb (n=lucasb@s154-5-252-231.bc.hsia.telus.net) |
| 17:37.04 | [TK]D-Fender | Witch_Doc: I've heard the latter mentioned once or twice.... tops |
| 17:37.22 | Paulius | [TK]D-Fender: What trunking services do you recommend? |
| 17:37.39 | [TK]D-Fender | ~itsplist-us |
| 17:37.39 | infobot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 17:37.46 | Paulius | There's tons of these out there? How does it work, do most of them just resell from some popular wholesaler? |
| 17:38.56 | [TK]D-Fender | Paulius: Or you could jsut go with Vonage or try to hack a MagicJack account. then you'd be the shizn1t y0! |
| 17:39.03 | Paulius | ... |
| 17:39.21 | Paulius | Your maturity surprises me. |
| 17:39.25 | [TK]D-Fender | </mock> |
| 17:40.16 | [TK]D-Fender | Paulius: Whatever it takes to penetrate that seemingly impermeable cranium of yours... |
| 17:40.56 | nkohh | oh shit, he's busting out the big words now. |
| 17:40.58 | nkohh | and grammar. |
| 17:41.01 | nkohh | look out |
| 17:41.37 | [TK]D-Fender | gathers the Grammar Rangers for another epic battle |
| 17:41.49 | [TK]D-Fender | CHARGE!!!!!!!!!! |
| 17:41.54 | [TK]D-Fender | grabs his Visa |
| 17:45.37 | *** join/#asterisk lanning (n=lanning@nat/yahoo/x-56dccd2dae08294e) |
| 17:46.26 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 17:48.52 | nkohh | learned on his first day to not antagonize [TK]D-Fender |
| 17:49.57 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 17:50.40 | Katty | Qwell: i found my new money maker. |
| 17:50.50 | Katty | Qwell: 3 stacks of fish, 10 spices = 100g |
| 17:51.13 | Katty | Qwell: perhaps profitable. we'll see. |
| 17:51.19 | Katty | Qwell: fish feast is no longer soulbound :> |
| 17:53.06 | *** join/#asterisk ingenius (n=alektro@69.90.72.173) |
| 17:53.17 | *** join/#asterisk Mw3 (i=mw3@mw3.hu) |
| 17:53.43 | [TK]D-Fender | nkohh: You learn quickly Padawan :) |
| 17:54.13 | Ziaeon | I recompiled my kernel recently, and it broke conference bridging. Strange, because I used the same config as the running kernel. I grepped my logs and notice meetme saying: error "Failed to create pseudo device" etc. Any idea what I might be missing in my kernel? |
| 17:54.37 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 17:54.46 | *** join/#asterisk infernixx (i=nix@unaffiliated/infernix) |
| 17:55.02 | [TK]D-Fender | Ziaeon: If you rebuilt your kernel you need to rebuild Zaptel/DAHDI |
| 17:56.00 | *** part/#asterisk mohawk (n=ross@host217-40-110-153.in-addr.btopenworld.com) |
| 17:56.17 | *** join/#asterisk PoWeRKiLL (n=lironech@CBL217-132-127-130.bb.netvision.net.il) |
| 17:56.22 | Ziaeon | [TK]D-Fender: alright, thanks. |
| 17:56.45 | nkohh | i like the name dahdi better |
| 17:56.53 | nkohh | sounds like part of a snoop dogg song when you pronounce it |
| 17:57.19 | nkohh | lodi dahdi, we like to party |
| 17:57.21 | [TK]D-Fender | nkohh: Digium could let fonality run away with all the gayest sounding names :) |
| 17:57.32 | nkohh | lol |
| 17:57.45 | wilsonj | lahdi dahdi we likes to party :) |
| 17:58.01 | wilsonj | we don't cause trouble, we don't bother nobody |
| 17:58.11 | [TK]D-Fender | nkohh: You'd swear there is a force in their ranks thats trying to circumvent corporate respectability |
| 18:00.18 | gordonjcp | gah, this is frustrating |
| 18:03.49 | *** join/#asterisk drudge` (i=anonymou@unaffiliated/drudge/x-837452) |
| 18:05.23 | ariel_ | What is in a Name.... Zaptel/Dahdi. Well my vote is for Zaptel. For over 6 years I have used zaptel.... I still can't get used to Dahdi (Much less say it correctly). |
| 18:05.54 | gordonjcp | can anyone confirm whether or not I need to do any more than set the rtp port range in rtp.conf and restart asterisk, to tell it which rtp ports to use? |
