| 00:02.48 | _-Jon-_ | What does this message mean again? I've gotten it a millioin times before and can never remember the fix... [Aug 27 20:02:13] NOTICE[26791]: res_musiconhold.c:544 monmp3thread: Request to schedule in the past?!?! |
| 00:04.11 | *** join/#asterisk [xpl] (i=xpl@84.126.207.62.dyn.user.ono.com) |
| 00:05.29 | [TK]D-Fender | _-Jon-_: means * is having trouble syncing audio. make sure you have a zaptel timer installed |
| 00:06.28 | _-Jon-_ | Odd, zaptel doesn't seem to be installed correctly though I swear I had it working before |
| 00:14.22 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 00:19.36 | SplasPood | has anyone seen this before? Set(~~EXTEN~~=${EXTEN}) |
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| 01:39.44 | Chicago | Has anybody written any MUDs or other doorway games for Asterisk??? |
| 01:40.17 | Qwell | Chicago: there was a guy who speech enabled Zork |
| 01:40.27 | *** join/#asterisk mvanbaak_ (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 01:40.52 | Qwell | http://www.venturevoip.com/news.php?rssid=1513 |
| 01:41.33 | Chicago | Qwell: Awesome, that is just what I am looking for. |
| 01:41.43 | Chicago | Qwell: Any other games you know of? |
| 01:42.19 | Qwell | I started writing code to do something with my MUD, but eventually scrapped it due to lack of interest... |
| 01:43.00 | Qwell | I was going to have it ask the user for a phone # when they create a char, and then call it to verify |
| 01:43.01 | *** join/#asterisk ShaneAu (n=shane-wo@203.56.250.52) |
| 01:43.33 | Chicago | Qwell: That sounds fair... the old BBS systems would make you receive a call to create an account. |
| 01:43.47 | Chicago | Qwell: I would like to see trade wars or legend of the red dragon on Asterisk. |
| 01:43.57 | Qwell | hmm |
| 01:44.06 | Qwell | You could probably do lotrd |
| 01:44.40 | Chicago | Qwell: Trade wars would be especially neat because if the planet is under attack, people could be rang in nearby sectors to take their turn. |
| 01:44.54 | Qwell | don't know what trade wars is |
| 01:45.47 | Chicago | Qwell: Trade wars was out a few years before lord. http://en.wikipedia.org/wiki/Tradewars |
| 01:48.35 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 01:48.52 | Chicago | Qwell: Did you make the part work where the user is called for verification after they create an account? |
| 01:49.15 | Qwell | no, I got lazy and stopped caring pretty quickly :p |
| 01:49.26 | Chicago | hehe, that happens |
| 01:49.37 | Chicago | If you want it bad enough you'll probably get around to it eventually. |
| 01:49.54 | Qwell | I grew bored of MUDs |
| 01:50.33 | Qwell | it really shouldn't be too difficult though. it would be as easy as a manager originate, and a bit of dialplan |
| 01:51.05 | Qwell | maybe put the pfile into some pending/ directory on creation, and then move it in after account verification |
| 01:51.13 | *** join/#asterisk Levonk (n=lk@adsl-75-62-142-94.dsl.lsan03.sbcglobal.net) |
| 01:51.35 | Chicago | Qwell: Something like that |
| 01:51.40 | Chicago | Last night in this channel I asked if anyone was using Windows Messenger SIP clients. |
| 01:52.21 | Chicago | After working on it for a bit yesterday and reading the SIP headers from asterisk cli... I know the Windows Messenger Client reports its agent as 'RTC/1.3.5470 (Messenger 5.1.0715)' |
| 01:53.07 | Chicago | Qwell: Do you know much about using the Windows Messenger as a SIP client, to help me troubleshoot some bugs on my end? |
| 01:53.16 | Qwell | nope.. |
| 01:53.51 | *** join/#asterisk SwK (n=SwK@freeswitch/developer/swk) |
| 01:55.42 | Chicago | Well, using Windows Messenger 5.1 I was able to get basic connectivity and talk back and forth. |
| 01:55.43 | *** join/#asterisk amessina (n=amessina@h-66-166-108-202.chcgilgm.covad.net) |
| 01:56.09 | Chicago | There is a registry modification I need to find/test which should enable a dialpad... otherwise there is no DTMF buttons built into Windows Messenger. |
| 01:56.20 | Chicago | Ask me why I want to use Windows for anything? |
| 02:15.29 | *** join/#asterisk rbd_ (n=rbd@adsl-074-229-183-112.sip.rmo.bellsouth.net) |
| 02:16.20 | rbd_ | hi guys...weird problem. when trying to start up asterisk, I am getting an error like "Unable to open pid file '/var/run/asterisk.pid': Permission denied" ...however astrundir in my /etc/asterisk/asterisk.conf is set to /var/run/asterisk ... /var/run/asterisk exists and has the correct permissions (asterisk:asterisk) |
| 02:16.28 | rbd_ | I did compile asterisk from source |
| 02:16.39 | rbd_ | but I didn't see an option for the run dir with ./configure ... |
| 02:17.45 | *** join/#asterisk Sargun (n=xbmodder@atarack/staff/sargun) |
| 02:18.56 | WimpMan | rbd_: Does the pid file exist? |
| 02:19.42 | rbd_ | WimpMan: no |
| 02:20.41 | WimpMan | Oh, I see, you don't get it where you want it. |
| 02:20.44 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.34.3) |
| 02:21.07 | WimpMan | Sorry, no idea. I didn't change any locations. |
| 02:21.32 | rbd_ | may be this init.d script I'm using...if I launch it as "asterisk -vvvc" it runs fine |
| 02:21.57 | brodiem | rbd_: you are running ast w/ a non-root user |
| 02:22.04 | WimpMan | Well that sound rather obvious then. |
| 02:22.06 | brodiem | mkdir /var/run/asterisk |
| 02:22.09 | brodiem | chown asterisk. /var/run/asterisk |
| 02:22.31 | brodiem | set your astrundir in asterisk.conf to the new dir |
| 02:23.10 | brodiem | and don't forget to change ownerships of other dirs that won't be writeable by default... /var/log/asterisk, /var/spool/asterisk, /var/lib/asterisk.. |
| 02:24.31 | brodiem | WimpMan: yes it makes it obvious when no system would allow other users but root to write to /var/run.. |
| 02:25.15 | WimpMan | Isn't it * itself changing user then? |
| 02:25.34 | WimpMan | * as user is still on my to-do list... |
| 02:26.04 | rbd_ | brodiem: yes that's all the case |
| 02:26.06 | brodiem | WimpMan: no, safe_asterisk is spawning the proc as the asterisk user |
| 02:26.16 | rbd_ | drwxrwxrwx 2 asterisk asterisk 40 2008-08-20 00:12 asterisk |
| 02:26.19 | rbd_ | even made it 777 to debug this |
| 02:26.53 | rbd_ | asterisk runs as the asterisk user...still for some reason it's trying to made a pid file in /var/run .... |
| 02:27.09 | brodiem | rbd_: did you set astrundir in asterisk.conf? |
| 02:27.55 | WimpMan | Does it set an alternate config dir? |
| 02:28.50 | brodiem | WimpMan: not sure what you're getting at |
| 02:28.55 | rbd_ | brodiem: yes. however I tried on another machine with an identical setup and it works fine. I did an md5sum on the asterisk binary and they are different. I will try reinstalling asterisk from source on the bad box and see if that fixes it |
| 02:29.36 | brodiem | I don't think that would have anything to do with it |
| 02:29.52 | rbd_ | yeah me neither...but at this point I don't know what else it could be.... :( |
| 02:29.53 | WimpMan | brodiem: I was thinking that maybe it doesn't read the config specifying the dir at all. |
| 02:31.22 | brodiem | rbd_: try launching asterisk with the "-U asterisk" flag from shell |
| 02:31.24 | brodiem | brb |
| 02:39.05 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 02:39.05 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 02:42.05 | jeev | oh great |
| 02:42.08 | jeev | it's leif ericson |
| 02:42.45 | lmadsen | the famous piliger |
| 02:42.52 | jeev | lol |
| 02:42.54 | lmadsen | pillager* |
| 02:43.00 | jeev | how's the new book coming along |
| 02:43.10 | lmadsen | almost started the outline |
| 02:43.17 | jeev | ok, dont forget to include me this time |
| 02:51.10 | Qwell | bribes lmadsen to be in the next book |
| 02:52.49 | Alton2 | hah, leif, hi |
| 02:53.24 | Alton2 | they allow canadians here? ;-) |
| 02:53.34 | Qwell | only deny |
| 02:54.06 | Qwell | I completely butchered that, and now I forgot what I was trying to say. |
| 02:54.11 | Qwell | ignore me |
| 02:54.50 | Alton2 | Leif probably doesn't remember me, we went back and forth a few times, a short Digium thing. |
| 02:55.29 | Alton2 | I think I'm very close to getting him to convert to slackware! :-) |
| 02:57.53 | *** join/#asterisk jameswf-home (n=james@ip72-223-0-183.ph.ph.cox.net) |
| 03:01.19 | lmadsen | slackware? ha! |
| 03:01.26 | lmadsen | if you believe that, you truly don't know me :) |
| 03:01.35 | lmadsen | Alton Moore? |
| 03:01.49 | lmadsen | remembers all his clients :) |
| 03:05.15 | *** join/#asterisk JenniferAkemi (n=akemi@dhcp-0-13-10-42-a4-3c.cpe.axion.ca) |
| 03:06.18 | brodiem | hey another slack user |
| 03:06.56 | WimpMan | too |
| 03:07.12 | Alton2 | yup, alton moore |
| 03:07.33 | *** join/#asterisk pcrane (n=pcrane@202.49.106.158) |
| 03:07.37 | Alton2 | The tide is taking hold! You are beginning to see the light! |
| 03:07.44 | leifmadsen | :) |
| 03:07.48 | leifmadsen | sorry... CentOS all the way |
| 03:07.50 | *** join/#asterisk luxxx77 (n=luxxx77@e176229022.adsl.alicedsl.de) |
| 03:07.50 | leifmadsen | :) |
| 03:07.55 | Alton2 | haha |
| 03:08.12 | Alton2 | I have cursed selinux more than a few times. You can take pride in that centos feature! |
| 03:08.19 | leifmadsen | I disable it :) |
| 03:08.28 | pcrane | I need a hand with some dialplan stuff |
| 03:08.31 | Alton2 | Yeah, have learned that and more. |
| 03:08.39 | pcrane | I've got a dialplan that looks like this: |
| 03:08.43 | leifmadsen | been years and I still don't know how to use it |
| 03:08.47 | leifmadsen | pcrane: use a pastebin |
| 03:08.51 | leifmadsen | if it is more than 3 lines |
| 03:08.57 | pcrane | it is exactly 3 lines |
| 03:08.58 | pcrane | ;) |
| 03:09.02 | brodiem | so, the new slack logo is cool :) |
| 03:09.05 | pcrane | v. simple dialplan |
| 03:09.12 | leifmadsen | :) |
| 03:09.17 | leifmadsen | pcrane: lets have at'r |
| 03:09.20 | pcrane | exten => s,1,Answer() |
| 03:09.20 | pcrane | exten => s,n,Beep() |
| 03:09.20 | pcrane | exten => s,n,Hangup() |
| 03:09.21 | pcrane | <PROTECTED> |
| 03:09.37 | pcrane | so, I've set a sip peer up so that it comes in to that context |
| 03:09.37 | Alton2 | it's an abomination! slackware displaying a graphic upon startup, heresy! |
| 03:09.41 | leifmadsen | I don't think Beep() is a dialplan application |
| 03:09.49 | pcrane | regardless... |
| 03:09.55 | leifmadsen | what is the problem? |
| 03:10.06 | leifmadsen | if the application isn't real, dialplan execution will stop |
| 03:10.16 | leifmadsen | and 's' does not mean "catch all" |
| 03:10.17 | pcrane | [Aug 28 15:10:03] NOTICE[1767]: chan_sip.c:14035 handle_request_invite: Call from '101' to extension '1' rejected because extension not found. |
| 03:10.23 | pcrane | hmm.. |
| 03:10.24 | leifmadsen | right |
| 03:10.33 | leifmadsen | you need to read the dialplan chapter of TFoT again |
| 03:10.35 | [TK]D-Fender | pcrane: "1" != "s" |
| 03:10.35 | leifmadsen | :) |
| 03:10.36 | Alton2 | yeah, I've wondered why s works in some contests and doesn't work in others. |
| 03:10.37 | pcrane | he |
| 03:10.42 | pcrane | yeah |
| 03:11.02 | leifmadsen | s is a special extension used by Macro()'s and analog Zaptel channels |
| 03:11.03 | [TK]D-Fender | ~stdextens |
| 03:11.04 | jbot | [~stdextens] The "s" Standard Extension : Where a call goes to when * does not know the destination of the call. Ex : Calls coming in on FXO ports (no DID), or from an ITSP that doesn't specify or where it was not set in the REGISTER line, or FXS port goes off-hook and "immediate=yes" in zapata.conf. "s" is also used to make IVRs & macros. |
| 03:11.06 | pcrane | thing is that I'm just trying to test that small part of a large complex time-based routing stuff (which I know works) |
| 03:11.07 | [TK]D-Fender | ^^^^ |
| 03:11.16 | pcrane | ok |
| 03:11.17 | leifmadsen | that is poorly worded |
| 03:11.28 | pcrane | because I dial 1, it knows where it should be going |
| 03:11.36 | [TK]D-Fender | pcrane: Thing is you're dialing a number and somehow magically thinking it'll hit an exten named "s" |
| 03:11.40 | leifmadsen | "Where a call goes to when * does not know the destination of the call" implies the 'i' extension kinda |
| 03:11.40 | pcrane | so, it looks for a exten => 1,1,something |
| 03:11.50 | leifmadsen | pcrane: yes, exactly |
| 03:11.50 | pcrane | ok |
| 03:11.54 | leifmadsen | or a pattern match |
| 03:11.59 | pcrane | so, how can I test that s thing? |
| 03:12.00 | leifmadsen | exten => _X,1,NoOp() |
| 03:12.01 | [TK]D-Fender | pcrane: "it knows"? No, its MATCHES what you dial and "1" sure isn't a LETTER |
| 03:12.11 | leifmadsen | pcrane: have an analog line come into asterisk |
| 03:12.12 | pcrane | so I dial s? |
| 03:12.13 | pcrane | :p |
| 03:12.18 | [TK]D-Fender | pcrane: YES. |
| 03:12.19 | Alton2 | oh, the s works sometimes.... |
| 03:12.20 | pcrane | heh |
| 03:12.21 | leifmadsen | [TK]D-Fender: can you turn down the antagonism please |
| 03:12.22 | pcrane | dial s |
| 03:12.24 | pcrane | and it works |
| 03:12.25 | pcrane | lol |
| 03:12.27 | [TK]D-Fender | pcrane: or change the PATTERN |
| 03:12.32 | pcrane | go figure |
| 03:12.33 | Alton2 | I've learned already, it's his trademark. :-) |
| 03:12.43 | leifmadsen | seriously, it's totally unnecessary |
| 03:12.49 | pcrane | I don't want to change that s thing |
| 03:12.52 | [TK]D-Fender | pcrane: If I tell it to only accept odd numbers should it not bitch at me when I give it an even one too? |
| 03:13.09 | pcrane | it's part of a larger more complex larger dialplan |
| 03:13.38 | leifmadsen | just remember 's' is not a catch all |
| 03:13.46 | pcrane | mmm |
| 03:13.49 | pcrane | I know that now |
| 03:13.54 | Alton2 | I'm still not clear when the darned things works. |
| 03:13.59 | leifmadsen | exten => _[a-zA-Z0-9]!,1,NoOp() would be a catch all |
| 03:14.02 | pcrane | I've only seen examples where it is used as a catch-a;; |
| 03:14.03 | Alton2 | I do use it, but I'm still not clear. :-) |
| 03:14.09 | pcrane | all* |
| 03:14.13 | leifmadsen | Alton2: it is used by analog lines mostly |
| 03:14.24 | leifmadsen | Alton2: or when you call a Macro() |
| 03:14.26 | Alton2 | hmm, I have analog lines here, that's why it must work so well! |
| 03:14.31 | [TK]D-Fender | leifmadsen: ALMOST a catch-all |
| 03:14.36 | leifmadsen | exten => _XXX,1,Macro(foobar,arg1) |
| 03:14.55 | [TK]D-Fender | leifmadsen: And we won't go into why that is dangerous ;) |
| 03:14.57 | leifmadsen | [macro-foobar] |
| 03:14.57 | leifmadsen | exten => s,1,NoOp(Value of ${ARG1}) |
| 03:14.58 | Alton2 | Ok, I'll try to keep the analog thing in mind. |
| 03:15.14 | leifmadsen | [TK]D-Fender: I'm just saying you can do a sort of catch all, but 's' is not how to do it |
| 03:15.27 | [TK]D-Fender | Alton2: Or when you register with an ITSP and don't tell them what exten to dial. |
| 03:15.29 | leifmadsen | Alton2: otherwise, most other channel types request a specific extension |
| 03:15.33 | pcrane | _.! would probably be better |
| 03:15.39 | leifmadsen | nope |
| 03:15.45 | [TK]D-Fender | leifmadsen: Yes, "s" is DID-challenged :) |
| 03:15.54 | [TK]D-Fender | pcrane: Close. |
| 03:15.57 | leifmadsen | that ! actually is unnecessary |
| 03:15.58 | Alton2 | I have learned to just use catch-all pattern matching instead of s. |
| 03:16.04 | pcrane | _!? |
| 03:16.06 | [TK]D-Fender | leifmadsen: CLOSE |
| 03:16.10 | leifmadsen | who can tell me the difference between _X. and _X! ? |
| 03:16.18 | [TK]D-Fender | Dammit you're BOTH still wrong... |
| 03:16.24 | [TK]D-Fender | leifmadsen: I can :) |
| 03:16.33 | leifmadsen | wtf are you blathering about? |
| 03:16.37 | WimpMan | too :-) |
| 03:17.01 | leifmadsen | I am not saying _X. or _X! is a catch all |
| 03:17.08 | WimpMan | There's only a difference when using overlap dialling. |
| 03:17.09 | leifmadsen | I'm asking if anyone knows the difference |
| 03:17.15 | pcrane | _X! will match the first occourance of X straight away without waiting for the remaining digits |
| 03:17.16 | [TK]D-Fender | leifmadsen: exten => _!,1,NoOp(***I** am a catch-all) |
| 03:17.20 | pcrane | _X. will wait |
| 03:17.23 | Alton2 | one is matched before the other |
| 03:17.36 | Alton2 | hah, I was kinda right. |
| 03:17.41 | leifmadsen | . means match *1* or more digits... ! means match *0* or more digits |
| 03:17.53 | leifmadsen | thus _X. requires at least 2 digits to be dialed to match |
| 03:17.54 | [TK]D-Fender | leifmadsen: clsoe again :) |
| 03:18.12 | [TK]D-Fender | leifmadsen: "." means 1 or more CHARATERS. Not Necessarily DIGITS :) |
| 03:18.35 | Alton2 | ok, fine, splitting hairs |
| 03:18.44 | leifmadsen | why must you emphasize so much? |
| 03:19.02 | Alton2 | competition? :-) |
| 03:19.08 | leifmadsen | compensating |
| 03:19.09 | [TK]D-Fender | leifmadsen: WhAt ArE YoU TaLkInG aBoUt?! |
| 03:19.28 | [TK]D-Fender | heh |
| 03:19.29 | leifmadsen | I appreciate all the help you give, but you really do come across as a dick, and it's totally unnecessary |
| 03:19.48 | leifmadsen | I wish you'd just tone it down |
| 03:19.54 | Alton2 | I ran into another old guy here last night. But I wonder what the average age of most Asterisk users is sometimes. |
| 03:20.02 | [TK]D-Fender | leifmadsen: thats a jab, if it seems competitive, its in a friendly way so we can all keep our wits sharp :) |
| 03:20.05 | leifmadsen | age has nothing to do with it |
| 03:20.38 | Alton2 | I dunno, seen so many teenage "computer geniuses". |
| 03:20.48 | leifmadsen | I know plenty of older guys who talk like a 14 y/o farmer, and 20 y/o's who talk like professional CEOs |
| 03:20.57 | leifmadsen | [TK]D-Fender: huh? |
| 03:21.04 | carrar | Farm talk is HOT |
| 03:21.10 | pcrane | lol |
| 03:21.19 | [TK]D-Fender | leifmadsen>I appreciate all the help you give, but you really do come across as a dick, and it's totally unnecessary |
| 03:21.35 | leifmadsen | right... I don't see where I mention competition |
| 03:21.41 | *** join/#asterisk [gnubie] (n=gnubie@203.177.180.52) |
| 03:21.53 | [gnubie] | waves to all.. |
| 03:21.53 | Alton2 | Maybe so, but usually there is a certain correlation between age and behaviour. Although not always. |
| 03:22.13 | pcrane | waves to [gnubie] |
| 03:22.17 | leifmadsen | Alton2: sometimes, but I wouldn't go on behaviour in this room to differentiate age |
| 03:22.22 | [gnubie] | anyone knows why i am getting this error? => http://paste.debian.net/15785/ |
| 03:22.31 | Alton2 | ok, you must know better than I! :-) |
| 03:22.56 | leifmadsen | I have met most of the active users in this room on at least one occasion in real life |
| 03:23.03 | [gnubie] | thanks pcrane |
| 03:23.23 | carrar | Have we met? |
| 03:23.31 | carrar | maybe I'm not active |