| 18:06.10 | gordonjcp | because it seems to cheerfully ignore that and use whatever port it feels like |
| 18:06.16 | tzafrir_laptop | ariel_, tell that to those guys selling Zaptel cards over at zaptel.com |
| 18:08.23 | seanbright | dah-dee |
| 18:08.28 | ariel_ | tzafrir_laptop: I understand the reason and why it was done. Just hard to say and get used to it.... |
| 18:08.28 | seanbright | i don't see how that's so hard |
| 18:10.21 | Qwell | sed -i -e 's/zaptel/dahdi/g' ~/.brainrc |
| 18:10.22 | Qwell | done and done |
| 18:11.47 | eppigy | huggles Katty |
| 18:11.52 | gordonjcp | Qwell: heh |
| 18:11.58 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 18:12.08 | gordonjcp | well, I'm getting exactly *nowhere* with this |
| 18:12.20 | gordonjcp | time to do something more productive with what's left of the day |
| 18:14.06 | *** join/#asterisk Brixius (n=Brixius@PDN-VBA.OnvoyInc.fw.onvoy.net) |
| 18:16.42 | Brixius | this is somewhat of a simple question that I should be able to answer, but can't seem to find on voip-info.org. I want to do a silent dial(ie no ringing tone to caller) what option should I pass to dial() |
| 18:16.57 | Qwell | Brixius: none. it's phone dependent. |
| 18:19.04 | KyleK | m: Provide Music on Hold to the calling party until the called channel answers. This is mutually exclusive with option 'r', obviously. Use m(class) to specify a class for the music on hold. |
| 18:19.26 | KyleK | Dial(Someone|m(silence)) |
| 18:19.35 | seb- | [TK]D-Fender: it isn't clear all modules i need to load to make sip calls....(I'm trying to lock down modules.conf) |
| 18:19.35 | Brixius | seems like there should be a way, I'm answering the call, playing a wav file, then passing the call onto another cti app for further processing, It's that outbound dial from asterisk I want to be silent. |
| 18:19.49 | Brixius | ok, I'll look into that. |
| 18:20.34 | [TK]D-Fender | seb-: I told you what to lock down. Don't go psycho or you're going to waste a lot of time for nothing |
| 18:22.06 | seb- | [TK]D-Fender: sorry...i must have misunderstood what you said yesterday then....i think you said to no load anything in modules.conf right...i did that but then sip calls wouldn't work |
| 18:22.39 | seb- | [TK]D-Fender: so i tried loading chan_sip.so and app_conference.so and then i needed one more but it still crashed |
| 18:22.52 | seb- | [TK]D-Fender: (res_features.so) |
| 18:24.03 | [TK]D-Fender | seb-: and the PBX core, and other dialplan apps. Dial, GotoIf, etc... holy crap stop being a nut about this |
| 18:24.27 | [TK]D-Fender | seb-: Just noload CHANNEL DRIVER modules you don't need and firewall AMI. |
| 18:25.09 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 18:25.56 | seb- | [TK]D-Fender: ah ok |
| 18:26.11 | *** join/#asterisk PoWeRKiLL (n=lironech@bzq-79-176-34-195.red.bezeqint.net) |
| 18:29.15 | *** join/#asterisk PoWeRKiLL (n=lironech@bzq-79-176-34-195.red.bezeqint.net) |
| 18:30.44 | gordonjcp | how can I test if rtp.conf is even being loaded by asterisk? |
| 18:32.25 | *** join/#asterisk SwK (n=SwK@freeswitch/developer/swk) |
| 18:32.43 | gordonjcp | okay, asterisk -vv seems to get that |
| 18:33.01 | gordonjcp | <PROTECTED> |
| 18:33.01 | gordonjcp | <PROTECTED> |
| 18:33.21 | gordonjcp | so why doesn't it use that? |
| 18:33.55 | gordonjcp | do I need to be voiced in this channel or something? |
| 18:34.04 | file | ...no |
| 18:34.07 | gordonjcp | okay |
| 18:34.21 | gordonjcp | finally someone appears to respond to something I say |
| 18:34.26 | gordonjcp | ;-) |
| 18:34.31 | file | but you do know those control local RTP ports only, aye? |