| 03:23.40 | [TK]D-Fender | [gnubie]: You shouldn't be loading ztdummy, and what is "wcopenpci"? |
| 03:23.43 | leifmadsen | not sexually active anyways |
| 03:23.48 | carrar | haha |
| 03:23.50 | leifmadsen | :) |
| 03:23.55 | leifmadsen | burn |
| 03:23.59 | carrar | stop spying on me!! |
| 03:24.27 | Alton2 | This is driving me to drink! I will return soon, suitably fortified. |
| 03:24.33 | [gnubie] | [TK]D-Fender: wcopenpci is i think for the voicetronix.. i am not loading ztdummy, it is loaded by default. maybe i will add ztdummy to my blacklist |
| 03:24.54 | leifmadsen | I've been tempted to open a bottle of wine all night.... |
| 03:25.03 | leifmadsen | I could possibly have some brandy and milk... |
| 03:25.06 | [gnubie] | [TK]D-Fender: isn't it that ztdummy is required if you want to have a meetme feature on asterisk? |
| 03:25.11 | Knightfal | [gnubie] Did you just load the drivers... I had a similar issue and it was because I didnt have the right drivers |
| 03:25.17 | leifmadsen | [gnubie]: only if you're not installing hardware |
| 03:25.21 | carrar | I'm contemplating on grabbing a beer since I'm not actively having sex |
| 03:25.30 | [gnubie] | Knightfal: no, i didn't |
| 03:25.30 | leifmadsen | lol |
| 03:25.38 | [TK]D-Fender | [gnubie]: "cat /proc/interrupts" , "ls -la /dev/zap" |
| 03:25.49 | [gnubie] | leifmadsen: the hardware is already attached to the box |
| 03:26.20 | leifmadsen | right, so you just need to load the appropriate driver for the hardware. That will give you the timing source you need for the applications that require it. ztdummy provides timing via the kernel should you have no hardware |
| 03:27.22 | [gnubie] | [TK]D-Fender: kindly check this => http://paste.debian.net/15786/ |
| 03:28.19 | [TK]D-Fender | [gnubie]: I don't see your card there... modprobe the device and recheck |
| 03:28.21 | Knightfal | [gnubie]: Just try installing the right drivers for your * ver I beat my self up over the same problem for a bit |
| 03:28.28 | [TK]D-Fender | [gnubie]: If you still odn't see it it might be dead. |
| 03:28.51 | [TK]D-Fender | Knightfal: He isn't even REACHING *. |
| 03:29.01 | Knightfal | Whaa |
| 03:29.15 | [gnubie] | [TK]D-Fender: when you say "modprobe", you mean, loading the device driver/module? |
| 03:29.28 | [TK]D-Fender | [gnubie]: Yes |
| 03:29.32 | [gnubie] | ok |
| 03:29.53 | [TK]D-Fender | Knightfal: his is a ztcfg error. I wouldn't even be thinking about * at this phase. |
| 03:29.58 | Knightfal | ya |
| 03:30.07 | Knightfal | I see now |
| 03:30.38 | Knightfal | [TK]D-Fender>: Do you live here... I always see you :) |
| 03:31.32 | [gnubie] | [TK]D-Fender: i can't rmmod the zaptel driver before i load it manually.. zaptel is being used by crc_ccitt, irda, etc.. |
| 03:31.42 | WimpMan | Using * doesn't leave time for a life, I guess :-) |
| 03:31.59 | Knightfal | Kinda took some of you to work.... Thay ask me about the features on there phones and i say "Read the Manual" :P |
| 03:32.17 | *** join/#asterisk JenniferAkemi- (n=akemi@dhcp-0-13-10-42-a4-3c.cpe.axion.ca) |
| 03:32.25 | Alton2 | Hah, this is why they join. |
| 03:33.25 | [gnubie] | [TK]D-Fender: ok.. i was able to rmmod zaptel.. now, i will load it manually |
| 03:36.17 | Chicago | Does asterisk version 1.4.21.2 include TCP/TLS support for SIP? I found this article from July of 2007. => http://www.venturevoip.com/news.php?rssid=1777 |
| 03:36.36 | *** join/#asterisk Keizer (n=Keizer@208.50.100.60) |
| 03:37.07 | Qwell | Chicago: 1.6 |
| 03:37.28 | drmessano | 1.6 is being released in Chicago? |
| 03:37.34 | drmessano | What about the rest of the world? |
| 03:37.45 | drmessano | God you guys are pissing me the fuck off |
| 03:37.50 | drmessano | I WANT 1.6 |
| 03:37.58 | Qwell | drmessano: rc3 |
| 03:38.33 | drmessano | I'm blogging all the RC3 jokes in my series "Is 1.6 vaporware?" on my blog.. keep talking, funnyman |
| 03:39.13 | [TK]D-Fender | drmessano: No, not vaporware,we have its... its just PermaBeta :) |
| 03:39.49 | jaytee | PermaBeta? that kinda rhymes with masturbator |
| 03:40.19 | Chicago | The issue about Windows Messenger SIP client RTC/1.3.5470 (Messenger 5.1.0715) Looks like it was fixed here -> http://bugs.digium.com/file_download.php?file_id=15454&type=bug and I don't know if that session timers support is in my version. |
| 03:40.31 | Chicago | Is the session-timers fix there included in asterisk version 1.4.21.2? |
| 03:40.44 | WimpMan | So 1.6 will have uilt in flash telephony support? *eg* |
| 03:40.47 | *** join/#asterisk juanjoc (n=juanjoc@host162.200-45-97.telecom.net.ar) |
| 03:41.23 | drmessano | 1.6 needs a new CMOS battery |
| 03:41.27 | drmessano | It's clock is broken |
| 03:43.26 | *** join/#asterisk mvanbaak (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 03:44.23 | Alton2 | what is flash telephony? |
| 03:45.13 | WimpMan | A softphone in flash for your Browser. We had that one earlier on. |
| 03:45.21 | Chicago | Alton2: I read WimpMan wrong, I thought he asked if 1.6 would have flash telepathy support. |
| 03:45.34 | Chicago | I would like telepathy support included in the next release. |
| 03:46.07 | Alton2 | Why would Asterisk need to include that? there are plenty of softphones around. |
| 03:46.10 | WimpMan | Chicago: Yes for the masturbation thingie. That's where the flash shit is needed. |
| 03:46.24 | Alton2 | I do like the console dial command to work. It is useful here and there, mainly for automation. |
| 03:47.20 | WimpMan | Alton2: As much as you need to masturbate, wen using it. Get the idea? |
| 03:47.49 | drmessano | WimpMan: If you are on IRC for sex, maybe Dalnet would be more appropriate? |
| 03:48.27 | Alton2 | I didn't get it, but I just figured they were way to intellectual for me. :-) |
| 03:48.32 | Alton2 | too |
| 03:48.59 | WimpMan | Ok, people often find my jokes too komplicated. Need to work on that. |
| 03:49.12 | drmessano | Alton2: I think WimpMan is clearly trying to "tap your foot" under the proverbial IRC bathroom stall. I would suggest finding the nearest IRCop and having him G-Lined |
| 03:50.55 | drmessano | Great |
| 03:51.06 | drmessano | Now you chased off the only woman on Freenode |
| 03:51.23 | Alton2 | Aw, she's always timing out. |
| 03:51.49 | drmessano | Oh, for the person who keeps messaging me, who shall remain nameless, I will NOT PUT ON A DIAPER for YOU, so STOP PM'ing me. |
| 03:52.04 | drmessano | Geez people, this is #asterisk (AKA SIRIUS BUZNES) |
| 03:52.15 | Alton2 | Hah, that reminds me of when I was 18 years old, went to some company new year's party, |
| 03:52.47 | Alton2 | last thing I remember (after drinking too much Stolichnaya vodka) was following around my coworker who was dressed as Baby New Year, |
| 03:52.55 | Alton2 | asking her if I could change her diaper. :-) |
| 03:53.10 | drmessano | lol |
| 03:54.03 | drmessano | If it couldn't be construed as child pornography, I would ask how that went in the context of the baby metaphor. |
| 03:54.41 | Alton2 | she was wearing what passed for a diaper. probably a bath towel. I was only concerned with how to remove it at the time. |
| 03:54.52 | Alton2 | alas, it was not to be! |
| 03:55.53 | Keizer | Hey guys |
| 03:57.03 | Keizer | How can I make sure that when someone calls my system and then I forward the call to another one of my phones that the remote call party id stays the same as the number that was first called so my provider won't drop the call? |
| 03:58.18 | WimpMan | something like callerid(num)=${exten} |
| 03:58.56 | Alton2 | I think in sip.conf |
| 03:59.00 | Alton2 | callerid=xxxxx |
| 03:59.06 | Alton2 | check into that |
| 03:59.30 | WimpMan | Wouldn't that only affect incomming calls? |
| 03:59.37 | Alton2 | that would be in the context of the peer that you use to forward the call, outgoing peer |
| 03:59.45 | Alton2 | I am not certain of that, please double check. |
| 03:59.53 | Alton2 | but I think that's how you set outgoing caller id |
| 04:00.15 | Keizer | But I need a diversion header in the INVITE |
| 04:00.35 | [TK]D-Fender | Keizer: And what are you doing not to acheive this? |
| 04:00.45 | [TK]D-Fender | s/not/now/ |
| 04:01.20 | WimpMan | Interesting bot feature. |
| 04:01.46 | Alton2 | hah, yes |
| 04:01.55 | Alton2 | darned bots can be good for something after all. |
| 04:02.18 | TJNII | ~hug Alton2 |
| 04:02.19 | jbot | ACTION hugs Alton2 |
| 04:02.49 | Keizer | Hah |
| 04:03.08 | Alton2 | There is a lot more kinkiness in this channel than I was previously aware of! :-) |
| 04:03.22 | Keizer | This is the problem I am having: |
| 04:03.23 | Keizer | http://www.freepbx.org/trac/ticket/1908 |
| 04:04.02 | [TK]D-Fender | Keizer: If you want to add a header, you can just go ahead and add it. |
| 04:04.23 | [TK]D-Fender | Keizer: "core show application sipaddheader" |
| 04:06.27 | Keizer | [TK]D-Fender, Cool |
| 04:06.35 | Keizer | Can you add a header real time with that? |
| 04:09.01 | [TK]D-Fender | Keizer: You do it before passing off the call. |
| 04:09.16 | [TK]D-Fender | Keizer: It'll get added to the invite |
| 04:11.41 | Keizer | [TK]D-Fender, I'm using the FreePBX interface so now I have to figure out how to make that change |
| 04:13.01 | [TK]D-Fender | Keizer: Yeah, thats usually a cue to "exit, stage left" |
| 04:13.21 | Keizer | Bigtime |
| 04:13.45 | Keizer | I think what I need to do is find out which macro is called up to do the call forwarding and add the header to that macro |
| 04:14.31 | Keizer | exten => s,1,SIPAddHeader(Diversion: <tel:+{ARG1}>\;reason=user=busy\;screen=no\;privacy=off) |
| 04:14.52 | Keizer | I need to find where the macro that does the forwarding exists now |
| 04:15.01 | jaytee | I use that very line |
| 04:15.10 | [TK]D-Fender | Keizer: Good, you're not clueless... just swimming upstream. At least there's hope... |
| 04:15.22 | [TK]D-Fender | Keizer: Good luck with all that :) |
| 04:15.38 | *** join/#asterisk jsolis (n=jimmy@200.106.75.107) |
| 04:16.12 | Keizer | lol |
| 04:16.40 | Keizer | jaytee, Do you know how I can go about finding the call forward bit so I can add it? |
| 04:17.48 | jaytee | no clue |
| 04:19.32 | jaytee | I use that line above after a Dial statement line above it fails with no answer after a timeout, then it adds the Diversion info to a new header and the next line in my macro dials my Exchange UM number via a sip/proxy. |
| 04:23.28 | *** join/#asterisk MrNaz (n=naz@ppp118-208-151-204.lns10.mel4.internode.on.net) |
| 04:32.16 | *** join/#asterisk echosyp (n=echosyp@adsl-66-140-104-8.dsl.lbcktx.swbell.net) |
| 04:33.16 | echosyp | i need an ATA i belive its called, that i can config myself |
| 04:33.32 | echosyp | i know alot of them come locked, anyone have a suggestion for me |
| 04:33.53 | echosyp | the goal is to use gizmo to ring the phone in my room |
| 04:35.46 | Alton2 | shouldn't come locked if you buy one fair and square |
| 04:38.28 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 04:38.46 | echosyp | so i could just go buy a linksys ATA and tell it to use gizmo sip info and plug my phone into that and be good? |
| 04:39.26 | drmessano | jaytee |
| 04:39.39 | jaytee | yeah |
| 04:40.06 | drmessano | I have nothing |
| 04:40.12 | drmessano | I was like JAYTEE! |
| 04:40.14 | drmessano | Then.. ____ |
| 04:40.17 | echosyp | http://www.linksys.com/servlet/Satellite?childpagename=US%2FLayout&packedargs=c%3DL_Product_C2%26cid%3D1117044308483%26site%3DUS&pagename=Linksys%2FCommon%2FVisitorWrapper |
| 04:40.23 | echosyp | that thing would work? |
| 04:41.10 | [TK]D-Fender | echosyp: "use gizmo sip info" huh? |
| 04:42.18 | echosyp | SIP Proxy: proxy01.sipphone.com |
| 04:42.25 | echosyp | to be more specific |
| 04:43.41 | echosyp | the linksys pap2 is locked to vonage |
| 04:43.50 | echosyp | i need one thats not locked |
| 04:43.55 | echosyp | so i can use gizmo |
| 04:44.30 | [TK]D-Fender | echosyp: www.telephonydepot.com |
| 04:45.45 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 04:48.13 | echosyp | is it unlocked? |
| 04:48.13 | echosyp | it doesn't say anything about it |
| 04:48.49 | [TK]D-Fender | echosyp: Nothing they sell is locked |
| 04:48.55 | echosyp | cool |
| 04:48.57 | echosyp | im buying it |
| 04:49.05 | echosyp | its gonna be sweet |
| 04:49.09 | [TK]D-Fender | echosyp: Any normal VoIP equipment dealer would not be selling locked devices |
| 04:49.31 | echosyp | well, if you go to a retail store and buy one its locked |
| 04:49.34 | *** join/#asterisk Levonk (n=lk@adsl-76-230-108-243.dsl.lsan03.sbcglobal.net) |
| 04:49.35 | echosyp | like, circuit city |
| 04:49.40 | echosyp | or best buy, |
| 04:49.48 | echosyp | anything voip is vonage crap |
| 04:50.10 | echosyp | and those would be my only two options for such a device locally |
| 04:51.08 | *** part/#asterisk jsolis (n=jimmy@200.106.75.107) |
| 04:51.37 | Alton2 | he means vonage |
| 04:53.18 | jblack | If you want voip equipment, buy it online. |
| 04:53.29 | Alton2 | must be |
| 04:53.44 | Alton2 | sheesh, no customer at a retail store would know what to do with an actual ATA |
| 04:54.48 | jblack | There's an agi script that I used that provides wakeup calls for the house. It's really useful to me. That makes me wonder if there's other stuff out there, just as useful |
| 04:55.57 | drmessano | OH WOW |
| 04:56.08 | drmessano | IPKALL rings thru to Grandcentral now |
| 04:56.10 | drmessano | errr |
| 04:56.19 | drmessano | Grandcentral rings thru to IPKALL now |
| 04:56.20 | jblack | It didn't before? |
| 04:56.22 | drmessano | No |
| 04:56.36 | drmessano | I was getting a friend set up |
| 04:56.38 | jblack | they blocked pstn calls? |
| 04:57.03 | echosyp | no |
| 04:57.12 | drmessano | Gave him my main PBX number, and I had it set up on too many accounts.. I told him "Fuck it, put your IPKALL in" |
| 04:57.18 | drmessano | and bam, somehow it worked |
| 04:57.32 | jblack | ~conference |
| 04:57.32 | jbot | conference is probably The channel conference can be reached via IAX2/asterisk.linuxguru.net , SIP/asterisk.linuxguru.net , and PSTN 1-206-424-2834. Contact jblack with any concerns. |
| 04:57.37 | jblack | Try tat one. it's an ipkall |
| 04:57.48 | drmessano | No no |
| 04:57.52 | drmessano | Listen |
| 04:58.34 | jblack | I'm listening |
| 04:58.50 | drmessano | If you put IPKALL as a destination for a Grandcentral DID, it was not putting the call thru... It kinda made sense since IPAKLLs business model is based on those switching fees... that maybe the calls were being switched over IP for free and they had them blocked.. perhaps |
| 04:58.57 | drmessano | But that is now working fine |
| 04:59.17 | jblack | so, the question is, did they miss a number, or is it a change in policy |
| 04:59.25 | drmessano | They missed two now |
| 04:59.28 | drmessano | Soon to be 3 |
| 04:59.43 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 04:59.43 | *** mode/#asterisk [+o denon] by ChanServ |
| 04:59.44 | drmessano | I'm moving some of my GC numbers to my IPKALL numbers |
| 04:59.50 | jblack | That 206 number above is an ipkall. |
| 05:00.01 | drmessano | Yes |
| 05:00.25 | jblack | did you just join? |
| 05:00.30 | drmessano | No |
| 05:00.42 | jblack | someone did, and they're not saying anything. :) |
| 05:01.47 | baliktad | I dropped a handful of MP3's in my MOH directory, most of them play fine but get cut off at some point with this message in the console: |
| 05:01.49 | baliktad | [Aug 27 21:58:27] WARNING[5409]: layer3.c:966 III_dequantize_sample: mpg123: Can't rewind stream by 7 bits! |
| 05:02.39 | baliktad | is it easier to try and fix this or just convert my audio files to another format |
| 05:02.42 | jblack | the mp3 decoder is limited in the mp3s it can play. |
| 05:02.51 | jblack | You're usually better off converting them. |
| 05:03.47 | baliktad | I'm not sure what format they should be in |
| 05:03.51 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
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| 05:29.32 | [gnubie] | waves.. gtg now.. thanks.. |
| 05:32.03 | *** join/#asterisk gerphimum (n=trekkie@cpe-24-174-138-171.satx.res.rr.com) |
| 05:44.39 | C4away | if a call comes in on a T1 channel does the system even know the call exists until it gets some information on the D channel? |
| 05:47.35 | WimpMan | it comes in by being announced on the D channel. |
| 05:47.46 | C4away | that's what I thought |
| 05:51.07 | drmessano | jblack |
| 05:51.26 | drmessano | Here's a side effect of using Grandcentral > IPKALL |
| 05:51.56 | drmessano | Wrong numbers keep activity on the line lol |
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| 07:06.22 | koenvi | hi all |
| 07:07.04 | *** join/#asterisk dmz (n=dmz@72-254-34-35.client.stsn.net) |
| 07:07.22 | koenvi | I just compiled and installed 1.6.0-beta3 on an Ubuntu Hardy 64bit, with dadhi for a dummy device. |
| 07:07.48 | koenvi | using Zoiper, I made a first call... but the sound is terrible |
| 07:08.39 | koenvi | sound is chopped, as if there's something not running with the right timing |
| 07:08.56 | drmessano | Does anyone have any grandcentral invites? |
| 07:09.10 | drmessano | beta3? |
| 07:09.13 | koenvi | should asterisk run ok on an 64bit system? |
| 07:09.14 | WimpMan | Probably your gcc is too recent. Thare's a known issue with some codecs. |
| 07:09.50 | DarKnesS_WolF | koenvi: change the codes try to use ulaw |
| 07:09.54 | DarKnesS_WolF | alaw / gsm |
| 07:09.55 | DarKnesS_WolF | and test |
| 07:09.57 | WimpMan | Try forcing alaw or ulaw to see if that could be the problem. |
| 07:10.30 | drmessano | jblack: Do you have any grandcentral invites? |
| 07:10.31 | koenvi | ok guys, thx for the tip. I'll have a look at codecs |
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| 07:24.05 | FabiOne | hi all |
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| 07:24.46 | DarKnesS_WolF | hi |
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| 07:52.54 | Andre101 | hello people.. does anyone know a good walkthrough setting up asterisk? it's been a while |
| 07:53.12 | Maliuta | ~book |
| 07:53.12 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 07:54.11 | Andre101 | cheers :) |
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| 08:01.55 | r0land | hello all |
| 08:02.18 | *** join/#asterisk darkskiez (n=mhb@cpc1-broo3-0-0-cust659.renf.cable.ntl.com) |
| 08:05.37 | r0land | my topology is as such : <sipPhone> --> <asterisk> --> <sipura3102> |
| 08:05.45 | r0land | everythings working pretty good.. |
| 08:06.05 | DarKnesS_WolF | r0land: so ? |
| 08:06.10 | r0land | though when i call/recieve a call from sipura, if i talk at the same time as the other end.. |