| 18:34.59 | gordonjcp | file: one would assume then that it wouldn't try to send an rtp stream out on a port outside that range? |
| 18:35.24 | file | it would send RTP packets to the IP address and port that the remote device told it to |
| 18:35.33 | file | the port could be outside the range specified in rtp.conf |
| 18:35.55 | gordonjcp | hrm |
| 18:36.01 | file | the *local* port that RTP is received on would be inside that range |
| 18:36.14 | gordonjcp | okay |
| 18:37.07 | *** join/#asterisk voxter (n=voxter@190.241.15.217) |
| 18:39.15 | gordonjcp | so basically I have to open all possible UDP ports on my firewall, "just in case" |
| 18:39.26 | file | well that depends on the direction you are talking about |
| 18:39.32 | gordonjcp | I have a sneaking suspicion that this is going to break stuff |
| 18:39.54 | file | you might be sending RTP to any IP address and any port |
| 18:40.10 | file | you will receive RTP packets on the port range you specify in rtp.conf |
| 18:40.24 | gordonjcp | yeah, and that's fine |
| 18:40.52 | file | if you are firewalling your outbound traffic, then yeah... unless you have total control over all devices involved and the network then you don't really know where it will go |
| 18:41.02 | gordonjcp | I have a port range set in rtp.conf, the same port range forwarded to the asterisk box on a nat router |
| 18:41.06 | gordonjcp | this worked before |
| 18:41.22 | file | okay so go through steps to isolate the issue |
| 18:41.39 | file | the call is up but you get no audio from the audio side, but they hear you? |
| 18:41.51 | gordonjcp | no |
| 18:41.56 | gordonjcp | I can hear the remote end |
| 18:42.15 | KyleK | so that means your port forwards work, maybe thiers doesn't? |
| 18:43.06 | gordonjcp | it's sipgate, I'm guessing I don't need to worry about them too much |
| 18:44.12 | file | okay, do an rtp debug |
| 18:44.16 | file | see where the traffic is coming from |
| 18:44.22 | file | and see where you are sending to |
| 18:44.38 | *** join/#asterisk simond (n=simon@syria.uc.org) |
| 18:45.11 | gordonjcp | Sent RTP packet to 217.10.79.30:19236 |
| 18:45.27 | simond | when a translation path fails (i.e. out of g729 licenses), is there some way to play back a g729 encoded file before the call is disconnected? |
| 18:46.15 | file | gordonjcp: and Got RTP packet from? |
| 18:46.36 | *** join/#asterisk jmodigb (i=daemon@65-119-213-34.dia.static.qwest.net) |
| 18:47.05 | gordonjcp | that's interesting, it's stopped showing me "got rtp packet from" |
| 18:49.27 | [TK]D-Fender | simond: Nope. |
| 18:51.32 | seb- | [TK]D-Fender: ever seen "Starting Asterisk PBX: Unable to setuid to 106 (asterisk)" |
| 18:51.35 | seb- | ? |
| 18:51.46 | seb- | [TK]D-Fender: i get that when i try to start * w/ init.d script |
| 18:53.18 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 18:54.22 | simond | [TK]D-Fender: any idea where that sudden-death action takes place so that I might change the behavior? |
| 18:56.29 | [TK]D-Fender | simond: Nope |
| 18:56.54 | [TK]D-Fender | seb-: Who've you been running * as all this time? |
| 18:57.04 | seb- | [TK]D-Fender: root :) |
| 18:57.25 | seb- | [TK]D-Fender: i've just done it on command line w/ asterisk -cvvv |
| 18:57.33 | [TK]D-Fender | seb-: Odds are you didn't change the owners of appropriate files/devices following this |
| 18:57.40 | *** join/#asterisk jeffspeff (n=jeffspef@c-98-240-118-231.hsd1.ky.comcast.net) |
| 18:57.46 | seb- | [TK]D-Fender: now that you kindly showed me how to lock it down i was finally going to start it properly |
| 18:57.58 | [TK]D-Fender | seb-: .... |
| 18:57.59 | [TK]D-Fender | ~book |