| 08:06.30 | r0land | the sound quality goes down the drain.. the other end's voice gets extremly low |
| 08:06.41 | Maliuta | codec? |
| 08:06.52 | Maliuta | is * simply doing passthrough? |
| 08:07.16 | r0land | Maliuta g711u |
| 08:07.21 | r0land | Maliuta yes |
| 08:07.53 | Maliuta | then there's nothing to implicate * in the problem |
| 08:08.10 | r0land | Maliuta if i talk directly from sipura, the sound is pretty good |
| 08:08.36 | r0land | the prob only occurs when im talking from a sipPhone thts connected to asterisk |
| 08:08.46 | *** join/#asterisk vale-ICS (n=vale@icsnet.demon.co.uk) |
| 08:09.01 | r0land | one quest.. whts the best codec available if bandwidth isnt an issue?! |
| 08:11.13 | Maliuta | gsm |
| 08:12.00 | r0land | hmm |
| 08:12.11 | r0land | would dtmfmode cause such an issue?! |
| 08:13.15 | DarKnesS_WolF | r0land: ulaw/alaw is better than GSM in quilty |
| 08:15.05 | r0land | oh ok |
| 08:15.10 | r0land | well im kinda bound to just a few |
| 08:15.13 | r0land | in sipura |
| 08:16.09 | DarKnesS_WolF | r0land: ulaw = G711U |
| 08:16.10 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 08:16.17 | r0land | DarKnesS_WolF hmm thts the one chosen |
| 08:16.39 | r0land | DarKnesS_WolF one more quest.. wht does "dtmfmode" do.. |
| 08:17.41 | r0land | i have it in sip.conf under [general] to dtmfmode = rfc2833 |
| 08:19.42 | jblack | the mars rovers are starting to bore me. 90 days my ass. |
| 08:20.40 | DarKnesS_WolF | r0land: dtmf is the tone u do let us say if u called a voicemail system from ur SPA u need to send DTMF like " passwors / mailbox or action " the SPA should send it in a mode which the asterisk will understand |
| 08:21.00 | DarKnesS_WolF | rfc2833 is the default there is also sip_infon " info " or inband |
| 08:21.03 | jblack | the spa has 2833, but it's called something else |
| 08:23.02 | r0land | oh ok |
| 08:23.03 | r0land | gr |
| 08:23.04 | r0land | gr8 |
| 08:23.06 | r0land | hi jblack :) |
| 08:23.41 | jblack | flood much? |
| 08:24.05 | r0land | i just changed sipura's to g711u as its default codec.. the sound on the other end is perfect they dont have a prob though on my end, if we speak the same time.. i hear my own voice's echo |
| 08:24.44 | jblack | In-band = rfc2833 |
| 08:24.56 | r0land | i add it in general in sp.conf ?! |
| 08:25.47 | jblack | you add it in the spa's web configuration. |
| 08:26.09 | jblack | dtmf mode should be somehere, either in-band, avt out-of-band ,etc. You want in-band. in sip.conf, set rfc2833 |
| 08:26.53 | r0land | jblack ive set dtmfmode under [general] in sip.conf allready |
| 08:27.04 | r0land | as for sipura ive set the default codec as g711u |
| 08:27.40 | jblack | Oh, you're taking about something different than I thought. you're having dtmf issues. |
| 08:27.49 | jblack | you're not, I mean. Instead, you're having echo. |
| 08:28.01 | r0land | heres wht im facing |
| 08:28.27 | jblack | is really tired, but he'll give it a shot, because you were so financially ncie the other day. :) |
| 08:28.51 | r0land | <PROTECTED> |
| 08:28.55 | *** join/#asterisk EI5GTB (n=EI5GTB@193.120.192.61) |
| 08:29.11 | jblack | it may be noise cancellation in the sipura. |
| 08:29.40 | jblack | Huh? |
| 08:29.58 | jblack | Dude, unless I'm confusing you with someone else, you were very nice. That was sincere |
| 08:30.05 | r0land | jblack :) |
| 08:30.13 | r0land | thanks |
| 08:30.33 | jblack | Check the sipura config and see if it has noise cancellation. Try turning that off first. |
| 08:30.47 | jblack | If that doesn't do it, try flipping echo cancellation in the phone if it's built in. |
| 08:31.13 | r0land | jblack i just enabled echo cancelation in sipura |
| 08:31.14 | r0land | ill check |
| 08:31.16 | jblack | my bet is noise cancellation. I see the sort of thing on this polycom here |
| 08:32.09 | r0land | u mean echo cancelation?! |
| 08:32.23 | jblack | i mean noise cancellation. |
| 08:32.36 | jblack | Try both, of course. I'm putting my money on noise cancellation. |
| 08:32.51 | jblack | It's coming out the speaker so loud, it's probably being detected by the phone as ambient noise. =) |
| 08:32.54 | r0land | dont have noice cancelation :( |
| 08:33.03 | r0land | ya thts it |
| 08:33.05 | r0land | jblack thts wht happens |
| 08:33.06 | jblack | then try echo, cause that can do it too |
| 08:33.12 | r0land | when the audio breaks i hear my own voice pretty LOUD |
| 08:34.12 | *** join/#asterisk Psychobilly (n=moi@adsl100-91.kln.forthnet.gr) |
| 08:34.18 | jblack | http://michigantelephone.wordpress.com/2007/11/16/linksys-and-sipura-adapter-users-check-your-rtp-packet-size-and-newtork-jitter-level/ |
| 08:34.33 | jblack | Interesting, that page. |
| 08:35.26 | jblack | That's exactly the page you want, r0land. |
| 08:36.00 | *** join/#asterisk _Krieger_ (n=krieger@ip-193.39.118.158.dukenet.odessa.ua) |
| 08:37.08 | jblack | btw, if you need to call yourself, you're welcome to use the conference if it makes itt easier. |
| 08:38.16 | *** join/#asterisk wonderworld (n=ww@ip-62-143-39-68.hsi.ish.de) |
| 08:40.40 | r0land | thanks jblack ill check it |
| 08:41.41 | Andre101 | hey people, I have 2 DID's with the one VoIP provider.. How do I tell asterisk to handle them differently? |
| 08:42.21 | jblack | i found something better than being montgomeried. roflmaod. http://www.youtube.com/watch?v=iEWgs6YQR9A |
| 08:42.27 | Maliuta | Andre101: well the sip.conf enteries will be different for 1 |
| 08:42.49 | Maliuta | Andre101: and "handle them different" could mean just about anything |
| 08:42.59 | florz | Andre101: what kind of protocol? |
| 08:43.06 | Andre101 | sip |
| 08:43.19 | florz | Andre101: on a single account, or one account per number? |
| 08:43.30 | Andre101 | florz: single account |
| 08:43.35 | Maliuta | Andre101: it also depends on how the provider is sending them in, if both are to the 's' extension you might want to use different dialplan contexts |
| 08:43.45 | _Krieger_ | is it possible to ask a peer to re-register on a server when it is made expired manually? |
| 08:44.13 | florz | Andre101: well, basically you are screwed =:-) |
| 08:44.18 | Maliuta | Andre101: if they are coming in to some other extensions then you can just have different things done in each extension |
| 08:44.59 | Andre101 | florz: that sucks.. they both come in on the s extension.. |
| 08:45.11 | Maliuta | Andre101: if they are coming into the same extension you might try checking some of the callerid info or other stuff in the SIP header and having a conditional extension |
| 08:45.15 | florz | Andre101: you'll have to either have the "luck" of having a provider that doesn't actually speak SIP but is "asterisk-compatible" instead, or you will have to have a look at the INVITE's headers and then parse them manually in the dialplan |
| 08:46.18 | florz | Andre101: well, that's because you specified that in the register statement - and because asterisk thinks that the request-URI's local part is a good approximation of the "destination number2 |
| 08:46.18 | Andre101 | when you say doesn't speak SIP but asterisk-compatible.. you mean IAX? |
| 08:46.37 | florz | Andre101: No, something very much resembling SIP |
| 08:46.44 | *** join/#asterisk Levonk (n=lk@adsl-75-62-132-123.dsl.lsan03.sbcglobal.net) |
| 08:47.25 | *** join/#asterisk synthetiq (n=roger@unl201395.nl.customer.alter.net) |
| 08:47.29 | florz | Andre101: there are providers out there that will put the destination number in the request-URI's local part, even though you specified a different local part when registering |
| 08:47.41 | *** join/#asterisk stimpie (n=stimpie@84-104-5-14.cable.quicknet.nl) |
| 08:48.26 | florz | that makes it easy to use with asterisk - but isn't compatible with the SIP RFC |
| 08:48.26 | Maliuta | florz: a provider I used to use sent things to "did@<myip>" |
| 08:48.26 | synthetiq | for jabber/jingle stuff in asterisk, is it possible for users to send messages to each other thru asterisk or is it jsut mainly a calling solution? i dont see how users can receive messages |
| 08:48.28 | Maliuta | rather than to 's' or something else |
| 08:48.46 | florz | Maliuta: you mean, as in sip:did@ip ? |
| 08:49.07 | *** join/#asterisk padski (n=padski@host217-45-159-149.in-addr.btopenworld.com) |
| 08:49.11 | florz | Maliuta: even though you hadn't specifieded that URI when registering? |
| 08:49.12 | Maliuta | florz: jah |
| 08:49.28 | Maliuta | florz: yup |
| 08:49.49 | florz | Maliuta: well, that's the SIP-incompatible behaviour I was speaking abount |
| 08:50.16 | Maliuta | florz: they were a crap provider anyhow, glad to be rid of them |
| 08:52.15 | Andre101 | hmm.. florz, so what steps do you recommend I take to get this working? |
| 08:52.29 | florz | Andre101: get rid of asterisk ... hrhr |
| 08:52.38 | Andre101 | lol |
| 08:52.43 | florz | Andre101: Well, as I said, have a look at the headers of the INVITE |
| 08:53.06 | florz | Andre101: and then write some dialplan code that extracts the number from whereever it can be found in the header |
| 08:53.52 | Andre101 | i tried just then.. i called both phone numbers, logged it all.. searched the log for the second number, nothing in there at all with the second number, only my main number |
| 08:54.12 | Maliuta | Andre101: did you have sip debugging on? |
| 08:54.24 | Andre101 | Yep.. |
| 08:54.39 | Maliuta | still got the logs? |
| 08:54.43 | Andre101 | yea |
| 08:54.52 | Maliuta | pastebin? |
| 08:55.05 | florz | well - if the SIP messages don't contain the destination number or anything that can be used to identify which number has been called, there won't be anything you can do at the receiver's side |
| 08:57.07 | Andre101 | Is this different for every call: Call-ID: 27312894-741511DD-BD3DE344-E0E92412@202.85.241.127 |
| 08:57.25 | florz | yeah, it has to be ;-) |
| 08:57.50 | Andre101 | looks like I am out of luck... |
| 08:57.52 | florz | but would be an unusual place to store numbering plan addressing information |
| 08:58.15 | Andre101 | I'll have to talk to my SIP provider, see if they can move that second number to another SIP account |
| 08:58.47 | florz | that's at least the easier way with asterisk, yeah |
| 09:01.08 | Andre101 | thanks a heap for your help.. |
| 09:01.53 | *** join/#asterisk gr0mit (n=tim@dhcp3.zuk40.mot-tools.co.uk) |
| 09:02.31 | Andre101 | also, how good is g729? Worth the effort? |
| 09:03.42 | *** join/#asterisk ToTo (n=ToTo@207.176.6.200) |
| 09:03.58 | florz | well, probably depends on your bandwidth costs |
| 09:04.26 | florz | but if that really is a problem, I'd probably first look at speex and ilbc |
| 09:05.39 | Andre101 | i have noticed when i play a dialtone or ringtone from the * box to an inbound caller via my VoIP provider, it sounds like crap, would using g729 fix this? |
| 09:06.51 | florz | obviously depends on whether the current codec is the reason for the problems? |
| 09:07.18 | florz | but in any case G.711 will fix that problem if G.729 does |
| 09:10.16 | Andre101 | ok, i'll check it out.. |
| 09:10.26 | Andre101 | thanks again for your help :) |
| 09:10.36 | Andre101 | I'm sure I'll be back soon asking another random question |
| 09:10.36 | C4away | Random answer[136]: i like your questions. (from Ingabish) |
| 09:10.43 | WimpMan | florz: Thanks for the hint regarding missed calls yesterday. Works well for SIP now. |
| 09:15.00 | Andre101 | florz: Online support with my SIP provider now, looks like he is going to give me a config to get this second DID working? |
| 09:18.04 | Andre101 | i got a feeling it's not going to work though |
| 09:18.35 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 09:24.06 | *** join/#asterisk my007ms (i=master@217.139.17.150) |
| 09:24.22 | my007ms | can i make asterisk route based in source ext ? |
| 09:24.49 | my007ms | for example ext 1-100 go out from trunk1 while other go from trunk2 |
| 09:28.16 | *** join/#asterisk s0lid (n=s0lid@60.52.253.84) |
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| 10:09.29 | *** join/#asterisk [netman] (n=netman@88.Red-83-50-82.dynamicIP.rima-tde.net) |
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| 10:22.36 | kaldemar | my007ms: exten => <somewhere>/<from_somewhere>,1,... |
| 10:23.44 | my007ms | kaldemar where i can find doc abut this ? thanks for help :) |
| 10:24.15 | kaldemar | my007ms: but if you're handling a range of caller id's, use gotoif to examine it and jump to the right extension. |
| 10:24.22 | kaldemar | ~thebook |
| 10:24.24 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 10:25.12 | kaldemar | read up on dialplans |
| 10:25.33 | my007ms | i read it once b4 long time ago |
| 10:25.40 | my007ms | and never see this pattern with / |
| 10:25.50 | my007ms | i will recheck it |
| 10:26.01 | my007ms | thanks much ;) |
| 10:26.02 | InsolentDreams | What's a good way to debug dundI? I am getting practically nothing from dundi debug |
| 10:26.34 | kaldemar | but as i said, examining CALLERID(num) is a better way. |
| 10:30.11 | *** join/#asterisk xenonex (n=xenonex@89.218.235.211) |
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| 11:27.32 | ph8 | has anyone got a grandstream? do you know if there's a linux desktop app that will interface with it? |
| 11:27.47 | yang | ph8: neer heard of it |
| 11:27.51 | yang | never |
| 11:28.00 | mgschwan | Hi, is there a way to debug the featurecodes (i.e. see why the feature does not work). I am trying automon with the default (*1) but it does not work, but attended transfer and blind transfer do work (so my dtmf settings are correct) |
| 11:28.10 | ph8 | yang: it's a popular type of phone iirc |
| 11:28.11 | yang | ph8: but you have the GUI for it |
| 11:28.24 | ph8 | i mean something that pops up when a call comes in etc |
| 11:28.26 | yang | ph8: I use GS phones, but i never heard of such application |
| 11:28.46 | yang | ph8: ah yes, check for TAPI www.xtelsio.com |
| 11:29.27 | yang | ph8: but that is for Windows OS |
| 11:31.57 | mgschwan | i guess this means i am on my own |
| 11:34.55 | viraptor | has anyone heard of meetme problems where different parties experience different call quality? (there are no network issues) - I couldn't google anything specific :/ |
| 11:35.34 | viraptor | (everyone on g711) |
| 11:36.10 | *** join/#asterisk Levonk (n=lk@adsl-75-62-142-249.dsl.lsan03.sbcglobal.net) |
| 11:45.04 | WimpMan | So if anyone else is annoyed with missed calls that were actually picked up on another phone, try this: http://www.yeti.dk/lsoft/asterisk-1.4.21.1.diff |
| 11:45.32 | WimpMan | Works with SIP and probably any ISDN interface, only tested with mISDN however. |
| 11:45.45 | *** join/#asterisk Gestahlt (n=chatzill@HSI-KBW-078-042-057-148.hsi3.kabel-badenwuerttemberg.de) |
| 11:45.49 | Gestahlt | Hi |
| 11:46.04 | Gestahlt | I have a few questions about possibilities in asterisk. |
| 11:46.19 | InsolentDreams | Heh, shoot |
| 11:46.25 | Gestahlt | I want to setup a Asterisk TK Server |
| 11:46.31 | Gestahlt | I have 4 MSN |
| 11:46.37 | Gestahlt | A few FritzCard PCI |
| 11:46.56 | Gestahlt | puh |
| 11:47.00 | Gestahlt | hard to explain in english |
| 11:47.02 | Gestahlt | lemme think |
| 11:47.04 | InsolentDreams | Eh, what is TK? |
| 11:47.07 | Gestahlt | Uh |
| 11:47.12 | InsolentDreams | I see this in german when I search |
| 11:47.21 | InsolentDreams | Oh nm, german proxy |
| 11:47.25 | InsolentDreams | haha, so what is it? |
| 11:47.25 | Gestahlt | Phoneserver? |
| 11:47.32 | Gestahlt | You have ISDN and 3 Phones |
| 11:47.38 | InsolentDreams | Sure |
| 11:47.40 | Gestahlt | and the TK assignes the numbers to the phones |
| 11:48.18 | InsolentDreams | Yeah, sure |
| 11:48.18 | Gestahlt | Basically that is what Asterisk is doing that afaik |
| 11:48.26 | Gestahlt | Just i want to use SIP Phones |
| 11:48.32 | InsolentDreams | Yeah, it can/does do that |
| 11:48.33 | Gestahlt | and Phone via Network |
| 11:48.43 | InsolentDreams | Ok, shouldn't matter how your phones connect to your asterisk box |
| 11:48.56 | InsolentDreams | They can use extensions via sip or iax |
| 11:49.02 | Gestahlt | Okay |
| 11:49.12 | Gestahlt | And how about setting up a fax server? |
| 11:49.16 | Gestahlt | I want to fax via network |
| 11:49.27 | InsolentDreams | And extensions can directly be setup to a DID. Ahh, ok well that's interesting. You need to make sure no compression is applied |
| 11:49.28 | Gestahlt | Usually i do that via a LAN Capi port and the Fritz Software |
| 11:49.41 | InsolentDreams | Ok... well what way are you trying to fax |
| 11:49.44 | InsolentDreams | Inbound or outbound |
| 11:49.47 | InsolentDreams | (or both?) |
| 11:49.47 | Gestahlt | Both |
| 11:49.50 | InsolentDreams | Ok... |
| 11:49.54 | InsolentDreams | From/to a real fax machine? |
| 11:49.57 | Gestahlt | No |
| 11:50.02 | Gestahlt | via Software |
| 11:50.13 | InsolentDreams | Ehm... |
| 11:50.23 | InsolentDreams | Well, g'luck on that one, there's some stuff out there on that |
| 11:50.30 | InsolentDreams | Just never been too successful myself |
| 11:50.35 | Gestahlt | right now my systems are running win2k pro, i got a CAPI LAN/Proxy server setup and send faxes over the network via CAPI LAN Clients and Fritz software |
| 11:51.12 | Gestahlt | I intented to setup a Xubuntu for asterisk, but i dont mind a win2k solution either |
| 11:51.21 | *** join/#asterisk IsUp (n=nocturne@88.255.77.200) |
| 11:51.22 | Gestahlt | if there is a win port for asterisk |
| 11:51.51 | Gestahlt | I also need a voicebox which i wanted to run via Asterisk |
| 11:51.57 | DarKnesS_WolF | Gary: using cygwn as i recall but it is not recommended install a GNU/Linux server just for asterisk and start playing with it |
| 11:51.59 | Gestahlt | and number redirection |
| 11:52.34 | DarKnesS_WolF | Gestahlt: define voicebox? |
| 11:52.48 | Gestahlt | uhhh.. a bit of a language barrier here.. |
| 11:52.56 | Gestahlt | When you got a incoming call |
| 11:52.58 | Gestahlt | after 5 rings |
| 11:53.01 | Gestahlt | answering machine |
| 11:53.03 | Gestahlt | thats the term |
| 11:53.05 | mchou | voicemail |
| 11:53.25 | *** join/#asterisk Levonk (n=lk@75.62.139.39) |
| 11:53.33 | mchou | Gestahlt: have babelfish handy :) |
| 11:53.52 | mchou | It will translate for you |
| 11:54.07 | WimpMan | better not |
| 11:54.13 | Gestahlt | mchou: *G* its good practice for me, i rarely have the occassion to speak english. I just read it a lot (manuals) |
| 11:54.27 | mchou | WimpMan: it's better than his English :) |
| 11:54.37 | InsolentDreams | Well, here's a cool place to practice ;) |
| 11:54.38 | WimpMan | I don't think so. |
| 11:54.43 | IsUp | any ideas about ramfs? |
| 11:54.48 | Gestahlt | ... |