| 18:58.00 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 18:58.04 | [TK]D-Fender | seb-: there is a chapter for this. |
| 18:58.06 | [TK]D-Fender | and.. |
| 18:58.09 | [TK]D-Fender | ~asterisk-non-root |
| 18:58.09 | infobot | [~asterisk-non-root] Running * as non-root is covered in chapter 13 of Asterisk:The Future of Telephony 2nd Edition (~book) and an article written by Leif Madsen at http://www.taug.ca/node/115 |
| 19:01.29 | *** join/#asterisk securevoip (n=securevo@173.15.197.73) |
| 19:02.08 | securevoip | i have actually simplified things and video still doesn't work... |
| 19:02.19 | securevoip | see any typos? http://pastebin.com/m71225ab9. using x-lite on both ends. i can do a direct IP call from x-lite client to x-lite client and all works well; put asterisk in the middle and it doesn't work. |
| 19:04.02 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 19:07.22 | *** join/#asterisk PoWeRKiLL (n=lironech@bzq-79-176-34-195.red.bezeqint.net) |
| 19:09.00 | *** join/#asterisk UQlev (n=yuriy@91.184.221.31) |
| 19:13.27 | seb- | [TK]D-Fender: looks like ubuntu package is b0rked...i tried doing a clean reinstall and still same error...PLEASE tell me if I install from source * 1.4 will still do that nice setuid security thing |
| 19:13.58 | *** join/#asterisk apocn (n=apo@unaffiliated/apocn) |
| 19:14.06 | [TK]D-Fender | prints the book onto 120lb bond paper, rolls it into a bat and starts swatting at seb- |
| 19:14.31 | Ziaeon | Is there a way to customize what fields asterisk keeps in its sql cdr? |
| 19:14.39 | Ziaeon | the cdr conf files only talk about csv |
| 19:14.45 | *** join/#asterisk Pazzo (n=ugelt@reserved-225136.rol.raiffeisen.net) |
| 19:14.48 | [TK]D-Fender | seb-: Ubuntu needs a few special touches to work, and the way it updates kernels would wreak havoc on Zaptel/DAHDI |
| 19:15.00 | [TK]D-Fender | seb-: For that you might be better off with distro packages |
| 19:15.21 | apocn | is it possible to move a channel from one queue to the other on the fly? (I tried using the Asterisk Manager 'Action: Redirect') but its closing the channel. Any hints? |
| 19:17.15 | KyleK | seb-: at worst you script sudo -u asterisk /usr/bin/asterisk |
| 19:20.23 | *** join/#asterisk PoWeRKiLL (n=lironech@bzq-84-110-213-221.red.bezeqint.net) |
| 19:21.13 | [TK]D-Fender | apocn: If its dropping them, odd are you're doing it wrong. |
| 19:22.16 | seb- | [TK]D-Fender: i don't use zaptel or dahdi |
| 19:22.39 | Ziaeon | seb-: it's used internally for some stuff, meetme etc |
| 19:22.42 | Ziaeon | (as far as I can tell) |
| 19:22.55 | apocn | [TK]D-Fender: oOps I didn't see the logs before... (Unable to join queue) so that means I have another issue, nothing to do with the redirect. |
| 19:22.58 | apocn | Thanks [TK]D-Fender |
| 19:23.07 | seb- | Ziaeon: yes..that's why i dug up app_conference which is a meetme replacement that doesn't need zaptel/dahdi |
| 19:28.26 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 19:32.13 | *** join/#asterisk theodred (n=nohost@205.207.102.154) |
| 19:37.39 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
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| 19:56.53 | Davidf88 | guys someone could help me pleae? |
| 19:58.52 | KyleK | dunno, i only know some stuff |
| 19:59.38 | *** join/#asterisk _Raptor_ (i=raptorbl@andariel.informatik.uni-erlangen.de) |
| 19:59.41 | [TK]D-Fender | I know other stuff |
| 19:59.47 | Davidf88 | hmm asterisk now, just installed but a lib is missing |
| 19:59.47 | [TK]D-Fender | And some things too |
| 19:59.51 | Davidf88 | libsqlite3 |
| 19:59.59 | Davidf88 | and also the web interface doesn't work |