| 11:54.52 | WimpMan | Babelfish is evil. |
| 11:55.01 | InsolentDreams | IsUp: what about it, and in what relation to asterisk? |
| 11:55.06 | Gestahlt | I never use translation engines... |
| 11:55.08 | mchou | there is nothing evil about babelfish |
| 11:55.18 | Gestahlt | I end up cursing.. which i can do fluently in the most languages |
| 11:55.33 | mchou | there are some things to be desired, but calling it evil is a real stretch |
| 11:56.26 | WimpMan | mchou: I don't know what languages you have tried it with, but it's german is really hard to understand. |
| 11:56.37 | mchou | Jawohl |
| 11:56.51 | mchou | WimpMan: try its |
| 11:56.57 | mchou | ot it's |
| 11:57.00 | mchou | not* |
| 11:57.19 | *** join/#asterisk moy (n=moy@CPE000d9d59afef-CM00140454d33e.cpe.net.cable.rogers.com) |
| 11:57.20 | mchou | babelfish would have gotten that correct |
| 11:57.21 | Gestahlt | Anyway, so i need a fax server, phone server (i want to record my calls), i want to phone via network (headset and soundcard) to ISDN. |
| 11:57.28 | Gestahlt | Is this possible? |
| 11:57.56 | mchou | maybe |
| 11:58.15 | Gestahlt | Maybe? What is lacking or is it just a matter of the setup? |
| 11:58.21 | mchou | everything except fax server has ok support in asterisk |
| 11:58.29 | Gestahlt | Ah i see |
| 11:58.34 | Gestahlt | thats what worried me |
| 11:59.44 | mchou | for fax server if you have analog just use a modem and hylafax |
| 11:59.47 | Gestahlt | Since i dont really know how to set it up. My current solution is you have a server with an ISDN Card installed and a CAPI server. You install on the Client machines a CAPI LAN Client and use the software shipped from Fritz for fax |
| 12:00.23 | mchou | depending on how "busy" your fax line is..... |
| 12:00.28 | Gestahlt | This is installed as a printer. When you want to fax the document, a dialog opens and you type in the number to dial and there you go |
| 12:00.34 | Gestahlt | works basically as a printer driver |
| 12:00.52 | Gestahlt | How busy? Well not much load |
| 12:01.05 | mgschwan | well my problem seems to be solely an automon problem, because if i exchange the feature code for bilndxfer and automon the automon still does not work but blindxfer still works (although with *1 instead of ##, which is quite obvious). And in the asterisk CLI when i do 'features show' automon shows up with current '##' |
| 12:01.10 | Gestahlt | I didnt have any good experiences with modem fax tho |
| 12:02.44 | mchou | all the German I learned I learned through "Return to Castle Wolfenstein" :) |
| 12:03.20 | ph8 | hey all, i'm trying to connect my new hardphone to asterisk, i have this in sip.conf: |
| 12:03.20 | ph8 | [Ph0] |
| 12:03.20 | ph8 | type=friend |
| 12:03.20 | *** join/#asterisk fnordus (n=dnall@70.71.225.48) |
| 12:03.20 | ph8 | host=dynamic |
| 12:03.23 | *** join/#asterisk The-Bat (n=The-Bat@203.199.114.33) |
| 12:03.27 | ph8 | oh and context=users |
| 12:03.37 | ph8 | i've configured the phone with the appropriate host and username Ph0 |
| 12:03.44 | ph8 | but it doesn't appear to be connected (sip show peers) |
| 12:03.45 | *** join/#asterisk af_ (n=getsmart@88-149-241-182.dynamic.ngi.it) |
| 12:03.58 | ph8 | if the phone tried to register and got rejected, would that show up in the asterisk messages log? |
| 12:04.08 | ph8 | i'm not sure whether it's actually getting to the server (might be because it's behind a router?) |
| 12:04.27 | mchou | ph8: RTFM dude |
| 12:04.47 | mchou | ~book |
| 12:04.47 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 12:05.11 | ph8 | i'm looking at it now |
| 12:05.17 | ph8 | i.e. already had it in front of me |
| 12:05.21 | ph8 | that looks right based on what the manual says |
| 12:07.40 | *** join/#asterisk lirakis (n=lirakis@65.200.191.241) |
| 12:07.49 | lirakis | hello everyone |
| 12:07.56 | ph8 | mchou: ? |
| 12:08.14 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:08.38 | lirakis | is any one here a developer for hire ??? I have a client looking to create a pretty simple conference server but I have too many projects now and can't take this one. I want to refer him to a private asterisk dev. |
| 12:10.01 | *** join/#asterisk russellb (n=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 12:10.01 | *** mode/#asterisk [+o russellb] by ChanServ |
| 12:14.01 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:15.41 | *** join/#asterisk scampbell (n=scampbel@mail.scampbell.net) |
| 12:17.02 | IsUp | ph8 |
| 12:17.04 | IsUp | set verbose 10 |
| 12:17.06 | IsUp | set debug 3 |
| 12:17.10 | IsUp | try reconnect your phone |
| 12:17.19 | ph8 | -vvvvv*10 ? |
| 12:17.28 | IsUp | asterisk -vvvgc |
| 12:17.29 | IsUp | and then type |
| 12:17.33 | IsUp | set verbose 10 |
| 12:17.34 | IsUp | set debug 3 |
| 12:18.18 | ph8 | done |
| 12:18.25 | IsUp | reconnect your phone now |
| 12:18.27 | ph8 | i think it might be the phone's fault tbh |
| 12:18.33 | IsUp | and check the CLI |
| 12:19.23 | ph8 | hmm maybe not |
| 12:19.28 | ph8 | rebooted the phone and all i see in CLI is Really destroying SIP dialog 'a7d5e6ae66ae6751@192.168.69.100' Method: REGISTER |
| 12:19.35 | ph8 | .100 being the address of my phone |
| 12:19.39 | ph8 | my PBX is out on the internet 85.* |
| 12:19.50 | ph8 | s/PBX/Asterisk |
| 12:20.43 | *** join/#asterisk Gat0rvean (n=gredish@64.191.128.145) |
| 12:20.44 | ph8 | does that mean some kind of auth error? |
| 12:20.51 | [TK]D-Fender | You should be doing SIP DEBUG to see whats going on. |
| 12:20.55 | [TK]D-Fender | ^^^^^^^^^ |
| 12:21.04 | IsUp | and connect your phone again |
| 12:21.09 | IsUp | try to register |
| 12:22.00 | ph8 | good call fender |
| 12:22.06 | ph8 | i appear to be getting a 401 sent to the phone |
| 12:22.11 | *** join/#asterisk nikolaev (n=nikolaev@78.83.145.98) |
| 12:22.22 | ph8 | after a couple of 100s |
| 12:22.22 | ph8 | <--- Transmitting (no NAT) to 192.168.69.100:5060 ---> |
| 12:22.22 | ph8 | SIP/2.0 100 Trying |
| 12:22.23 | ph8 | those |
| 12:22.32 | ph8 | is asterisk having trouble communicating with it because it's behind a router? |
| 12:22.42 | ph8 | i assumed the phone would establish a connection and maintain it |
| 12:22.59 | IsUp | stop asterisk |
| 12:23.01 | IsUp | type |
| 12:23.05 | IsUp | nc -vvlp 5060 |
| 12:23.08 | IsUp | on your pbx server |
| 12:23.12 | IsUp | and then try to connect with |
| 12:23.17 | IsUp | telnet yourpbxserverip 5060 |
| 12:23.17 | Nugget | telnet is eeeeeeevil! |
| 12:23.26 | IsUp | telnet is cool |
| 12:23.28 | IsUp | :> |
| 12:23.33 | ph8 | works |
| 12:23.45 | [TK]D-Fender | .... PASTEBIN <- |
| 12:23.53 | ph8 | lol tell me you're kidding |
| 12:24.02 | [TK]D-Fender | IsUp: SMRT |
| 12:24.03 | ph8 | in the ssh terminal where i'm running netcat i see 'foo' which i typed in my telnet sesh |
| 12:24.10 | ph8 | or you mean pastebin my config? |
| 12:24.22 | [TK]D-Fender | ph8: your SIP DEBUG |
| 12:24.23 | IsUp | [TK]D-Fender: smrt? |
| 12:24.49 | [TK]D-Fender | IsUp: Homer Simpson reference |
| 12:24.57 | ph8 | http://pastebin.com/m6335961d |
| 12:24.58 | IsUp | d'oh |
| 12:25.08 | [TK]D-Fender | IsUp>telnet yourpbxserverip 5060 <- SIP is... **UDP** |
| 12:25.28 | IsUp | got it |
| 12:25.34 | florz | [TK]D-Fender: No, SIP is TCP _AND_ UDP =:-> |
| 12:25.40 | IsUp | btw, Grandstream is nice |
| 12:25.42 | IsUp | i am using too |
| 12:25.57 | ph8 | thx, it was recommended ;) |
| 12:26.19 | [TK]D-Fender | florz: Not in * yet it isn't. |
| 12:26.26 | ph8 | so what do you think's wrong? do i need to be forwarding ports to my phone? |
| 12:26.29 | nikolaev | Is there a way to run a prepaid billing on the asterisk PBX extensions? |
| 12:26.47 | florz | [TK]D-Fender: yeah, sure - which doesn't make your statement true ... |
| 12:26.52 | IsUp | i can't get it ph8, your pbx server on your LAN? and your phone? |
| 12:26.57 | ph8 | nikolaev: Presumably you could pipe the call logs to something? |
| 12:26.57 | [TK]D-Fender | ph8: SIP/2.0 401 Unauthorized <- phone is set wrong, and <--- Transmitting (no NAT) to 192.168.69.100:5060 ---> <- looks like your system is not correctly set up for NAT. Go read the GUIDE. |
| 12:26.59 | [TK]D-Fender | ~sipnat |
| 12:27.00 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 12:27.02 | [TK]D-Fender | ^^^^^ |
| 12:27.10 | ph8 | PBX server i s on the internet (datacentre), phone is in my house behind a router IsUp |
| 12:27.24 | ph8 | ooo |
| 12:27.25 | ph8 | guidefs |
| 12:27.27 | ph8 | * guides |
| 12:27.42 | IsUp | ah |
| 12:27.45 | IsUp | got it |
| 12:28.00 | IsUp | try DMZ on your router |
| 12:28.02 | IsUp | to your phone |
| 12:28.04 | nikolaev | ph8: correct. but I am wondering about if there is something already developed for this purposes |
| 12:28.19 | ph8 | [TK]D-Fender: That's not quite my situation (the guide) |
| 12:28.24 | ph8 | my server's public but my phone's natt'ed |
| 12:28.26 | ph8 | i'll try dmz isup |
| 12:28.31 | IsUp | ok |
| 12:28.34 | [TK]D-Fender | ph8: No. |
| 12:28.40 | [TK]D-Fender | phGo follow the GUIDE <---- |
| 12:28.42 | *** join/#asterisk bbryant (n=Brett_Br@adsl-212-77-233.chs.bellsouth.net) |
| 12:28.49 | ph8 | the one you just linked me to is wrong? (aocomputing) |
| 12:28.59 | [TK]D-Fender | phNo, it isn't |
| 12:28.59 | IsUp | Fender |
| 12:29.03 | IsUp | calm down |
| 12:30.06 | ph8 | [TK]D-Fender: So i put nat=yes and externip type lines inside my [Ph0] context? |
| 12:30.39 | [TK]D-Fender | ph8: "nat=yes" , "canreinvite=no", "qualify=yes". |
| 12:30.54 | ph8 | sexy |
| 12:30.56 | ph8 | no externip stuff |
| 12:31.00 | ph8 | ok lets try that |
| 12:31.28 | ph8 | woo |
| 12:31.31 | ph8 | humps [TK]D-Fender |
| 12:31.44 | [TK]D-Fender | ph8: If your * is on a public IP, no, you should not need it. That also only belongs under [general]. It is not for anywhere else |
| 12:32.39 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:32.39 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 12:33.09 | ph8 | erm |
| 12:33.15 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 12:33.18 | ph8 | i just put what you told me under my [Ph0] context and that works now |
| 12:33.22 | ph8 | my * is public, my phone is natt'ed |
| 12:33.37 | [TK]D-Fender | ph8: All covered in the guide. |
| 12:33.51 | ph8 | the guide discusses a natted server and a public phone... |
| 12:34.12 | [TK]D-Fender | ph8: shows what to do for * behind, remote phone behind, and ITSP's / remote *'s you don't need to accomodate |
| 12:34.29 | ph8 | ah i was looking at the aocomputing one, not the wiki |
| 12:35.37 | [TK]D-Fender | phI'm referring to the aocomputing one. You just saw one little thing MORE than you needed and wrote it off |
| 12:36.37 | [TK]D-Fender | ph8: Sorry it didn't have a speceific header with a section that described only your situation, but you really will have to learn to read. |
| 12:36.48 | ph8 | i looked through it and was almost there when you gave me the lines |
| 12:36.55 | ph8 | i would have tried putting externip in though |
| 12:37.08 | Andre101 | anyone know a good skype-sip gateway? |
| 12:37.43 | lmadsen | Andre101: it doesn't really exist because skype is a proprietary protocol |
| 12:38.14 | [TK]D-Fender | ph8: Stop trying random stuff and read the details. You were about to shove [general] - only parameters into your peers. The WIKI has a list of whats applicable where. |
| 12:38.34 | rbd_ | hi guys. still having a weird problem with asterisk trying to use /var/run/ instead of /var/run/asterisk ...totally stumped here...see http://pastebin.com/d30b90bce |
| 12:38.58 | *** join/#asterisk EmleyMoor (i=phil@topdeck.tinsleyviaduct.com) |
| 12:39.05 | *** join/#asterisk Levonk (n=lk@adsl-76-227-117-238.dsl.lsan03.sbcglobal.net) |
| 12:39.27 | ph8 | [TK]D-Fender: Well, thanks |
| 12:39.45 | [TK]D-Fender | ph8: You're welcome. |
| 12:39.51 | *** join/#asterisk JenniferAkemi (n=akemi@c207.134.168-110.clta.globetrotter.net) |
| 12:40.19 | rbd_ | looks like it sees the asterisk.conf file fine... I have /var/run/asterisk in there ... asterisk has access to /var/run/asterisk no problem, etc... this is a source compiled version of 1.4.21.2 |
| 12:40.23 | EmleyMoor | zaptel modules won't build against kernel 2.6.26 on Debian - how can I solve this? |
| 12:41.01 | lmadsen | EmleyMoor: use an older kernel? |
| 12:41.07 | lmadsen | there is a bug on the bug tracker right now for it |
| 12:41.51 | lmadsen | EmleyMoor: http://bugs.digium.com/view.php?id=13308 |
| 12:42.03 | EmleyMoor | lmadsen: I can limp on with what I've got for now but there is only that kernel in lenny - I will have to wait a bit, I guess |
| 12:42.20 | lmadsen | feel free to comment on that bug if you have anything further to add |
| 12:42.42 | *** part/#asterisk lirakis (n=lirakis@65.200.191.241) |
| 12:43.13 | EmleyMoor | Nothing more on that one... |
| 12:43.14 | *** join/#asterisk MrNaz (n=naz@122.110.179.52) |
| 12:43.32 | *** join/#asterisk DarylVOIP (n=daryl@75.147.121.177) |
| 12:44.32 | *** join/#asterisk MrNaz (n=naz@122.110.179.52) |
| 12:44.38 | EmleyMoor | I think for safety reasons I need to select a different default kernel on that machine for now |
| 12:48.39 | rbd_ | re my problem, figured it out, I was using an old asterisk.conf from an old deb install, not the one made when you do `make samples` |
| 12:49.44 | WimpMan | So it did indeed not read the config. |
| 12:50.14 | tzafrir_laptop | EmleyMoor, use SVN |
| 12:50.24 | tzafrir_laptop | EmleyMoor, what version do you use? |
| 12:56.23 | EmleyMoor | tzafrir_laptop: At present, that which Debian supplies - 1.4.11 |
| 12:56.42 | tzafrir_laptop | ok |
| 12:56.53 | x86 | ouch, 1.4.11 is oldschool |
| 12:57.09 | EmleyMoor | Got slightly more pressing matters right now like asterisk itself needing a patch |
| 12:57.10 | x86 | I don't have much room to talk though, I run 1.4.12.1 on 7 production servers here at work |
| 12:57.31 | x86 | EmleyMoor: always best to compile it yourself |
| 12:59.13 | ph8 | my reseller offers an IAX outbound or a SIP outbound trunk - which one do i want? If I take SIP can my grandstream hardphone connect directly to it? rather than putting additional load on the * install? |
| 12:59.18 | ph8 | Is that even recommended? |
| 13:00.10 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 13:00.46 | EmleyMoor | ph8: Leaving aside the advisability or otherwise of GrandStream, if you want to use asterisk, you wouldn't want to connect the phone directly to it, as a rule. |
| 13:01.07 | EmleyMoor | (to the provider, that is) |
| 13:01.24 | ph8 | i see |
| 13:01.34 | ph8 | in that case, is a SIP trunk or IAX trunk better one way or the other for asterisk? |
| 13:01.53 | ph8 | from what i've read IAX seems to be pretty clever |
| 13:01.59 | EmleyMoor | IAX is a more native protocol as far as Asterisk goes, and is what I would go for |
| 13:02.26 | ph8 | thx |
| 13:03.15 | EmleyMoor | My provider allows me to use either IAX or SIP - I tend to do IAX first then try SIP if it fails |
| 13:03.17 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 13:03.28 | EmleyMoor | Incoming, I choose IAX |
| 13:03.49 | ph8 | I think i have SIP incoming |
| 13:04.01 | ph8 | i'm buying off gradwell.com |
| 13:04.04 | EmleyMoor | Asterisk is fine with it |
| 13:04.15 | ph8 | i could have chosen iax mind |
| 13:04.16 | iCEBrkr | I've always been a fan of IAX for linking 2 or more asterisk boxes. But other than that, it's SIP all the way :P |
| 13:05.38 | iCEBrkr | heads into work |
| 13:05.39 | iCEBrkr | & |
| 13:05.50 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:06.03 | EmleyMoor | The beauty of Asterisk is it can do just about anything you're likely to want |
| 13:06.30 | EmleyMoor | (except make the tea, as my partner keeps suggesting) |
| 13:06.43 | *** join/#asterisk poller (n=poller@poller.se) |
| 13:06.47 | poller | Hello |
| 13:07.00 | lmadsen | EmleyMoor: it probably could if you got a compatible tea maker :) |
| 13:07.02 | poller | Is there any way to trigger some kind of shell script on the server when peers registers? |
| 13:07.14 | lmadsen | poller: not that I'm aware of... |
| 13:07.16 | EmleyMoor | lmadsen: Yes... |
| 13:07.24 | poller | Maybe some log file I can parse, hmm |
| 13:07.39 | EmleyMoor | I plan one day to X10 most of my house - and maybe even have that linked to asterisk |
| 13:07.49 | brodiem | EmleyMoor: I did that :) |
| 13:08.03 | WimpMan | poller: I'd go for ami. |
| 13:08.12 | poller | ami? |
| 13:08.20 | lmadsen | Asterisk Manager Interface |
| 13:08.25 | lmadsen | that does seem to make sense |
| 13:08.45 | lmadsen | assuming there is an event triggered upon registration. I don't use the AMI nearly enough |
| 13:09.03 | WimpMan | IIRC it is. |
| 13:09.05 | [TK]D-Fender | brodiem: Ditto |
| 13:09.09 | poller | I want to keep track of registrations, for statistics purpose |
| 13:09.14 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 13:10.26 | EmleyMoor | brodiem: The possibilities it would bring - make a phone call from the motorway and the heating will be warmed up by the time you get home... |
| 13:10.27 | brodiem | [TK]D-Fender: what X10 interface are you using? |
| 13:10.57 | [TK]D-Fender | brodiem: CM11A |
| 13:11.13 | brodiem | EmleyMoor: since I go home to Canada for vacation a lot it comes in useful when I can have an IVR to control stuff, also useful for viewing streamed video from the cameras, since I can change cameras while watching |
| 13:11.20 | [TK]D-Fender | brodiem: Old classic, and "heya2" for CLI access. Unfortunately no 2-way modules.. |
| 13:11.44 | *** join/#asterisk gbr_ (n=gbr@200.103.96.98) |
| 13:11.59 | brodiem | [TK]D-Fender: you mean heyu? |
| 13:12.00 | *** join/#asterisk Nate187 (n=Nate187@gw.bigrivertel.net) |
| 13:12.07 | EmleyMoor | I will have to see what is available that I could use... |
| 13:12.25 | [TK]D-Fender | brodiem: It was updated to "heyu2" , yeah I typo'd |
| 13:12.27 | EmleyMoor | I plan to think about this in the new house as and when I've got one |
| 13:12.37 | [TK]D-Fender | brodiem: I didn't reinstall it all since my last move however |
| 13:14.01 | brodiem | [TK]D-Fender: ah ok, I been using one of the CM19A (USB), where it just creates a /dev/cm19a0 device, where you just cat text and redirect to the device to control devices |
| 13:14.36 | brodiem | er that came out messed up, but anyway |
| 13:15.13 | [TK]D-Fender | brodiem: No, was perfectly readable. I remember the USB one was undocumented and X-10 wasn't too friendly about it. We'v got a usable stack now? |
| 13:15.34 | brodiem | [TK]D-Fender: Yep, and the linux driver works way better than the windows version :) |
| 13:15.46 | brodiem | let me get the link |
| 13:16.37 | ph8 | hmm |