| 20:00.42 | [TK]D-Fender | Davidf88: GUI's are supported in their own channel, not here |
| 20:00.56 | [TK]D-Fender | Davidf88: As for a missing lib, check with your distro's channel |
| 20:01.02 | KyleK | sounds like the install didn't work properly anyways :-/ |
| 20:01.09 | Davidf88 | is was the pre insalled iso cd |
| 20:01.12 | *** join/#asterisk Kisu (n=k@daniel1117.broker.freenet6.net) |
| 20:01.30 | KyleK | pre installed as in a livecd? |
| 20:02.07 | Davidf88 | yeah, live cd install |
| 20:02.48 | Qwell | Nothing uses libsqlite in AsteriskNOW |
| 20:03.27 | Davidf88 | so restarting httpd shouldn't need libsqlite then? |
| 20:03.39 | Qwell | Nope. |
| 20:03.50 | *** join/#asterisk kash_ (n=ptx0@2001:5c0:1000:a:0:0:0:133) |
| 20:03.51 | Qwell | what other repositories did you install? |
| 20:03.55 | *** join/#asterisk leif[mobile] (n=leifmads@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:03.55 | *** mode/#asterisk [+o leif[mobile]] by ChanServ |
| 20:05.23 | Davidf88 | I didn't install any other repositories |
| 20:05.28 | Davidf88 | and yum doesn't work |
| 20:05.32 | Davidf88 | cause libsqlite is missing |
| 20:05.54 | Qwell | What distro (and version) did you install? |
| 20:07.30 | [TK]D-Fender | Davidf88: Why does yum not work jsut because you're missing a lib? which one do you THINK you should be grabbing? |
| 20:08.18 | eppigy | you are a disgrace to daves everywhere |
| 20:08.29 | Davidf88 | the error, is python error, error leading to this problem was : libsqlite3.so.0: Cannot open shared object |
| 20:08.50 | Qwell | Davidf88: focus. answer my questions please.. |
| 20:09.14 | Davidf88 | what do you mean which one? |
| 20:09.20 | Qwell | What distro (and version) did you install? |
| 20:09.42 | *** join/#asterisk j_kroon (n=jkroon@dsl-240-131-22.telkomadsl.co.za) |
| 20:10.11 | Davidf88 | its asteriskNOW 1.5.0-i386 |
| 20:10.22 | [TK]D-Fender | Davidf88: Show us your yum attempt |
| 20:11.47 | Davidf88 | [root@asterisk tmp]# yum udate |
| 20:11.47 | Davidf88 | There was a problem importing one of the Python modules |
| 20:11.47 | Davidf88 | required to run yum. The error leading to this problem was: |
| 20:11.47 | Davidf88 | <PROTECTED> |
| 20:11.47 | Davidf88 | Please install a package which provides this module, or |
| 20:11.49 | Davidf88 | verify that the module is installed correctly. |
| 20:11.51 | Davidf88 | It's possible that the above module doesn't match the |
| 20:11.53 | Davidf88 | current version of Python, which is: |
| 20:11.55 | Davidf88 | 2.4.3 (#1, Jan 21 2009, 01:10:13) |
| 20:11.57 | Davidf88 | [GCC 4.1.2 20071124 (Red Hat 4.1.2-42)] |
| 20:11.59 | Davidf88 | If you cannot solve this problem yourself, please go to |
| 20:12.01 | Davidf88 | the yum faq at: |
| 20:12.02 | Deeewayne | O.O |
| 20:12.03 | Davidf88 | <PROTECTED> |
| 20:12.05 | Davidf88 | <PROTECTED> |
| 20:12.50 | Qwell | Davidf88: /j #asterisknow, please |
| 20:13.24 | eppigy | why are you trying to yum update anyway |
| 20:14.13 | Davidf88 | basically to resolve any dependencies, and work out why the web interface is broken |
| 20:14.18 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 20:15.42 | eppigy | well do not use update |
| 20:15.47 | eppigy | to resolve dependancies |
| 20:15.57 | eppigy | install the missing depens |
| 20:16.37 | eppigy | you can use "yum search <package name>" |
| 20:16.44 | eppigy | or as an exmaple to the above |
| 20:16.54 | eppigy | "yum whatprovides libsqlite3.so.0" |
| 20:17.03 | eppigy | to see what would provide that lib |
| 20:17.11 | eppigy | which package |
| 20:18.55 | philippel | question: if I have a hint made of two device (let's say) SIP/210&SIP/220, one device is Idle and the other is onHold, what should the hint show? Idle or onHold? |
| 20:19.03 | Qwell | eppigy: how's he gonna run yum if yum is broken? :p |
| 20:19.16 | [TK]D-Fender | philippel: OnHold. the greater of the two |
| 20:19.41 | philippel | [TK]D-Fender so the fact that it seems to show busy means it is a bug? (cause i agree with you completely) |
| 20:19.54 | philippel | oops not busy |
| 20:19.57 | philippel | it shows Idle |
| 20:20.18 | [TK]D-Fender | philippel: or you configured your peer wrong. |
| 20:20.32 | [TK]D-Fender | philippel: Which is prettt common. |
| 20:20.38 | philippel | what might I check in the peer configuration? |
| 20:21.40 | Corydon76-dig | philippel: call-limit -- do you have one? |
| 20:21.49 | philippel | [TK]D-Fender cause here's what I'm seeing: Idle & onHold = idle (when it should show onHold, and idle & busy = busy which is what I would exect |
| 20:21.58 | philippel | I should but let me confirm |
| 20:22.13 | [TK]D-Fender | philippel: Show us your config. |
| 20:22.27 | philippel | yes I do, I'll paste bin the config |
| 20:24.57 | eppigy | Qwell: 8[] |
| 20:25.29 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 20:26.20 | philippel | [TK]D-Fender it's a big config so I edited out except the two extensions and general section: http://pastebin.ca/1415673 |
| 20:26.42 | philippel | Asterisk 1.4.21.1 |
| 20:28.05 | *** join/#asterisk Magicblaze0071 (n=sony@CS2247.cs.fsu.edu) |
| 20:28.12 | Corydon76-dig | [TK]D-Fender: They all have to be onhold, for the devstate to show onhold |
| 20:29.07 | philippel | Corydon76-dig I would venture to suggest that is not what one would expect - it is inconcsistent with behaviour like busy |
| 20:29.28 | philippel | if I have a hint of Idle&Busy I would expect that to show busy and it does |
| 20:29.36 | [TK]D-Fender | philippel: Yup, thought so |
| 20:29.48 | [TK]D-Fender | philippel: type=friend <--- needs to be PEER. |
| 20:29.50 | Corydon76-dig | philippel: yes, but if you're onhold, then you're technically available |
| 20:29.56 | philippel | using that same 'argument' I would expect onHold & idle to be onHold |
| 20:30.38 | philippel | Corydon76-dig but one would expect the combined state to be onHold so that the logic can decide if you should be availble or not |
| 20:30.45 | Corydon76-dig | philippel: I'm not arguing; I'm simply telling you that's what the code is written to do presently |
| 20:30.53 | philippel | [TK]D-Fender I'll try friend and see what it does |
| 20:30.59 | Corydon76-dig | philippel: My dog is not in this fight |
| 20:31.37 | philippel | Corydon76-dig I appreciaite it - what's I'm trying to determine, should it be changed? e.g. is it a valid bug |
| 20:32.11 | [TK]D-Fender | BBIAB |
| 20:32.14 | Corydon76-dig | philippel: The proper place to discuss such things is the -dev list |
| 20:33.12 | philippel | Corydon76-dig ok - just to clarify then, it is returning the correct thing (I don't need to make other changes to test) and it's a matter of discussing if it should be changed or not on the dev list? |
| 20:33.40 | Corydon76-dig | philippel: correct |
| 20:33.58 | philippel | thanks |
| 20:34.33 | Corydon76-dig | I suspect it's the case that someone is preferably idle and we only return another status if they're really unavailable |
| 20:37.28 | philippel | well when you combine Unavailable + Idle (as in not registered) you get Idle when they really are not busy, though I would conjecture that Idle is the preferable return code in that case because Unavailable really means they are not there at all to say |
| 20:38.38 | philippel | but onHold is a state 'one step above' Idle, I would think most pepple would expect it to return onHold in the proposed instance as [TK]D-Fender had |