| 13:16.56 | brodiem | in fact the VA11A video USB device works way better in linux too |
| 13:17.07 | ph8 | i've configured my IAX2 outbound but whenever i dial all i get is a busy signal IAX2/193.111.201.98:4569-2 is circuit-busy ABD |
| 13:17.09 | ph8 | erm |
| 13:17.11 | ph8 | AND Everyone is busy/congested at this time (1:0/1/0) |
| 13:17.11 | ph8 | <PROTECTED> |
| 13:17.13 | ph8 | is what i meant to add |
| 13:17.14 | ph8 | sorry for spam |
| 13:17.24 | *** join/#asterisk tobias (n=tobias@user-0c2hj2e.cable.mindspring.com) |
| 13:17.24 | ph8 | is that a fault on behalf of my provider/ |
| 13:18.34 | brodiem | [TK]D-Fender: http://lemaymd.com/main.php?frag=software |
| 13:19.00 | brodiem | [TK]D-Fender: doesn't actually need to be used as an XMMS front-end like it may sound like |
| 13:19.00 | x86 | ph8: show us your Dial string |
| 13:19.18 | ph8 | just spotted something from the book :p sec |
| 13:19.19 | [TK]D-Fender | brodiem: S'ok, you can rest assured I'd read it in full :) |
| 13:19.32 | [TK]D-Fender | brodiem: Bookmarked to take home.. |
| 13:19.32 | brodiem | hehe |
| 13:20.23 | ph8 | I wasn't stripping out the 9 for outside line with EXTEN:1 and the result was a busy signal |
| 13:20.33 | *** join/#asterisk ElSonico (n=tav@nat/ibm/x-a0f4aa590bf7f170) |
| 13:20.52 | ph8 | so if i dial a number that does not exist, i get a busy signal? |
| 13:20.54 | *** join/#asterisk mcab (n=mb@mostly-harmless.ca) |
| 13:22.18 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 13:24.04 | [TK]D-Fender | ph8: That is a generic message. go look at real debug to find out what actually happened |
| 13:26.57 | ph8 | [TK]D-Fender: If the number's unknown i dial and my IAX2 outbound provider returns circuit-busy it appears |
| 13:27.00 | ph8 | i've got that verbose 10 debug 3 on |
| 13:27.39 | [TK]D-Fender | phthat is NOT debug. I thought you'd have learned your lesson what I told you to look at the SIP DEBUG for your phone issue. |
| 13:28.47 | *** join/#asterisk kkjoe (n=opera@217.7.238.226) |
| 13:29.05 | kkjoe | is there any irc channel for spandsp rxfax problems ? |
| 13:35.04 | ph8 | [TK]D-Fender: iax debug? :p |
| 13:35.19 | ph8 | doesn't exist |
| 13:35.30 | [TK]D-Fender | ph8: "iax2 debug" <--- |
| 13:35.43 | ph8 | *CLI> iax2 debug |
| 13:35.43 | ph8 | No such command 'iax2 debug' |
| 13:36.55 | *** join/#asterisk mog (n=mog@nat/digium/x-ad3699e575f6eab7) |
| 13:36.55 | *** mode/#asterisk [+o mog] by ChanServ |
| 13:37.32 | *** join/#asterisk Levonk (n=lk@adsl-75-62-142-142.dsl.lsan03.sbcglobal.net) |
| 13:37.51 | *** join/#asterisk sambalbij (n=ivo@vpn1.nextpertise.nl) |
| 13:39.27 | sambalbij | Hi, anyone got some good T.38 faxing experiences with asterisk? I have 1 carrier working and 1 not, but the one which isn't working isn't giving any problems when i connect a ata box directly. Anyone who can give me some directions to look at? |
| 13:42.35 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 13:44.08 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 13:44.21 | puzzled | hi |
| 13:44.57 | *** join/#asterisk LeddyHM (i=leddy@da-club.with.my.beerandcondoms.com) |
| 13:47.02 | *** join/#asterisk scampbell (n=scampbel@mail.scampbell.net) |
| 13:48.28 | puzzled | I need to use _X but only for 0-5 and 7-9. anyone recall how that was done? is it [1-5,7-9] ? |
| 13:48.39 | *** join/#asterisk ddunavant (n=David@75.145.240.14) |
| 13:49.58 | sambalbij | andere partij niet, maar als ik ata box rechtstreeks gebruikt richting hun werkt het wel |
| 13:50.36 | lmadsen | puzzled: remove the comma |
| 13:50.41 | lmadsen | [1-57-9] |
| 13:51.15 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 13:51.27 | *** join/#asterisk Odd_Bloke (n=oddbloke@daniel-watkins.co.uk) |
| 13:51.50 | puzzled | lmadsen: thanks |
| 13:52.08 | Odd_Bloke | Hello all, we're looking to store SIP registrations in a MySQL database. Is this possible, and can someone direct me to where I can learn more about it? |
| 13:52.34 | [TK]D-Fender | Odd_Bloke: ... |
| 13:52.37 | [TK]D-Fender | ~book |
| 13:52.37 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 13:52.38 | [TK]D-Fender | ^^^^ |
| 13:52.43 | [TK]D-Fender | Odd_Bloke: its in there. |
| 13:59.19 | *** join/#asterisk irisht (n=irisht@cpe-70-122-11-142.austin.res.rr.com) |
| 14:00.45 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 14:01.07 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 14:03.34 | *** join/#asterisk s0lid (n=s0lid@219.93.152.130) |
| 14:03.54 | *** join/#asterisk l2trace99 (n=jr@75.112.133.235) |
| 14:05.40 | *** join/#asterisk mgroman (n=miles@adsl-68-252-62-180.dsl.wotnoh.ameritech.net) |
| 14:06.49 | *** join/#asterisk [xpl] (i=xpl@84.126.207.62.dyn.user.ono.com) |
| 14:08.41 | synthetiq | for jabber/jingle stuff in asterisk, is it possible for users to send messages to each other thru asterisk or is it jsut mainly a calling solution? i dont see how users can receive messages |
| 14:10.37 | brodiem | synthetiq: IIRC it cannot do anything with incoming messages |
| 14:12.27 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 14:12.37 | *** join/#asterisk bamsefar (i=emj@core.serv.emj.se) |
| 14:12.38 | bamsefar | Hello |
| 14:12.41 | *** part/#asterisk kkjoe (n=opera@217.7.238.226) |
| 14:14.05 | aliver | Do polycom phones always look for the same files when they boot and are configured for tftp? |
| 14:14.21 | bamsefar | I need to execute a script upon successful registration of a sip peer, any suggestions? |
| 14:14.32 | *** join/#asterisk seanmh (i=seanmh@216.31.101.107) |
| 14:15.46 | _Krieger_ | are snoms rfc3680-compliant? |
| 14:16.13 | _Krieger_ | that`s 'reginfo' |
| 14:20.14 | *** join/#asterisk adr3nalin3 (n=afink@asa.redglaze.com) |
| 14:24.56 | *** part/#asterisk eric256 (n=ejh@72.1.122.62) |
| 14:30.10 | [TK]D-Fender | aliver: Yes |
| 14:30.32 | *** join/#asterisk jeffspeff (i=jeff@c-98-240-113-135.hsd1.ky.comcast.net) |
| 14:32.56 | *** join/#asterisk jeffspeff2 (i=jeff@c-98-240-113-135.hsd1.ky.comcast.net) |
| 14:38.45 | *** join/#asterisk eliel (n=eliel@200.61.172.61) |
| 14:39.38 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.95.130) |
| 14:40.22 | *** join/#asterisk luckyaba (n=lucky@ip68-6-98-146.sb.sd.cox.net) |
| 14:44.11 | *** join/#asterisk n3hxs (n=HAMming@151.196.87.132) |
| 14:46.09 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 14:48.21 | *** join/#asterisk VJFROMGT (n=vjfromgt@user-12lcpf0.cable.mindspring.com) |
| 14:48.42 | VJFROMGT | I want to set a custom ringback, eg a song, can this be done? |
| 14:49.04 | *** join/#asterisk spokra (n=spokra@host093-179-187.sea0.speakeasy.net) |
| 14:49.08 | [TK]D-Fender | VJFROMGT: MoH <- "core show application dial" |
| 14:49.18 | aliver | http://pastebin.com/d46ec672c <--- See that error on line 17? Any idea what could cause that when the user clearly has a config in voicemail.conf ? |
| 14:49.22 | *** join/#asterisk pigpen (n=mark@fw.seamans.cc) |
| 14:49.27 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 14:49.30 | aliver | what other files would one need to look at to figure that one out? |
| 14:49.55 | jeev | go fender, its your birthday, go fender, it's your birthday |
| 14:52.47 | brodiem | aliver: For what user? you're calling up Voicemail() without specifying a user. |
| 14:53.03 | [TK]D-Fender | aliver: - Executing VoiceMail("SIP/4.68.250.148-08371000", "@default") in new stack <-- do YOU see a box # being passed on there? |
| 14:53.10 | [TK]D-Fender | aliver: I kniow I sure don't |
| 14:53.30 | *** join/#asterisk devilsoulblack (n=aandaluz@srv.ec-gye.internet.geainternacional.com) |
| 14:53.32 | [TK]D-Fender | aliver: And I see a call being passed on from a raw IP and not even matching a peer entry |
| 14:53.33 | devilsoulblack | hi |
| 14:53.46 | [TK]D-Fender | aliver: You're shooting blanks across the board right now |
| 14:54.01 | devilsoulblack | any one know how to get the log of one days from /var/log/asterisk/full |
| 14:54.07 | brodiem | lol |
| 14:54.16 | jeev | what do you recommend for billing fender? keep in mind, 3 servers will probably end up with data because i have 2 back up servers |
| 14:54.19 | brodiem | devilsoulblack: grep |
| 14:54.26 | devilsoulblack | yes i use grep |
| 14:54.44 | [TK]D-Fender | jeev: MS Excel + CSV |
| 14:54.57 | brodiem | devilsoulblack: so what's the question? |
| 14:55.15 | *** join/#asterisk osiris (n=osiris@c-71-205-9-42.hsd1.mi.comcast.net) |
| 14:55.17 | brodiem | s/question/problem/ |
| 14:55.23 | devilsoulblack | but i used cat /var/log/asterisk/full | grep "Aug 26" > 26-08-08.txt |
| 14:55.36 | jeev | what should i use to get to csv |
| 14:55.38 | devilsoulblack | but i dont get nothing :( |
| 14:55.49 | brodiem | devilsoulblack: because it doesn't use that date format.. |
| 14:56.14 | devilsoulblack | what ist the date format ? |
| 14:56.25 | *** join/#asterisk jeffspeff (i=jeff@c-98-240-113-135.hsd1.ky.comcast.net) |
| 14:56.36 | aliver | [TK]D-Fender yeah, I noticed the problem, but I'm flummoxed as to the solution. How do I begin to troubleshoot that issue? Should I start with sip debug? I've scrutinized the voicemail.conf for blank entries and examined my sip.conf to make sure I have mailboxes set for everyone, but obviously I'm looking in the wrong spot. |
| 14:56.47 | *** part/#asterisk VJFROMGT (n=vjfromgt@user-12lcpf0.cable.mindspring.com) |
| 14:56.53 | brodiem | devilsoulblack: grep "^\[2008-08-26" /var/log/asterisk/full |
| 14:57.07 | [TK]D-Fender | aliver: - Executing VoiceMail("SIP/4.68.250.148-08371000", "@default") in new stack <-- do YOU see a box # being passed on there? No box in your DIALPLAN being called. |
| 14:57.10 | devilsoulblack | <PROTECTED> |
| 14:57.14 | devilsoulblack | mmm letme check |
| 14:57.17 | brodiem | devilsoulblack: you could also just use logrotate to rotate them daily.. |
| 14:57.22 | [TK]D-Fender | aliver: How much bigger does the neon sign have to be? |
| 14:57.34 | Kobaz | as big as they come |
| 14:57.37 | aliver | [TK]D-Fender I guess I need a laser light show sign. |
| 14:57.51 | aliver | I can see the box number is missing. |
| 14:57.55 | [TK]D-Fender | sets his laser printer on KILL and waits for a co-wrker to pass in front. |
| 14:58.15 | aliver | I'm just wonder what that "SIP/4.68.250.148-08371000" is. |
| 14:58.18 | Kobaz | [TK]D-Fender: what do you do for work anyway? |
| 14:58.26 | Kobaz | [TK]D-Fender: i see you're on irc like 24/7... heh |
| 14:58.44 | [TK]D-Fender | aliver: that is the CHANNEL of the current call. Its what tells me "I wish I could match a peer entry like everyone else" |
| 14:59.00 | [TK]D-Fender | Kobaz: IT dept for a non-tech company |
| 14:59.11 | Kobaz | ah, so plenty of slack time then |
| 15:00.17 | [TK]D-Fender | Kobaz: I multi-task well |
| 15:00.32 | aliver | [TK]D-Fender okay, that makes sense. Does that stem for this earlier line 13? |
| 15:00.33 | aliver | -- Executing Dial("SIP/4.68.250.148-08371000", "SIP/|10") in new stack |
| 15:00.44 | aliver | It looks like it's trying to dial "SIP/" which is nothing right? |
| 15:00.59 | Kobaz | aliver: correct |
| 15:01.25 | [TK]D-Fender | aliver: very broken dialplan you're calling there... |
| 15:02.21 | devilsoulblack | brodiem, i used cat /var/log/asterisk/full | grep "^\[2008-08-26" > 26-08-08.txt |
| 15:02.25 | devilsoulblack | but dont work |
| 15:02.49 | brodiem | devilsoulblack: then the text "[2008-08-26" doesn't exist in your full log. |
| 15:03.22 | *** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-3b6f2d7a9f9dd5e3) |
| 15:03.22 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:03.43 | aliver | http://pastebin.com/d6389a982 <--- here is my very broken dialplan. Fire up the flame thrower and tell me where my idiocy lies. |
| 15:04.02 | *** join/#asterisk MrNaz (n=naz@ppp118-208-159-139.lns10.mel4.internode.on.net) |
| 15:04.12 | aliver | Usually calls that come in that don't match anything will give me a 404 error. |
| 15:04.57 | devilsoulblack | brodiem, http://pastebin.ca/1187452 check |
| 15:05.47 | aliver | The bad thing is that I don't have much room to experiment with making things "nice" because once it works people get pissed if it's broken. |
| 15:06.14 | aliver | I really have grown to hate the goddamn "phone" issue. |
| 15:09.20 | [TK]D-Fender | aliver: look for your extension pattern overlap for that call and realize what is being called & overriden. |
| 15:09.34 | aliver | okay, I'll try. |
| 15:09.43 | aliver | thanks for the tip |
| 15:10.26 | [TK]D-Fender | aliver: If you want things redone properly from scratch, my rates are very accessible :) |
| 15:10.47 | *** join/#asterisk eXistenZ (n=existenz@unaffiliated/existenz) |
| 15:12.00 | aliver | [TK]D-Fender dude, if it were up to me I'd hire you in a heartbeat. |
| 15:12.13 | aliver | we'd _save_ money and avoid all my fumbling. |
| 15:12.28 | aliver | I'm a Unix systems programmer, not an asterisk expert. |
| 15:12.39 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 15:12.49 | brodiem | devilsoulblack: ok so clearly you've got different date stamps, so "Aug 28" is what you need to match.. |
| 15:14.08 | *** join/#asterisk jeffspeff2 (i=jeff@c-98-240-113-135.hsd1.ky.comcast.net) |
| 15:14.13 | brodiem | devilsoulblack: keep in mind you'll only match those lines with the date included in it |
| 15:14.43 | aliver | ${DNID} == current caller's SIP-ID ? |
| 15:14.59 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-afc3305c784dc9e3) |
| 15:15.07 | aliver | n/m |
| 15:19.48 | IsUp | i have Sangoma A104 |
| 15:20.05 | IsUp | my port 2 connected to GSM gateway |
| 15:20.19 | IsUp | my port 1 connected to telco |
| 15:20.28 | IsUp | how can i control Dial via GSM Gateway? |
| 15:20.47 | IsUp | dial(zap/2-xx/4554777) xx is chan |
| 15:20.49 | IsUp | not workin |
| 15:20.53 | IsUp | it always goes to |
| 15:20.54 | IsUp | 2-1 |
| 15:21.37 | IsUp | <PROTECTED> |
| 15:21.38 | IsUp | -- Zap/2-1 is proceeding passing it to SIP/8071-b6273ec8 |
| 15:21.49 | adr3nalin3 | Could someone please help me with setting up sip trunks asterisk to asterisk? I am using gui but if I can get the general concept I can make it work. Thanks. |
| 15:21.52 | IsUp | how can fix it? |
| 15:22.27 | aliver | I'm trying to make a rule that says "if you call your own extension, you get voicemailmain()" What should I use as a match for that? I was thinking just ._ |
| 15:22.49 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:23.31 | [TK]D-Fender | aliver: Stop making overlapping patterns. That one is also specifically a "shoot on sight" offense. Grab some coffee and think long & hard on it |
| 15:24.14 | [TK]D-Fender | IsUp: Every zaptel channel is independant. You do not specific the SPAN along with it. |
| 15:24.24 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 15:24.27 | IsUp | what you mean? |
| 15:24.31 | [TK]D-Fender | IsUp: Go look at every other Zaptel based dial sample out there. |
| 15:24.49 | [TK]D-Fender | IsUp: Part of what I'm saying is there is no "_" in your call to Dial. |
| 15:24.54 | [TK]D-Fender | "-" |
| 15:25.49 | IsUp | exten => s,1,Dial(Zap/1-2/1234) ; Dial 1234 on span 1, port 2 |
| 15:26.05 | aliver | something like "exten => ._,1,Gotoif(${CALLERID(dnid)} == ${EXTEN}:2?5)" |
| 15:26.06 | *** join/#asterisk elred (i=sauron@fucksheep.org) |
| 15:26.08 | IsUp | can't i control, span 2, port 2, channel xxx |
| 15:26.13 | IsUp | ? |
| 15:26.48 | [TK]D-Fender | IsUp: Every channel is unique. You do NOT specify the span. |
| 15:27.06 | *** join/#asterisk EI5GTB (n=EI5GTB@193.120.192.61) |
| 15:27.44 | IsUp | hmm i couldnt understand lemme explain |
| 15:27.55 | IsUp | my gsm gateway has 32 gsm slot |
| 15:28.03 | IsUp | how can i send call to specific slot? |
| 15:28.20 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 15:28.21 | IsUp | my gsm gateway connected to my sangoma with PRI |
| 15:28.24 | aliver | How can you avoid making extensions overlap. I thought extensions would match the most specific entry in the dialplan. |
| 15:28.28 | IsUp | to port 2 |
| 15:28.47 | aliver | Is the idea to use Goto() with different contexts? |
| 15:29.04 | [TK]D-Fender | IsUp: and I told you to look at how every other Zap channel is dialed. |
| 15:29.10 | aliver | I guess I'm confused about the exact meaning of "overlap" |
| 15:29.21 | *** join/#asterisk c4t3l (n=root@74.95.210.124) |
| 15:29.46 | [TK]D-Fender | IsUp: Every single channel on each port is a UNIQUE number. You do not need to specify a SPAN number for it to pick the right one. Each is UNIQUE |
| 15:29.58 | outtolunc | geez, just tell him to group dial already <G> |
| 15:30.12 | [TK]D-Fender | aliver: Exten => 123,1....... Exten => _12X,1..... OVERLAP. |
| 15:30.21 | [TK]D-Fender | outtolunc: Nah. |
| 15:30.34 | *** join/#asterisk Levonk (n=lk@adsl-75-62-137-48.dsl.lsan03.sbcglobal.net) |
| 15:31.55 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 15:32.47 | *** join/#asterisk ElSonico (n=tav@xdsl-179-7.nblnetworks.fi) |
| 15:36.07 | *** part/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 15:39.23 | *** part/#asterisk elred (i=sauron@fucksheep.org) |
| 15:39.29 | IsUp | Fender |
| 15:39.30 | IsUp | got time? |
| 15:41.35 | *** join/#asterisk Levonk (n=lk@75.62.130.219) |
| 15:42.45 | mgroman | ha |
| 15:43.11 | *** join/#asterisk aliver (n=aliver@ip-216-17-149-97.rev.frii.com) |
| 15:43.16 | aliver | [TK]D-Fender I just can't see how that's bad. For, example, using your example, what if I want 123 do do something special but I want 124-129 to just ring their phone. |
| 15:43.48 | bamsefar | Is there any way to get asterisk to log the IP address from which a peer registered? |
| 15:44.50 | aliver | [TK]D-Fender I guess what I'm saying is, in that case what is the "right" way to do it? |
| 15:45.16 | viraptor | has anyone tried app_conference on vmware environment before? (and knows how does it compare in quality/accuracy to meetme with zttest ~99.99%?) |
| 15:45.50 | *** join/#asterisk ToTo (n=ToTo@209.8.41.202) |
| 15:46.18 | Psychobilly | aliver 123,1,foo, _12[4-9],1,bar |
| 15:47.21 | *** join/#asterisk mfdavid (n=user@189.100.103.252) |
| 15:47.23 | mfdavid | hi all |
| 15:48.06 | mfdavid | sorry for the dumb question but... I configured my sip client under asterisk, but now, how can I make a phone call? (command line is preferable) |
| 15:48.12 | *** join/#asterisk smurf (n=smurf@debian/developer/smurf) |
| 15:48.50 | lmadsen | mfdavid: 'dial' if you have chan_oss or chan_alsa loaded |
| 15:50.09 | aliver | Psychobilly okay, that's cool, but isn't that still "overlap" ? |
| 15:50.12 | [TK]D-Fender | mfdavid: Configure some way for * to reach the PSTN and set up your dialplan so it can take a call from your SIP client and then dial out that PSTN connection. |