| 20:39.46 | philippel | or similarlly (and it is what is returned) when ringing + idle, you get ringing |
| 20:41.32 | *** join/#asterisk jstrom (i=johan@core.stromnet.se) |
| 20:43.05 | jstrom | i got a cisco 7940 which doesnt seem to handle international characters in callerid(name) very well.. is there some smart way to, when my dialplan does something like Dial(SIP/mycisco..)&Dial(SIP/myxlite), strip these chars when calling the cisco but not the xlite? |
| 20:44.06 | jstrom | shorter version: i want to Dial() multiple phones, with "filtered" CALLERID(name) on one specific.. |
| 20:44.09 | jstrom | any neat hacks? :) |
| 20:45.36 | philippel | jstrom dial the Cisco phone through a local channel in a context that will filter as needed and then send to the actual device |
| 20:46.02 | philippel | Dial(SIP/212&Local/223@strip-my-cid) |
| 20:46.12 | philippel | where 223 is your cisco and 212 something else |
| 20:49.17 | jstrom | ah, yes :) thank you! |
| 20:50.20 | jstrom | hm is there builtins for modifying strings like that? ie replace ö with o |
| 20:50.40 | philippel | regex |
| 20:50.49 | jstrom | gret :) |
| 20:50.50 | jstrom | great |
| 20:50.54 | jstrom | checks docs |
| 20:51.25 | philippel | hmm - regex for comares, come to think of it, I don't recall off hand if they can manipulate |
| 20:51.48 | *** join/#asterisk tleuthauser (i=user@travis.broadbandip.net) |
| 20:52.05 | jstrom | hm, seems to work only for matching? |
| 20:54.44 | tleuthauser | I am having problems receiving faxes with a HP Color LaserJet 2320Fxi connected to a Grandstream HT502. Sending works fine. Has anyone run into similar problems? |
| 20:55.29 | nkohh | this is off topic, but does anyone know where those places that list every telephone number and their carrier/other info get their information? is there a database somewhere? I'd like to be able to use something like that to integrate for some stuff I'm doing with Asterisk. and grabbing it from a mysql or something would be much easier than scraping webpages |
| 20:56.35 | jeff | nkohh: go visit telcodata.us and you can pay for a reasonable subscription in your preferred format |
| 20:56.40 | *** join/#asterisk telnettech (i=telnette@gw.percipia.com) |
| 20:56.52 | *** join/#asterisk generalhan (n=asd@about/windows/staff/generalhan) |
| 20:56.56 | nkohh | jeff: interesting, thank you. |
| 20:58.06 | generalhan | hey all, i have an extension that i want to wait for the user to enter a few digits and then take their response and pass it to different context, what cmd do i need to look into ? |
| 21:00.20 | Qwell | nkohh: number portability makes those types of databases unreliable though |
| 21:00.43 | nkohh | Qwell: ahh, I can imagine. thanks. |
| 21:01.19 | KyleK | !npa 651 644 |
| 21:01.24 | beek | generalhan: Read* |
| 21:01.27 | KyleK | aww |
| 21:01.50 | generalhan | beek: wow, simple as that huh!? thanks ! |
| 21:01.58 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 21:02.27 | KyleK | 16516441452 is probably a US number right? I wonder if I could sign my canadian cellphone up for the american do not call :-/ |
| 21:02.30 | beek | generalhan: There are a couple of variations: core show applications like read |
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| 21:21.12 | *** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com) |
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| 21:33.46 | tleuthauser | is there a channel dealing primarily with T.38 in Asterisk? |
| 21:35.27 | *** part/#asterisk tleuthauser (i=user@travis.broadbandip.net) |