| 15:50.26 | aliver | Psychobilly wait n/m I see. |
| 15:50.31 | Psychobilly | aliver not at all |
| 15:50.38 | aliver | I gotcha. |
| 15:50.48 | aliver | I think I see now. |
| 15:51.03 | aliver | You want to avoid overlap since it makes debugging such a PITA. |
| 15:51.13 | aliver | even if you have to write extra, more specific stuff. |
| 15:52.02 | mfdavid | thanks! |
| 15:54.51 | aliver | However, in my case I have a block of like 200 numbers all ending with 7XX. Now some "important" people want special shit like having their internal numbers ring their cell phone after 2 rings internall. Are you saying I still shouldn't have a _7XX pattern if it overlaps with 5 people out of 200? |
| 15:56.27 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 15:57.07 | Psychobilly | aliver there are other ways to provide services like call forward etc in ur dialplan |
| 15:59.19 | *** join/#asterisk fnordus (n=dnall@70.71.225.48) |
| 15:59.23 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.157) |
| 16:05.08 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 16:07.01 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 16:08.29 | mfdavid | When I try to dial to a phone, I get this: No such extension '118786XXXX' in context 'default'. Any tips? |
| 16:09.06 | DarylVOIP | starting with the obvious....do that extension exist int he default context? |
| 16:10.07 | mfdavid | thats an external phone number. Dont know how to setup that =( |
| 16:10.36 | EI5GTB | whats the cheapest ata's money can buy? |
| 16:10.37 | mocker | mfdavid: It still has to be an extension you can dial. |
| 16:10.54 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 16:11.01 | Kobaz | hmmm |
| 16:11.07 | Kobaz | i'm playing with unloading and loading chan_zap |
| 16:11.11 | DarylVOIP | Ahhh...yeah..need t be an extension, or a wildcard most reasonably, that will go to your carrier/term device. |
| 16:11.14 | *** part/#asterisk mgroman (n=miles@adsl-68-252-62-180.dsl.wotnoh.ameritech.net) |
| 16:11.22 | Kobaz | asterisk crashed after doing a series of loads and unloads of chan_zap |
| 16:11.33 | Kobaz | i should compile with debugging symbols |
| 16:12.04 | Kobaz | Aug 28 12:10:30 pbx-b kernel: asterisk[15964]: segfault at 10 ip b710e7f5 sp b73b3cd0 error 4 in libpri.so.1.0[b7106000+21000] |
| 16:12.20 | tzafrir_laptop | Kobaz, what versions? |
| 16:12.28 | Kobaz | 1.4.21.2 |
| 16:12.44 | Kobaz | i have an older libpri, i'll try a new one |
| 16:13.16 | tzafrir_laptop | any chance you could test SVN? chan_zap^Wchan_dahdi has gone through changes there |
| 16:13.30 | Kobaz | yeah |
| 16:13.44 | eric_hill | I just had an asterisk box go unresponsive. It was pingable, but none of the services were responsive. The console wouldn't accept any keypresses either. Any thoughts on where to look? |
| 16:14.01 | Kobaz | eric_hill: i've seen asterisk start chewing 100% cpu and bring down the box |
| 16:14.14 | mocker | eric_hill: kernel panic? |
| 16:14.17 | Kobaz | eric_hill: run top, and try to replicate what caused it to lock up |
| 16:14.37 | eric_hill | I've seen 100% a few times, but the box was still responsive... It didn't look like a kernel panic. The console was still sitting at a login prompt. |
| 16:15.23 | eric_hill | I've never seen linux hang like that. I've always seen a kernel panic or other nastiness. |
| 16:15.27 | Kobaz | i'm running libpri 1.4.1 |
| 16:16.14 | Kobaz | [Aug 28 12:16:07] ERROR[17340]: chan_zap.c:9469 start_pri: Unable to open D-channel 48 (Device or resource busy) |
| 16:16.17 | Kobaz | [Aug 28 12:16:07] ERROR[17340]: chan_zap.c:11394 setup_zap: Unable to start D-channel on span 2 |
| 16:16.21 | Kobaz | i still get that... after doing an unload/load |
| 16:17.03 | Kobaz | zap is completely broken after unloading/loading |
| 16:17.30 | FabiOne | I've the same issue @ my home with a x100p clone |
| 16:18.29 | IsUp | Kobaz |
| 16:18.29 | Kobaz | i've got a sangoma 102d |
| 16:18.33 | IsUp | try |
| 16:18.36 | IsUp | wanrouter stop |
| 16:18.38 | IsUp | and then |
| 16:18.41 | IsUp | wanrouter start |
| 16:18.45 | IsUp | wait 10 sec |
| 16:18.48 | IsUp | and then |
| 16:18.50 | outtolunc | kobaz, stop trying to have channel = x-48 in zapata.conf make it x-47 |
| 16:18.51 | IsUp | ztcfg -vvv |
| 16:19.02 | outtolunc | you do not 'use' the d chans |
| 16:19.31 | Kobaz | oh, i have another card in here |
| 16:19.40 | Kobaz | the t1's arent 1-48 |
| 16:19.55 | c4t3l | E1 or t1 |
| 16:20.07 | IsUp | Kobaz |
| 16:20.16 | IsUp | wanrouter stop; wanrouter start |
| 16:20.19 | IsUp | wait 10 sec and then |
| 16:20.21 | IsUp | ztcfg -vvv |
| 16:20.24 | Kobaz | yeap |
| 16:20.26 | Kobaz | sec |
| 16:20.33 | outtolunc | kobaz: you still need to 'skip over' that channel as it is defined in zaptel |
| 16:20.41 | outtolunc | so skip over it in zapata |
| 16:20.44 | Kobaz | IsUp: t1:pri |
| 16:20.50 | IsUp | ss7? |
| 16:20.51 | Kobaz | outtolunc: oh i know, i do |
| 16:21.01 | Kobaz | b8zs esf |
| 16:22.20 | IsUp | <PROTECTED> |
| 16:22.25 | IsUp | [Aug 28 19:21:54] WARNING[12027]: app_dial.c:1106 dial_exec_full: Unable to create channel of type 'ZAP' (cause 0 - Unknown) |
| 16:22.28 | IsUp | but when i type |
| 16:22.31 | IsUp | zap show channels |
| 16:22.33 | IsUp | i can see 40 |
| 16:22.39 | IsUp | why is that |
| 16:23.03 | IsUp | also i have same problem on channel 39 too |
| 16:23.14 | IsUp | but 38 works fine |
| 16:23.18 | IsUp | also 37 too |
| 16:23.22 | IsUp | and 41 too.. |
| 16:24.11 | IsUp | any ideas? |
| 16:24.23 | *** join/#asterisk Maliuta (n=nikolai@kiev.lusan.id.au) |
| 16:24.55 | *** join/#asterisk hfb (n=hfb@pool-96-247-108-198.lsanca.dsl-w.verizon.net) |
| 16:25.56 | *** join/#asterisk pta200 (n=paolo@dsl211-220-225.nyc1.dsl.speakeasy.net) |
| 16:27.00 | pta200 | is there an application or function inside asterisk to check if a file exists on the server filesystem or is it something you have to write as an AGI script? |
| 16:31.39 | *** join/#asterisk eliel (n=eliel@200.61.172.61) |
| 16:33.02 | *** join/#asterisk mvanbaak_ (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 16:33.19 | [TK]D-Fender | pta200: "core show function STAT" |
| 16:35.29 | pta200 | thank you |
| 16:36.33 | *** join/#asterisk Levonk (n=lk@76.227.116.156) |
| 16:36.46 | Kobaz | okay i'm back |
| 16:37.10 | Kobaz | pbx-b*CLI> module unload chan_zap.so |
| 16:37.13 | Kobaz | pbx-b*CLI> |
| 16:37.13 | Kobaz | Disconnected from Asterisk server |
| 16:38.10 | Kobaz | itsup: i can't stop wanrouter: 16 - Device or resource busy |
| 16:38.26 | Kobaz | where'd he go |
| 16:39.32 | *** part/#asterisk pta200 (n=paolo@dsl211-220-225.nyc1.dsl.speakeasy.net) |
| 16:40.15 | *** join/#asterisk Sargun (n=xbmodder@atarack/staff/sargun) |
| 16:42.33 | *** join/#asterisk r0land (n=roland@193.227.191.91) |
| 16:42.36 | r0land | hello all |
| 16:42.39 | r0land | hi [TK]D-Fender |
| 16:43.05 | r0land | [TK]D-Fender i read throughout the book and i didnt find the solution for this! so this time you cant say "ppl are too lazy to learn, so they come and ask instead" |
| 16:43.17 | r0land | i have a lil prob... |
| 16:43.47 | r0land | my calls are locally overlapping... |
| 16:43.49 | r0land | any idea?! |
| 16:44.26 | Kobaz | you'll need to elaborate |
| 16:45.11 | r0land | Kobaz i have the following topology <softphone> --> <asterisk> --> sipura3102 |
| 16:45.55 | r0land | if i call one sipura, and i recieve another on the other sipura.. the call overlaps! i could hear the other person talking! and if they close the line.. my call hangs up as well |
| 16:46.09 | r0land | if this helps..! i have both of my sipura's config in the same context |
| 16:46.49 | *** join/#asterisk JenniferAkemi (n=akemi@c207.134.168-110.clta.globetrotter.net) |
| 16:48.49 | viraptor | is there any point in setting audiobuffers > 2 in meetme.conf if calls go into meetme through dejittering "Local/.../nj" channel first? |
| 16:51.04 | Kobaz | oh, that's quite strange |
| 16:51.22 | Kobaz | you have two supura's? |
| 16:54.04 | *** join/#asterisk Maliuta (n=nikolai@kiev.lusan.id.au) |
| 16:54.37 | r0land | yes |
| 16:54.39 | r0land | Kobaz yep :) |
| 16:55.11 | Kobaz | so you're placing calls to both, from different phones? |
| 16:55.39 | Kobaz | or from the soft phone, using two lines, one to each sipura? |
| 16:55.50 | r0land | Kobaz today's incident occured when i called from one softphone>>sipura#1 and my wife recieved a call on sipura#2 which ringed on her softphone |
| 16:56.02 | r0land | <Kobaz> or from the soft phone, using two lines, one to each sipura? |
| 16:56.37 | r0land | while i was talking.. i heard my wife recieving her call and heard her talking.. when i asked her to hang up! she did.. but tht caused my line to break as well |
| 16:57.04 | Kobaz | hmm |
| 16:57.22 | *** join/#asterisk fasttrack (n=joeljose@218.248.68.63) |
| 16:57.35 | Kobaz | and that always happens? |
| 16:57.44 | r0land | happened 4 times today |
| 16:58.20 | *** join/#asterisk CanWood (n=chatzill@24.108.64.80) |
| 16:58.21 | Kobaz | do you have some sort of wiggity dialplan that auto places the calls into a conference or something |
| 16:58.35 | fasttrack | hy all |
| 16:59.28 | r0land | Kobaz not at all |
| 17:04.11 | IsUp | does zapata.conf support channels like |
| 17:04.13 | IsUp | channel => 63,68-70 |
| 17:04.18 | IsUp | or |
| 17:04.18 | IsUp | channel => 45-46,48-60 |
| 17:04.19 | IsUp | ? |
| 17:04.41 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 17:04.41 | fasttrack | i was trying to set up a simulated lab for my voip as depicted in http://www.blindhog.net/how-to-build-a-voice-lab-with-gns3-and-vmware/. I am running into initial prob with softphones.. I got the gns3 up and ready. but when using the sJphone for linux, i suddenly got intimidated by this prob. I invoked 2 instances of the softphone, one as root and other as normal user. i can call from root to normal user but not vice versa.. when i do that, the norma |
| 17:05.00 | Kobaz | r0land: that really makes no sense at all, the only thing i can think of is just looking at the sip debug |
| 17:05.15 | Kobaz | IsUp: yeah |
| 17:05.35 | IsUp | Kobaz |
| 17:05.40 | IsUp | i am using sangoma too |
| 17:06.01 | IsUp | my port 2 connected to my GSM Gateway (Stargate) |
| 17:06.06 | Kobaz | i've always been interested in being able to change analog/t1 properties without having to restart asterisk |
| 17:06.13 | Kobaz | and it just doesn't work |
| 17:06.32 | IsUp | i am unable to use port 40 and 39 |
| 17:06.44 | Kobaz | that's odd |
| 17:06.56 | IsUp | and unable to use 47 too |
| 17:07.04 | IsUp | but 47 defined in zaptel |
| 17:07.04 | Kobaz | what happens on those channels? |
| 17:07.06 | IsUp | bchan=32-46,48-62 _ dchan=47 |
| 17:07.13 | IsUp | 47 is not working, thats normal |
| 17:07.16 | IsUp | it's defined in zaptel but |
| 17:07.23 | IsUp | 40 or 39.. its strange |
| 17:07.28 | IsUp | 38 works 37 works fine |
| 17:07.36 | IsUp | when i am trying to call |
| 17:07.39 | IsUp | lemme see |
| 17:07.53 | IsUp | <PROTECTED> |
| 17:07.57 | Kobaz | so you have a quad span or something? or do you have another card in there that's using the lower channels? |
| 17:07.58 | IsUp | [Aug 28 20:07:38] WARNING[14192]: app_dial.c:1106 dial_exec_full: Unable to create channel of type 'ZAP' (cause 0 - Unknown) |
| 17:08.02 | IsUp | <PROTECTED> |
| 17:08.16 | *** join/#asterisk matt_keys (i=smatt@usaregs.com) |
| 17:08.17 | IsUp | i have a card too |
| 17:08.40 | fasttrack | guys..... any help. please! |
| 17:08.42 | mfdavid | << Hangup on console >> :(:(:( |
| 17:09.22 | IsUp | Kobaz how can i list my cards? i forgot the command |
| 17:09.27 | kaldemar | fasttrack: what's that got to do with asterisk? |
| 17:09.40 | fasttrack | one thing in common |
| 17:09.45 | fasttrack | VOIP |
| 17:10.20 | Kobaz | zttool |
| 17:10.34 | IsUp | zttool is not installed |
| 17:10.39 | IsUp | its not in zaptel source dir too |
| 17:10.40 | fasttrack | the channel "#voip" is so inactive. its given in the title that asterisk has more people. |
| 17:10.45 | IsUp | there was another command to see it |
| 17:10.48 | IsUp | umm |
| 17:10.57 | kaldemar | fasttrack: first of all, your problem was cut in "when i do that, the norma". and, you're horribly mistaken if you think all servers work alike. |
| 17:10.57 | Kobaz | ztcfg? |
| 17:11.07 | IsUp | ah |
| 17:11.10 | IsUp | zap show status |
| 17:11.24 | IsUp | wanpipe1 card 0, wanpipe1 card 1 and Wildcard TDM2400P Board 1 |
| 17:11.40 | IsUp | i have Sangoma A104 and Wildcard TDM2400P Board 1 |
| 17:11.47 | Kobaz | k |
| 17:11.59 | fasttrack | when i do that, the normal user just dials into itself ;(. And the other prob is.. how do we talk b/w these 2 phones? i have a laptop, with a "headphone cum mic" as well as "inbuilt mic and speaker".. but that makes it still just one mic and speaker port right? ;( |
| 17:12.36 | IsUp | any ideas Kobaz? |
| 17:12.52 | fasttrack | kaldemar: i dont have any servers up yet |
| 17:13.16 | *** join/#asterisk arpu (n=arpu@chello080109017114.12.14.vie.surfer.at) |
| 17:13.28 | fasttrack | i just want to know the logic. |
| 17:13.40 | matt_keys | i ran yum update two days ago but didn't reboot until this morning. it hosed up wanpipe. i tried to run the setup script again and it says completed but it's throwing a bunch of zt symbol errors and won't load the module. here's the pastebin: http://pastebin.com/d19dca245 |
| 17:13.59 | kaldemar | fasttrack: well stop messing around with two softphones on the same computer and try something useful like using two computers. |
| 17:14.14 | fasttrack | how can u run multiple instance of a softphone? how do u speak and hear the conv b/w them if u have just one mic and speaker port? |
| 17:14.27 | fasttrack | k |
| 17:14.32 | kaldemar | you don't. |
| 17:14.42 | fasttrack | would vmware help? |
| 17:14.46 | viraptor | fasttrack: you don't run multiple instances - 99% of softphones will not support it |
| 17:14.50 | kaldemar | forget it. |
| 17:14.56 | TJNII | Looks like someone's unclear on the concepts.... |
| 17:15.01 | fasttrack | http://www.blindhog.net/how-to-build-a-voice-lab-with-gns3-and-vmware/ |
| 17:15.14 | kaldemar | seriously, don't waste your time with that. |
| 17:15.26 | fasttrack | he uses just one computer |
| 17:15.39 | fasttrack | he got ccie(voice) ;( |
| 17:15.53 | kaldemar | go ask him then. :) this is an asterisk channel. |
| 17:16.25 | *** join/#asterisk oilinki3 (n=oil@ppp-124-120-19-243.revip2.asianet.co.th) |
| 17:16.30 | oilinki3 | pop? |
| 17:16.37 | outtolunc | matt_keys: you have to recompile zaptel with the new kernel (and make sure you kernel source/header match it).. |
| 17:16.40 | *** join/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-2221ab5d258f1ff3) |
| 17:16.40 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 17:17.05 | matt_keys | outtolunc : tried recompiling zap a min ago, everything looked fine |
| 17:17.11 | outtolunc | zaptel: version magic '2.6.23.15-80.fc7 SMP mod_unload 686 4KSTACKS ' should be '2.6.23.17-88.fc7 SMP mod_unload 686 4KSTACKS ' |
| 17:17.17 | fasttrack | see.. am not trying to be rude here. 1) am not hijacking any conv.. or flodding or even offtopic.. i'm still talking abt voip. 2) one of the most qualified ppl i expect are here |
| 17:17.33 | outtolunc | it is telling you right there, that teh zaptel magic version is not matching the loaded kernel |
| 17:17.58 | matt_keys | outtolunc : ok thanks, will recheck everything and recompile |
| 17:19.20 | IsUp | how can i list spans? |
| 17:19.33 | IsUp | i have Sangoma and Digium Wildcard TDM on the same machine |
| 17:19.38 | IsUp | which one loaded first? |
| 17:21.37 | SwK | anyone around hsv have a Gen 1 iphone they wanna part with |
| 17:23.40 | fasttrack | so most of you are of opinion that i cannot use 2 softphones on same machine. ?. K |
| 17:24.13 | kaldemar | it doesn't make sense. |
| 17:24.21 | fasttrack | sense? |
| 17:24.27 | fasttrack | for testing |
| 17:24.37 | TJNII | fasttrack: it doesn't work. |
| 17:24.43 | outtolunc | you could, if you had 2 (or more) audio devices, and have each softphone listing on diff ports |
| 17:25.13 | kaldemar | they would have to listen to different ports for signalling and be able to share the same audio hardware, but that's just plain useless if you ask me. |
| 17:25.16 | *** join/#asterisk r0land (n=roland@193.227.191.91) |
| 17:25.17 | r0land | hello all |
| 17:25.24 | r0land | any1 knows where can i get asterisk's book! |
| 17:25.31 | kaldemar | ~thebook |
| 17:25.31 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 17:25.38 | r0land | thank you |
| 17:26.10 | fasttrack | K. now i get the picture. its looking more logical. so i am wondering if ther is any hack for this? i guess not? |
| 17:26.52 | viraptor | fasttrack: get another box, or try calling asterisk's echo channel |
| 17:27.04 | TJNII | Yes! Let's hack for testing! That's a spectacular idea! Making the testing environment complicated and unstable is the best way to determine sutability of a device. |
| 17:27.07 | kaldemar | fasttrack: and anyway, this is not a general VoIP support channel. if you have a problem with asterisk, you're more than welcome to come here for help. but with that kind of issues, you're not likely to get anyone to hold your hand. |
| 17:27.27 | fasttrack | echo channel? |
| 17:27.37 | mort_gib | fasttrack: virtual machines, but still audio on the same hardware, you are better off with two boxes |
| 17:27.45 | fasttrack | kaldemar: i promise to leave in a min. |
| 17:28.00 | mort_gib | Ata's are not that expensive |
| 17:28.08 | mort_gib | Handy too |
| 17:28.18 | fasttrack | hmmm.. |
| 17:28.25 | fasttrack | so i just need an ata? |
| 17:28.33 | kaldemar | fasttrack: i'm not saying you should leave, just that you're wasting your time here with that issue. |
| 17:28.56 | carrar | IRC is great for wasting time |
| 17:29.05 | mort_gib | Analogue Telephone Adapter (I think) -Bottom line is it turns your average Analogue phone into a SIP device |
| 17:29.08 | fasttrack | kaldemar: no, its getting forward, many are helping. i am more clearer now than before. |
| 17:29.56 | fasttrack | kwel.. yeah yeah.. ata 180 something from cisco...etc |
| 17:30.17 | mort_gib | You don't need a Cisco one... |
| 17:30.57 | fasttrack | i need atleast h.323 and sip |
| 17:31.31 | mort_gib | for testing?? |
| 17:31.39 | fasttrack | coz i am studying for cvoice 6.0. but thats ok.. i'll manage. now i got the picture. thank you. |
| 17:31.47 | *** join/#asterisk angryuser (n=angryuse@95.250.146.195.dynamic.adsl.abo.nordnet.fr) |
| 17:31.49 | *** join/#asterisk km- (n=pgrace@2001:4830:1215:2:20c:29ff:fee9:9689) |
| 17:32.03 | fasttrack | i need it for gatekeeper and gateway studies. |
| 17:34.16 | fasttrack | thank you all for the time. i will come back with my approach after i succeed in setting up the lab. thank you. |
| 17:38.20 | *** join/#asterisk dioedu (n=dioedu@201.7.117.114) |
| 17:38.42 | matt_keys | outtolunc : it's not letting me install kernel-devel-i686.. it says it's already got a newer version installed. should I tell it to force install both kernel-i686 and kernel-devel-i686? |
| 17:40.19 | *** join/#asterisk tkbeat (n=tk@g228093183.adsl.alicedsl.de) |
| 17:41.10 | *** join/#asterisk hardwire (n=hardwire@rdbk-12639.mtaonline.net) |
| 17:43.27 | dioedu | is there any problem with zapata.conf in realtime static ? I can't configure more than one channel in my table... i try many configurations with cat_metric and var_metric and all failed... |
| 17:44.11 | dioedu | i have a TDM24XX card... |
| 17:44.35 | *** part/#asterisk fasttrack (n=joeljose@218.248.68.63) |
| 17:46.07 | dioedu | if i configure channel 2 with cat_metric 2 and channel 1 with cat_metric 1, asterisk just show me channel 2 configured... |
| 17:48.27 | dioedu | in debug file, asterisk write the correct select in database... found all lines configured, but it doesn't work's... |
| 17:57.04 | *** part/#asterisk matt_keys (i=smatt@usaregs.com) |
| 17:57.35 | adr3nalin3 | Hey guys if I recompile asterisk will it leave the users vm boxes intact? I am going to back up /etc/asterisk first. Anything else I should worry about? |
| 17:58.12 | *** join/#asterisk sjb_gt (n=sachajbe@71-12-78-112.dhcp.spbg.sc.charter.com) |
| 17:58.56 | IsUp | after |
| 17:59.01 | IsUp | rmmod wctdm24xxp |
| 17:59.03 | IsUp | i getting |
| 17:59.16 | IsUp | [Aug 28 20:43:25] WARNING[16662]: chan_zap.c:903 zt_open: Unable to specify channel 24: Device or resource busy |
| 17:59.16 | IsUp | [Aug 28 20:43:25] ERROR[16662]: chan_zap.c:7160 mkintf: Unable to open channel 24: Device or resource busy |
| 17:59.24 | IsUp | [Aug 28 20:40:44] WARNING[15988]: app_dial.c:1106 dial_exec_full: Unable to create channel of type 'ZAP' (cause 66 - Channel not implemented) |
| 17:59.40 | dioedu | how is zaptel.conf ? |
| 17:59.43 | dioedu | IsUp ? |
| 18:00.32 | dioedu | these errors in rmmod ? |
| 18:00.52 | IsUp | no |
| 18:00.55 | IsUp | when i start asterisk |
| 18:01.16 | *** part/#asterisk sjb_gt (n=sachajbe@71-12-78-112.dhcp.spbg.sc.charter.com) |
| 18:01.17 | dioedu | ah ok |
| 18:02.43 | dioedu | this card works before this error ? or never works ? |
| 18:02.44 | IsUp | help |
| 18:02.45 | IsUp | urgent |
| 18:03.27 | dioedu | i had a problem with one channel in my card and i needed to disable this channel in zaptel.conf and zapata.conf... |
| 18:03.57 | dioedu | but pastebin your zaptel.conf |
| 18:04.25 | kaldemar | IsUp: do you expect zaptel to work after you've removed the module? |
| 18:04.25 | oilinki3 | I start to be happy with asterisk |
| 18:04.56 | DarylVOIP | IsNotUp - why did you remove the module? |
| 18:05.29 | DarylVOIP | did you add it back before restarting? |
| 18:05.57 | IsUp | no |
| 18:05.59 | IsUp | how can i add |
| 18:06.02 | IsUp | i tried modprobe |
| 18:06.04 | IsUp | what is else? |
| 18:06.10 | DarylVOIP | try insmod |
| 18:06.45 | seanbright | question... i ran 'shutdown -h now' and now my machine isn't responding |
| 18:06.48 | seanbright | help |
| 18:06.51 | seanbright | urgent |
| 18:06.51 | dioedu | ztcfg |
| 18:06.53 | IsUp | insmod wctdm24xxp? |
| 18:06.54 | DarylVOIP | But why did you remove it to begin with? |
| 18:06.57 | DarylVOIP | lol |
| 18:07.01 | DarylVOIP | yes |
| 18:07.02 | IsUp | wrong |
| 18:07.03 | IsUp | but |
| 18:07.06 | IsUp | insmod: can't read 'wctdm24xxp': No such file or directory |
| 18:07.12 | IsUp | where should i execute it? |
| 18:07.17 | DarylVOIP | you have to give it a full path to the module |
| 18:07.24 | dioedu | lsmod show you zaptel and wctdm24xxp ? |
| 18:07.24 | IsUp | base.c GpakApi.c GpakApi.o GpakCust.h gpakenum.h GpakHpi.h Makefile.kernel26 wctdm24xxp.ko wctdm24xxp.mod.o |
| 18:07.24 | IsUp | base.o GpakApi.h GpakCust.c GpakCust.o gpakErrs.h Makefile wctdm24xxp.h wctdm24xxp.mod.c wctdm24xxp.o |
| 18:07.26 | IsUp | which one? |
| 18:07.33 | IsUp | no zaptel |
| 18:07.37 | IsUp | no wctdm24xxp |
| 18:07.44 | DarylVOIP | insmod /blah/vlah/akjsd/wctdm24ccp.o |
| 18:07.49 | dioedu | modprobe zaptel |
| 18:07.51 | Qwell | what? no |
| 18:07.57 | dioedu | and modprobe wctdm24xxp |
| 18:07.58 | *** join/#asterisk lesouvage (n=lesouvag@62.140.137.125) |
| 18:07.58 | outtolunc | haha |
| 18:08.00 | DarylVOIP | if he removed one and can't modprobe..... |
| 18:08.13 | IsUp | insmod: error inserting 'wctdm24xxp.o': -1 Invalid module format |
| 18:08.22 | DarylVOIP | maybe I'm just misunderstanding here. Since I don't understand why he unloaded it to begin with. |
| 18:08.23 | dioedu | IsUp, try modprobe zaptel first |
| 18:08.39 | dioedu | don't try to understand... |
| 18:08.39 | dioedu | :p |
| 18:08.52 | DarylVOIP | lol. OK. I'll just watch the show. |
| 18:08.52 | IsUp | not working |
| 18:08.58 | IsUp | root@lamassu:/usr/src/zaptel/wctdm24xxp# insmod wctdm24xxp.mod.o |
| 18:08.58 | IsUp | insmod: error inserting 'wctdm24xxp.mod.o': -1 File exists |
| 18:09.11 | dioedu | IsUp, and modprobe zaptel ? |
| 18:09.15 | IsUp | i tried |
| 18:09.20 | dioedu | and ? |
| 18:09.28 | IsUp | not in lsmod |
| 18:09.37 | dioedu | :p |
| 18:09.39 | IsUp | how can i setup it quickly? |
| 18:09.50 | dioedu | do you compile zaptel ? |
| 18:09.56 | IsUp | configure, make, make install |
| 18:10.02 | IsUp | thats enough? |
| 18:10.06 | dioedu | yes... |
| 18:10.06 | dioedu | :p |
| 18:10.16 | IsUp | i have wanpipe |
| 18:10.18 | IsUp | should i stop it? |
| 18:10.44 | dioedu | which version of zaptel and asterisk ? |
| 18:11.52 | seanbright | this is like watching two quadrapoligics trying to jerk each other off |
| 18:12.11 | IsUp | still not working |
| 18:12.15 | seanbright | which i am known to do |
| 18:12.18 | IsUp | [Aug 28 21:12:04] WARNING[9483]: channel.c:3232 ast_request: No channel type registered for 'ZAP' |
| 18:12.18 | IsUp | [Aug 28 21:12:04] WARNING[9483]: app_dial.c:1106 dial_exec_full: Unable to create channel of type 'ZAP' (cause 66 - Channel not implemented) |
| 18:12.23 | IsUp | i configured make make install zaptel |
| 18:12.34 | outtolunc | really did not need that image in my head |
| 18:12.36 | seanbright | did you restart asterisk afterwards? |
| 18:13.10 | *** join/#asterisk vascodx (n=augusto@190.189.44.112) |
| 18:13.15 | dioedu | did you configure zaptel and zapata ? |
| 18:13.19 | dioedu | i think not |
| 18:13.39 | km- | seanbright: you jerk off quadripalegics? |
| 18:13.45 | IsUp | i resterted it |
| 18:13.59 | km- | I tells ya, Se Anbright is one upstanding guy ;) |
| 18:14.02 | IsUp | yes i configured |
| 18:14.05 | IsUp | i have conf backups |
| 18:14.12 | IsUp | ive uploaded it |
| 18:14.24 | IsUp | byt |
| 18:14.25 | IsUp | but |
| 18:14.28 | IsUp | zaptel is not in lsmod |
| 18:14.33 | seanbright | km-: excellent contibution. |
| 18:14.42 | seanbright | s/conti/contri/ |
| 18:14.51 | eric_hill | Does anyone know a good way to limit the number of calls to a phone in a queue to 1 call at a time, but still allow the queue member to place second calls with the first on hold? |
| 18:14.55 | eric_hill | call-limit seems to be an all-or-nothing setting... |
| 18:14.57 | IsUp | any ideas? |
| 18:14.58 | IsUp | urgent |
| 18:14.59 | km- | seanbright: hey, even the quads need some love. |
| 18:15.40 | *** part/#asterisk vascodx (n=augusto@190.189.44.112) |
| 18:15.53 | IsUp | Dary or seanbright |
| 18:15.55 | seanbright | eric_hill: ringinuse in queues.conf |
| 18:15.56 | IsUp | any ideas? |
| 18:15.56 | dioedu | if you compile zaptel and can't modprobe it, is really weird... |
| 18:15.58 | km- | s/(\w{1}).*?/hehe $1/ |
| 18:16.11 | km- | hmm |
| 18:16.11 | Qwell | eric_hill: like, it either limits or it doesn't? |
| 18:16.15 | brodiem | eric_hill: call-limit isn't all or nothing, you can tell it how many calls are allowed |
| 18:16.15 | km- | jbot told me to go to hell |
| 18:16.23 | seanbright | IsUp: i would help, but your english sucks and you keep saying "urgent" which makes me want to stab you |
| 18:16.25 | km- | he means it works both directions |
| 18:16.34 | km- | he wants call-limit to be limit incoming queue calls to 1 |
| 18:16.42 | km- | but still allow the queue member to make outbound calls |
| 18:16.45 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 18:16.47 | seanbright | ringinuse |
| 18:16.53 | seanbright | ringinuse |
| 18:17.10 | [TK]D-Fender | echo-cancels seanbright |
| 18:17.11 | km- | I'm not that up on queues, was never something I worked with |
| 18:17.21 | seanbright | oh i have an idea! |
| 18:17.27 | seanbright | ringinuse in queus.conf |
| 18:17.35 | seanbright | queues* |
| 18:18.00 | eric_hill | With a call-limit=1, only one call is attempted to the phone at a time. With call-limit=10, I get RINGNOANSWER to the phone because the agent is on the phone with another caller. |
| 18:18.06 | IsUp | seanbright, lines are in use |
| 18:18.09 | IsUp | i should fix it |
| 18:18.18 | seanbright | IsUp: i wasn't talking to you |
| 18:18.24 | IsUp | i reconfigure make make install zaptel |
| 18:18.27 | IsUp | and then modprobe zaptel |
| 18:18.30 | eric_hill | See here: http://pastebin.com/m5f9f2a12 |
| 18:18.30 | IsUp | but its not in lsmod |
| 18:18.45 | IsUp | i have sangoma a104 and digium wildcard |
| 18:18.58 | *** join/#asterisk vascodx (n=augusto@190.189.44.112) |
| 18:18.58 | seanbright | eric_hill: have you considered ringinuse in queues.conf? |
| 18:19.00 | km- | eric_hill: are you using ringinuse=no in queues.conf like seanbright said? |
| 18:19.12 | seanbright | i'm kinda getting tired of repeating myself |
| 18:19.13 | eric_hill | One sec... |
| 18:19.23 | IsUp | ah |
| 18:19.26 | IsUp | i got zaptel on lsmod |
| 18:19.28 | km- | seanbright: you could have quit long ago, ya know. |
| 18:19.40 | dioedu | :p |
| 18:19.48 | IsUp | ah no |
| 18:19.51 | eric_hill | seanbright! You're a genius! It was commented out. |
| 18:19.52 | IsUp | its not |
| 18:19.53 | km- | seanbright: you can lead a horse to water, but you just cant shove the hose down his throat sometimes. |
| 18:20.01 | seanbright | km-: why are you speaking to me in a familiar way? |
| 18:20.27 | km- | seanbright: because I am familiar with everyone. would you rather me address you in some other tone? |
| 18:20.31 | seanbright | eric_hill: yes i know. good luck and god speed. |
| 18:20.37 | seanbright | km-: yes, how about "not at all" |
| 18:20.58 | km- | seanbright: you wound me. |
| 18:22.47 | eric_hill | hands seanbright a beer. |
| 18:23.17 | seanbright | eric_hill: that will only work if you have a call-limit set |
| 18:23.31 | seanbright | eric_hill: even if it is arbitrarily high (like 99) |
| 18:23.31 | IsUp | root@lamassu:/usr/src/zaptel# lsmod | grep zaptel |
| 18:23.31 | IsUp | zaptel 189988 52 wanpipe,ztdummy,zttranscode,wctdm24xxp |
| 18:23.31 | IsUp | crc_ccitt 3072 1 zaptel |
| 18:23.33 | *** join/#asterisk nighty^ (n=nighty@host217-37-109-17.in-addr.btopenworld.com) |
| 18:23.40 | IsUp | still not loading chan_zap.so |
| 18:23.41 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 18:23.41 | IsUp | please someone help |
| 18:23.46 | eric_hill | seanbright: I have it set to 5 right now, and so far, so good. Gracias. |
| 18:23.52 | seanbright | yup |
| 18:24.09 | *** join/#asterisk Levonk (n=lk@76.237.15.7) |
| 18:24.36 | km- | isup: what is the error code saying? |
| 18:24.56 | IsUp | [Aug 28 21:24:42] WARNING[12342]: chan_zap.c:903 zt_open: Unable to specify channel 24: Device or resource busy |
| 18:24.56 | IsUp | [Aug 28 21:24:42] ERROR[12342]: chan_zap.c:7160 mkintf: Unable to open channel 24: Device or resource busy |
| 18:25.10 | IsUp | [Aug 28 20:39:15] WARNING[15882]: app_dial.c:1106 dial_exec_full: Unable to create channel of type 'ZAP' (cause 66 - Channel not implemented) |
| 18:25.17 | seanbright | stop pasting in the channel |
| 18:25.18 | IsUp | [Aug 28 20:39:57] WARNING[15861]: pbx.c:2948 ast_register_application: Already have an application 'ZapSendKeypadFacility' |
| 18:25.21 | IsUp | okay |
| 18:25.25 | seanbright | use a pastebin |
| 18:26.54 | [TK]D-Fender | ~pb |
| 18:26.54 | jbot | [~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 |
| 18:27.03 | seanbright | yes... that ^^^ |
| 18:27.06 | dioedu | IsUp, pastebin return of ztcfg -vv |
| 18:27.43 | M1s3ry | if chan_zap.so isn't loaded... have you tried module load chan_zap.so? |
| 18:28.19 | IsUp | yes i tried |
| 18:28.32 | dioedu | M1s3ry, i think will show the same error... |
| 18:29.22 | IsUp | i am configuring zaptel-1.4.11 now |
| 18:29.29 | IsUp | i hope it works |
| 18:29.52 | M1s3ry | well IsUp for the wildcard (as long as it's in warranty) you can call digium support (256-428-6000), however for the sangoma card, dunno... |
| 18:30.18 | IsUp | i think digium cannot support to idiots like me :> |
| 18:30.21 | dioedu | why don't you pastebin what i said ? |
| 18:30.24 | IsUp | i just removed mod |
| 18:30.25 | IsUp | pff |
| 18:30.28 | IsUp | wait dio i'll |
| 18:30.31 | IsUp | when finish zap |
| 18:30.56 | M1s3ry | IsUp, trust me, they can help. |
| 18:31.09 | M1s3ry | s/they/we |
| 18:31.12 | M1s3ry | whatev |
| 18:32.05 | *** join/#asterisk EI5GTB (n=EI5GTB@193.120.192.61) |
| 18:36.58 | *** join/#asterisk vascodx (n=augusto@190.189.44.112) |
| 18:37.16 | *** join/#asterisk franciscoer (n=francisc@190.189.44.112) |
| 18:41.13 | IsUp | anyone going to help? |
| 18:41.54 | IsUp | http://pastebin.ca/1187711 |
| 18:42.16 | DarylVOIP | No one can. There's no enough duct tape in the world to kepp our heads from exploding. |
| 18:42.21 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:43.01 | *** join/#asterisk lesouvage (n=lesouvag@62.140.137.125) |
| 18:44.07 | *** join/#asterisk franciscoer (n=francisc@190.189.44.112) |
| 18:44.10 | IsUp | it's a production IVR |
| 18:44.13 | IsUp | i should really fix it |
| 18:44.47 | DarylVOIP | You should really have not broken it to begin with. What were you doing and why? I missed the beginning of this one. |
| 18:45.31 | IsUp | i was unable to use some channels on dial/zap |
| 18:45.33 | IsUp | i tried |
| 18:45.37 | IsUp | unload digium |
| 18:45.41 | IsUp | coz i have sangoma and wildcard |
| 18:45.44 | IsUp | i typed |
| 18:45.51 | IsUp | rmmod wctdm24xxp |
| 18:45.54 | IsUp | and story begins |
| 18:46.05 | DarylVOIP | OK...so hold on...you couldn't use some channels. Were you able to use them before? |
| 18:46.32 | IsUp | yes |
| 18:46.48 | IsUp | i was able to use. and i have configuration files backup |
| 18:46.52 | DarylVOIP | And what has changed since then? OS or kernel updates? |
| 18:47.01 | IsUp | nothing |
| 18:47.03 | IsUp | i just |
| 18:47.06 | IsUp | rmmod wctdm24xxp and |
| 18:47.08 | IsUp | shutdown -r now |
| 18:47.14 | DarylVOIP | Before you started typing. |
| 18:47.14 | IsUp | restarted my server |
| 18:47.18 | IsUp | nothing |
| 18:48.06 | IsUp | on cat /proc/zaptel/1 it says: IRQ misses: 147 |
| 18:48.10 | IsUp | for Digium |
| 18:48.10 | DarylVOIP | and touching things.....that's why I'm trying to figure out. You just lost the use of some channels without anything at all changing before you started removing, rebooting, etc? |
| 18:48.38 | IsUp | i didnt know rmmod deletes permanetly |
| 18:48.48 | DarylVOIP | It doesn't. |
| 18:49.11 | IsUp | what should i do? |
| 18:49.23 | IsUp | its production and its really urgent |
| 18:49.27 | DarylVOIP | So, here's the thing: did you server used to come up from a reboot and work properly? |
| 18:49.27 | IsUp | i should fix it in some way.. |
| 18:49.35 | DarylVOIP | As in, all the modules loaded and daemons started? |
| 18:49.51 | IsUp | yes. it's up after reboot |
| 18:49.59 | IsUp | yes all daemons started |
| 18:50.30 | DarylVOIP | So you should put everything back to the way it was, figure out if it's your circuit or the card, and start over. If you've changed nothing else, it's one of the two things. Does that no make sense? |
| 18:51.05 | IsUp | i've tried reconfigure zaptel (same version) |
| 18:51.17 | IsUp | and |
| 18:51.19 | *** join/#asterisk unenough (n=unenough@CBL217-132-93-165.bb.netvision.net.il) |
| 18:51.24 | IsUp | should i start zaptel wanrouter ztcfg an order? |
| 18:51.32 | *** join/#asterisk write_erase (n=Olivier@royale.aixmarseille.com) |
| 18:51.40 | IsUp | i am just typing wanrouter start after 10 sec ztcfg -vvv and then asterisk -vvvgc |
| 18:51.43 | unenough | hi, I'm trying to connect my asterisk to callcentric |
| 18:51.46 | DarylVOIP | You're one of those people who just starts changing things at random when you have a problem I see. |
| 18:51.51 | mfdavid | I can answer the phone and play a sond. Half the job is done. But I cannot make a call yet :-( if I try something like console dial 11878xxxxx it says No such extension '1187867587' in context 'default'. Can someone point me to a place that shows me how t |
| 18:52.00 | IsUp | no Dary |
| 18:52.01 | IsUp | i am not |
| 18:52.11 | unenough | i did what callcentric say you should do, and set up an extension "123" with context=to-callcentric |
| 18:52.14 | IsUp | i thought two cards can be problematic |
| 18:52.22 | IsUp | i just want to unload digium |
| 18:52.58 | DarylVOIP | It can be, but if you had it working, changed nothing, and just lost some channels, reconfiguring is NOT the way to go. There are other things to tak into account first. Like ig a channel is hung, if you have a provider issue, or if you have a hardware issue. |
| 18:53.02 | unenough | and in extensions.conf i have: exten => _XX,1,Dial(SIP/${EXTEN}@callcentric) (under [to-callcentric]), but when I dial from 123 to a callcentric number I get: 404 not found |
| 18:53.16 | unenough | how can i figure this out? |
| 18:53.19 | DarylVOIP | Now it sounds like you're so far from your initial configuration that no one can really tell you where to go back to, because no one knows your config. |
| 18:53.47 | DarylVOIP | Except, hopefully, you. |
| 18:54.13 | *** join/#asterisk lowtek (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 18:54.13 | DarylVOIP | If this is that serious of an issue, you need to get off of IRC, take out your credit card, and call Digium. They will log in and fix it. |
| 18:54.21 | kaldemar | unenough: what number did you dial? the Dial line only matches to numbers with two digits. |
| 18:54.30 | unenough | oh, how do i fix that? |
| 18:54.34 | unenough | the _XX? |
| 18:54.38 | [TK]D-Fender | unenough: CHANGE IT |