| 21:39.57 | *** part/#asterisk securevoip (n=securevo@173.15.197.73) |
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| 21:44.17 | jameswf | I <3 FREE KFC |
| 21:51.43 | Katty | what about free hugs? |
| 21:51.43 | *** part/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com) |
| 21:51.44 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 21:53.55 | VaGoNeTaS | is back from the dead. Gone: 3d 1h 35m 53s |
| 21:54.11 | VaGoNeTaS | i got an issue on one of our customers |
| 21:54.21 | VaGoNeTaS | is a small call center with 12 agents |
| 21:54.36 | VaGoNeTaS | sometimes they can talk but the customer on the line cant hear the agents |
| 21:54.42 | VaGoNeTaS | i was looking on the logs |
| 21:54.50 | VaGoNeTaS | and i found this |
| 21:54.55 | VaGoNeTaS | i think it can be this |
| 21:54.57 | VaGoNeTaS | [May 7 17:19:31] DEBUG[17383] chan_dahdi.c: Set option AUDIO MODE, value: ON(1) on DAHDI/1-1 |
| 21:54.57 | VaGoNeTaS | [May 7 17:19:31] DEBUG[17383] chan_dahdi.c: Not yet hungup... Calling hangup once with icause, and clearing call |
| 21:54.57 | VaGoNeTaS | [May 7 17:19:31] DEBUG[17383] chan_dahdi.c: Set option AUDIO MODE, value: OFF(0) on DAHDI/1-1 |
| 22:05.40 | *** join/#asterisk cesar_CR (n=cesar@201.195.239.11) |
| 22:10.07 | *** join/#asterisk Qwell (i=north@pdpc/sponsor/digium/Qwell) |
| 22:10.07 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 22:10.30 | *** join/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 22:10.30 | *** mode/#asterisk [+o russellb] by ChanServ |
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| 22:15.22 | VaGoNeTaS | anybody knows? |
| 22:18.17 | KavanS | anyone know why I would see the [May 7 15:09:11] NOTICE[3713]: chan_sip.c:13885 handle_request_invite: Call from '' to extension '95035551212' rejected because extension not found. |
| 22:25.47 | KavanS | can someone suggest an SIP proxy for debian? |
| 22:25.52 | philippel | question: If I set the FORWARD_CONTEXT blank Set(FORWARD_CONTEXT= ) to clear if from a previous value, will it be treated as not set? Here is the source code, I think it implies if it's blank it will be ignored: |
| 22:26.00 | philippel | <PROTECTED> |
| 22:26.00 | philippel | <PROTECTED> |
| 22:26.12 | Davidf88 | asterisk giving dhcpo out on an interface how? |
| 22:30.57 | VaGoNeTaS | does anyone knows |
| 22:31.08 | VaGoNeTaS | [May 7 17:19:31] DEBUG[17383] chan_dahdi.c: Set option AUDIO MODE, value: ON(1) on DAHDI/1-1 |
| 22:31.08 | VaGoNeTaS | [May 7 17:19:31] DEBUG[17383] chan_dahdi.c: Not yet hungup... Calling hangup once with icause, and clearing call |
| 22:31.08 | VaGoNeTaS | [May 7 17:19:31] DEBUG[17383] chan_dahdi.c: Set option AUDIO MODE, value: OFF(0) on DAHDI/1-1 |
| 22:46.14 | *** join/#asterisk saftsack (n=saftsack@p57924C8A.dip.t-dialin.net) |
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| 23:05.08 | KavanS | anyone know why I would see the [May 7 15:09:11] NOTICE[3713]: chan_sip.c:13885 handle_request_invite: Call from '' to extension '95035551212' rejected because extension not found. |
| 23:06.46 | *** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com) |
| 23:06.59 | *** join/#asterisk rob0 (i=rob0@sorry.nodns4.us) |
| 23:08.38 | *** join/#asterisk cb` (n=doug@97.90.19.127) |
| 23:08.42 | cb` | hey..what hte heck is WARNING[1102232496]: chan_zap.c:5903 do_monitor: Whoa.... |
| 23:09.03 | cb` | I'm owned but but found |
| 23:09.04 | cb` | lol |
| 23:12.01 | VaGoNeTaS | is away: Fell asleep on keyboard... <<eDK/VgN>> [ Logging, Page: On ] |
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| 23:43.51 | Witch_Doc | anyone know how to get a google voice invite? |
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