| 18:54.54 | IsUp | genzaptelconf: CAS signalling on span 3 conflicts with HDLC with FCS check on channel 71. |
| 18:55.00 | kaldemar | unenough: dialplan patterns is the keyword. read up on it. |
| 18:55.10 | unenough | k thanks. |
| 18:55.29 | unenough | i figured it was something in that direction :) just another newbie here |
| 18:56.13 | *** join/#asterisk DarylVOIP (n=daryl@75.147.121.177) |
| 18:59.16 | DarylVOIP | So what does ztcfg –vvvvv tell you? |
| 19:03.27 | unenough | kaii, [TK]D-Fender is it a good idea to put _. instead of that _XX ? |
| 19:03.34 | *** join/#asterisk Levonk (n=lk@adsl-76-238-251-184.dsl.lsan03.sbcglobal.net) |
| 19:03.44 | unenough | or will that be problematic... |
| 19:03.46 | *** join/#asterisk r0land (n=roland@193.227.191.91) |
| 19:03.48 | r0land | hello all |
| 19:03.55 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:04.19 | r0land | im tryig to fix my dial plan so if i call a line and its not available it tries another.. |
| 19:04.30 | r0land | ive added the following : exten =>_02,1,Dial(SIP/304) exten =>_02,2,Dial(SIP/300) |
| 19:04.38 | r0land | first time i call "02" i take a line.. |
| 19:04.58 | r0land | thoug if i call 02 again , i dont hear anything nothing happens..! |
| 19:05.00 | r0land | any idea why ? |
| 19:05.14 | unenough | hmm I guess using _177. will be better than just _., ok thanks anyways! |
| 19:05.50 | seanbright | whenever i feel like i might not be an adequate PBX admin, guys like IsUp come along and put it all in perspective |
| 19:05.58 | seanbright | it's like watching COPS |
| 19:06.34 | DarylVOIP | Good point. |
| 19:06.39 | outtolunc | its almost friday <G> |
| 19:06.58 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 19:07.04 | seanbright | thank jeebus |
| 19:07.22 | *** join/#asterisk korihor (n=korihor@200.7.99.2) |
| 19:07.31 | [TK]D-Fender | unenough: no, "_." is a shoot-on-sight offense |
| 19:07.47 | unenough | [TK]D-Fender, i was guessing it's bad, but why exactly? |
| 19:07.58 | seanbright | because it matches *everything* |
| 19:08.02 | [TK]D-Fender | unenough: because it will match EVERYTHING |
| 19:08.04 | Qwell | not everything |
| 19:08.12 | [TK]D-Fender | all the BAD stuff anyways |
| 19:08.13 | Qwell | just anything with 1 or more digits |
| 19:08.29 | seanbright | you mean |
| 19:08.34 | seanbright | EVERYTHING with 1 or more digits |
| 19:08.36 | unenough | [TK]D-Fender, i understand, but what bad stuff for example |
| 19:08.38 | [TK]D-Fender | Qwell: Actually it DOES catch "everything". What it DOESN'T catch is "nothing" :) |
| 19:08.39 | Qwell | _! = everything |
| 19:08.40 | seanbright | see? EVERYTHING still applies. |
| 19:08.45 | unenough | i'm not so aware of the security implications here |
| 19:08.55 | *** join/#asterisk brown10101 (n=celso@cpc1-walt6-0-0-cust156.popl.cable.ntl.com) |
| 19:09.28 | outtolunc | is transfered to exten '' at nowhere |
| 19:11.44 | r0land | any1 has an advice about how to fix my dial plan to b able to dial multiple lines till i get one thts not busy! |
| 19:12.02 | r0land | this is what i added thinking tht it might do the trick.. : exten =>_02,1,Dial(SIP/304) exten =>_02,2,Dial(SIP/300) |
| 19:12.21 | *** join/#asterisk gr0mit (n=tim@81.187.32.146) |
| 19:13.35 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 19:15.16 | *** join/#asterisk bijit (n=benji@190.241.15.48) |
| 19:18.39 | r0land | can i use gotoif instead ?! |
| 19:18.45 | r0land | instead of using priorities.. |
| 19:18.51 | lowtek | tries rm -fR / just for fun ... |
| 19:19.22 | Nugget | lowtek: http://macnugget.org/stuff/unix-horror-story.txt |
| 19:19.32 | [TK]D-Fender | r0land: .... what are you going to "goto"? ANOTHER PRIORITY. FFS jsut dial them back to back. There is no better way |
| 19:20.02 | r0land | i tried with the priority issue! |
| 19:20.10 | jblack | wonders what's up |
| 19:20.12 | [TK]D-Fender | r0land: what "issue"? |
| 19:20.22 | r0land | as shown above.. it didnt work! when the first line is used, and i call its extension i dont hear anything.. no line but no congestion either |
| 19:20.29 | r0land | exten =>_02,1,Dial(SIP/304) exten =>_02,2,Dial(SIP/300) <<-- |
| 19:20.31 | [TK]D-Fender | r0land: You'd better learn to PASTEBIN the backup for your problems.... |
| 19:20.46 | [TK]D-Fender | r0land: pastebin your dialplan and the failed CLI output |
| 19:20.51 | r0land | k |
| 19:21.19 | jblack | [TK]D-Fender: You're doing to die of a heart attack by 40 with the way you take things. All this stress isn't good for your ticker. ;) |
| 19:21.42 | jblack | imagines [TK]D-Fender with kids. |
| 19:21.56 | jblack | 'Look, you moron! you lift the seat BEFORE you pee!' |
| 19:21.58 | [TK]D-Fender | jblack: Kids learn... its too late for this guy... |
| 19:22.22 | r0land | [TK]D-Fender http://pastebin.com/df438bb3 |
| 19:22.33 | lowtek | Nugget: lol, great story, I would be crying ... I've done it before in a development VM but never in production (knocking feverishly on wood as I type) |
| 19:22.46 | [TK]D-Fender | r0land: And the rest? |
| 19:22.49 | Nugget | yeah, it's a great story. I can't say I'd do that well in the situation. |
| 19:23.49 | r0land | cli : http://pastebin.com/d794af5b3 |
| 19:24.50 | [TK]D-Fender | r0land: And whats the problem? |
| 19:24.52 | r0land | as i see it.. its stil trying to call the first one |
| 19:25.05 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 19:25.09 | [TK]D-Fender | r0land: which it should |
| 19:25.34 | r0land | when i call 02.. i get a line from 304... but when i call 02 again.. it still takes the first 02 priority which is 304 instead of jumping to next priority and taking 300 |
| 19:27.01 | [TK]D-Fender | r0land: Is it BAD that it tries 304 first? You stopped the pastebin without showing what HAPPENED after the 2nd call to it |
| 19:27.12 | jblack | r0land: You can pastebin your dialplan |
| 19:27.12 | unenough | ok, i managed to get a call out (asterisk -> callcentric) but not IN, my [from-callcentric] context says: exten => s,1,Dial(SIP/123) |
| 19:27.32 | r0land | jblack i did! |
| 19:27.32 | unenough | but when I call, I get: Call from '' to extension '1777...' rejected because extension not found. |
| 19:27.37 | jblack | Generally, * uses the most restricted pattern in the event that multiple patterns match. |
| 19:27.43 | jblack | r0land: Cool! Where is it? |
| 19:27.52 | r0land | jblack http://pastebin.com/df438bb3 cli: http://pastebin.com/d794af5b3 |
| 19:28.10 | [TK]D-Fender | r0land: AND? |
| 19:28.16 | unenough | so what's the problem now |
| 19:28.16 | lowtek | Do voicemail greetings have to be executable (+x) for asterisk to modify them for some odd reason? |
| 19:28.33 | jblack | r0land: And what's the problem again? |
| 19:28.35 | jblack | I just woke up |
| 19:28.48 | [TK]D-Fender | unenough: You don't have an extension in the context it's looking inside of to match THAT number. |
| 19:28.57 | r0land | jblack if line 304 is busy, i want aterisk to 300 |
| 19:29.03 | jblack | Ok. lets start off on something easy. Look at line 61. See that _02 ? It looks like a pattern, but it's not. |
| 19:29.08 | r0land | jblack if line 304 is busy, i want aterisk to jump to 300 |
| 19:29.40 | unenough | [TK]D-Fender, how can I see which context it's looking in? |
| 19:29.46 | jblack | you may as well call _02 02. |
| 19:29.51 | [TK]D-Fender | r0land: You didn't show us WHAT HAPPENS when you call 304 and it's "busy". |
| 19:30.10 | r0land | [TK]D-Fender <r0land> cli : http://pastebin.com/d794af5b3 |
| 19:30.12 | jblack | Ok, so, you want someone to try 304, and if there's no answer, dial 300. |
| 19:30.13 | r0land | i did :) |
| 19:30.15 | jblack | Ok. So, do this. |
| 19:30.18 | r0land | jblack yes exactly |
| 19:30.34 | jblack | exten => 300,1,Dial(SIP/person) |
| 19:30.42 | jblack | exten => 304,1,Dial(SIP/anotherperson) |
| 19:30.52 | *** join/#asterisk Assid (n=assid@unaffiliated/assid) |
| 19:30.56 | jblack | exten => 304,2,Goto(thecontext,300,1) |
| 19:31.02 | [TK]D-Fender | r0land : http://pastebin.com/m352096ec |
| 19:31.24 | jblack | I should have written that on pastebin instead of here. sorry guys |
| 19:31.31 | Assid | hi.. i have monitor() with m option to mix the final in/out files. however, that doesnt seem to be happening. i still have 2 seperate files |
| 19:31.33 | r0land | [TK]D-Fender cos nothing showed after.. it simply stopped till i closed the line |
| 19:31.47 | Assid | any idea how to get that up? |
| 19:32.04 | [TK]D-Fender | r0land: NOTHING? No "ringing? NOTHING? You telling me it did not give ANY indication of progress? |
| 19:32.13 | r0land | [TK]D-Fender not at all! nothing |
| 19:32.20 | [TK]D-Fender | Assid: verify you have SOX installed |
| 19:32.28 | Assid | [TK]D-Fender: yes sox is installed |
| 19:32.33 | [TK]D-Fender | r0land: I want to see the call right through then end |
| 19:32.40 | Assid | [TK]D-Fender: located in /usr/bin/sox |
| 19:32.41 | r0land | [TK]D-Fender k il pastebin again |
| 19:33.27 | *** join/#asterisk Levonk (n=lk@adsl-76-230-109-27.dsl.lsan03.sbcglobal.net) |
| 19:33.33 | Assid | http://assid.pastebin.com/d16bad1ac |
| 19:33.42 | jblack | r0land. Let me make you a simpler dialplan |
| 19:33.45 | Assid | i removed the called number |
| 19:35.21 | r0land | [TK]D-Fender http://pastebin.com/d6849fe62 |
| 19:35.39 | r0land | jblack i prefer learning on my own if u dont mind! |
| 19:35.49 | r0land | i know this might end up in causing [TK]D-Fender a heart attack.. :P |
| 19:35.56 | Assid | native bridging.. nice |
| 19:36.00 | unenough | How can i find out which context was being used for the incoming call? |
| 19:36.18 | jblack | Ok. THen you need to read the book more. |
| 19:36.56 | outtolunc | i don't see why you just don't limit your sip devices to 1 occurance |
| 19:36.58 | r0land | jblack i am..! and tried the priority issue to get asterisk to jump to another line if the first is busy! but its not actually working as expected |
| 19:37.26 | jblack | No. I don't accept that. |
| 19:37.41 | jblack | You learn how to do something by reading and following good examples. |
| 19:37.46 | jblack | Hold please. |
| 19:38.09 | r0land | jblack tried exten =>_01,1,Dial(SIP/304/300) as well as advised in the book! didnt work either |
| 19:38.34 | outtolunc | that would be Dial(SIP/304&SIP/300) |
| 19:38.46 | jblack | You've got something like 3 extensions for each person. |
| 19:38.50 | jblack | That's bad mojo |
| 19:38.57 | jblack | what is this spa conext? |
| 19:39.12 | *** join/#asterisk curtn (n=curtis@ip-6.net-80-236-71.issy.rev.numericable.fr) |
| 19:39.20 | Assid | err quick question.. if i reinvite and end with native bridging.. the asterisk system will still log the call and its duration correct? |
| 19:39.38 | [TK]D-Fender | r0land: If your device does nothing when receiving a call when busy, then it has issues. To get around this " core show application chanisavail" <--- |
| 19:40.00 | jblack | qBleh. I can't figure out what you're doing in that dialplan |
| 19:40.01 | [TK]D-Fender | jblack: [spa] is a psycho mess of stuff that should all be in the same context. |
| 19:40.26 | jblack | I agree on the mess part. |
| 19:43.58 | Assid | yteah its being recorded |
| 19:44.05 | Assid | so rather have canreinvite for this |
| 19:44.38 | unenough | Ok, i've pastebined my dialplan and sip conf. Can somebody help me figure why incoming calls from callcentric don't work? http://www.pastebin.ca/1187783 |
| 19:44.55 | Assid | [TK]D-Fender: whenever your free.. would appreciate some light why this doesnt want to give me a single file |
| 19:45.01 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 19:45.06 | r0land | jblack have another quest while putting tht on hold..! |
| 19:45.32 | unenough | the error i get in asterisk's console is: Call from '' to extension '1777....' rejected because extension not found. |
| 19:45.53 | r0land | sipphone -- asterisk -- sipura... just now my call over lapped with another call! i heard the attendant and then heard some1 answerinfg the phone.. when i asked them to hang up.. they did.. but my line broke as well |
| 19:46.10 | r0land | jblack any idea? |
| 19:46.55 | [TK]D-Fender | unenough: You don't have an extension in the context it's looking inside of to match THAT number. <--- I've already answered you. |
| 19:47.20 | *** part/#asterisk dioedu (n=dioedu@201.7.117.114) |
| 19:47.36 | unenough | [TK]D-Fender, i didn't understand what that means exactly |
| 19:47.55 | [TK]D-Fender | unenough: it means you don't have an extension there that will MATCH that number |
| 19:47.56 | unenough | i have the number with context from-callcentric |
| 19:48.07 | outtolunc | r0land: it would be nice if you would fix your dialplan as it must be throwing errors/warnings |
| 19:48.45 | r0land | outtolunc u think it might b caused to the famous "mess" in my dialplan ? |
| 19:48.49 | [TK]D-Fender | outtolunc: He never fixes anything. He jsut runs around like a headless chicken jumping from problem to problem solving nothing. |
| 19:48.53 | outtolunc | each one of your incoming-.... has 2 exten => 4 's |
| 19:49.13 | outtolunc | yeah, i should have just left my trap shut <G> |
| 19:49.22 | [TK]D-Fender | outtolunc: Ditto :| |
| 19:49.25 | unenough | [TK]D-Fender, WHERE do i need to MATCH the number, in the line exten => ? in the context from-callcentric? |
| 19:49.40 | unenough | I really don't get what you're saying... |
| 19:49.48 | [TK]D-Fender | unenough: enable SIP debug and see what context it's looking in. |
| 19:50.11 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 19:50.39 | Assid | okay got my issue |
| 19:52.10 | jblack | r0land: http://pastebin.com/m8e9148e |
| 19:52.54 | jblack | r0land: That paste I gave you is much closer to what most people use. |
| 19:53.07 | unenough | [TK]D-Fender, it says in "from-callcentric" |
| 19:53.37 | [TK]D-Fender | unenough: Then thats the context you should be making a proper matching exten in. |
| 19:53.50 | unenough | [TK]D-Fender, ok got it, working. thanks. |
| 19:54.12 | [TK]D-Fender | jblack: I actually wasted my time and gave him a COMPLETE rewrite and fix only to have it discarded |
| 19:54.38 | unenough | now the problem is that when I hang up on asterisk's extension's side, the call stays connected on the callcentric network |
| 19:55.23 | mfdavid | is there an easy way to use asterisk to make calls from a cell phone? |
| 19:55.33 | jblack | [TK]D-Fender: Oh, he's one of those types? |
| 19:55.42 | [TK]D-Fender | jblack: Yup |
| 19:56.00 | *** join/#asterisk amessina (n=amessina@2001:470:1f11:68:20e:cff:fe01:d5ec) |
| 19:56.08 | Assid | for strftime can i set 'America/New_York' in the 2nd option? |
| 19:56.11 | jblack | One of those "I'd rather blunder it through the right way and inconvienance everyone around me for weeks, than spend 5 minutes learning from a good example" types ? |
| 19:56.16 | jblack | bleh. |
| 19:56.19 | *** join/#asterisk mirrorcolor (n=mirrorco@unaffiliated/mirrorcolor) |
| 19:56.25 | jblack | One of those "I'd rather blunder it through the wrong way and inconvienance everyone around me for weeks, than spend 5 minutes learning from a good example" types ? |
| 19:57.13 | jblack | Those are the worst types. They're black holes. |
| 19:57.14 | unenough | i've added another exten => step with Hangup after the Dial(SIP/123), but the call still stays connected after the asterisk-side extension hangs up |
| 19:57.45 | jblack | unenough: Are you using an ATA that's hooked up to an analog phone? |
| 19:58.14 | unenough | no, it's all SIP |
| 19:58.22 | unenough | and softphones |
| 19:58.30 | [TK]D-Fender | jblack: Good example? No I gave a DROP IN replacement |
| 19:58.43 | jblack | [TK]D-Fender: That's what I just did with that pastebin. |
| 19:59.16 | jblack | unenough: I'd do a high level verbose, and make sure you're going through the dialplan the way you intend. |
| 20:04.30 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:07.49 | unenough | jblack, the only thing i get when the call is disconnected on the asterisk side, is "== Spawn extension (from-callcentric, 1777..., 1) exited non-zero on 'SIP/1777....'" |
| 20:09.30 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
| 20:10.28 | unenough | seems that asterisk doesn't hang up the callcentric side properly |
| 20:10.32 | unenough | when the call ends |
| 20:10.43 | r0land | jblack [TK]D-Fender http://pastebin.com/dfb58631 <<-- is this better? :) |
| 20:11.50 | jblack | Keep me out of it. |
| 20:12.07 | [TK]D-Fender | ditto |
| 20:12.26 | r0land | its not better? |
| 20:14.55 | *** join/#asterisk jeffspeff (i=jeff@c-98-240-113-135.hsd1.ky.comcast.net) |
| 20:15.19 | jblack | unenough: Sorry. Didn't realize you answered |
| 20:15.43 | jblack | unenough: You're not seeing the dialplan? Dial up debug too, then. |
| 20:16.18 | jblack | Some combination of verbose and debug will show you the dialplan step by step, as it's being executed. |
| 20:17.06 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 20:17.32 | unenough | jblack, I do get this: Executing [1777...@from-callcentric:1] Dial("SIP/5062-b6f098a8", "SIP/123") in new stack |
| 20:18.31 | *** join/#asterisk Levonk (n=lk@adsl-75-62-135-85.dsl.lsan03.sbcglobal.net) |
| 20:20.02 | unenough | jblack, That seems to imply that it's using the from-callcentric context, which really has that as the first step (dial sip/123). but the only thing I get when i hangup is what i printed above ("Spawn...") |
| 20:22.15 | unenough | this is annoying because it's the last problem here... |
| 20:24.22 | *** join/#asterisk ManxPower (n=manxpowe@adsl-156-119-179.msy.bellsouth.net) |
| 20:28.54 | [TK]D-Fender | checkout time, heading home. |
| 20:28.56 | *** join/#asterisk mvanbaak_ (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 20:31.04 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 20:32.16 | xheliox | Can something like 'switch => Realtime/context@extensions be used in extensions.conf with other non-realtime patterns/matches? Or does it have to sit in a context by itself? |
| 20:33.11 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 20:33.32 | ManxPower | xheliox: as I understand it, that is not the correct usage of switch => |
| 20:33.52 | xheliox | Hmm? |
| 20:33.53 | ManxPower | I could be wrong, but I believe Realtime and the switch => are not used with each other. |
| 20:34.02 | ManxPower | switch specifies a REMOTE PBX to query. |
| 20:34.15 | lmadsen | there is also a realtime switch I believe.... |
| 20:34.16 | ManxPower | Realtime specifies a local or remote database server to query. |
| 20:34.21 | WimpMan | Or "remote" database? |
| 20:34.23 | *** join/#asterisk bradleyprice86 (n=bradleyp@fw.datafax.net) |
| 20:34.45 | ManxPower | WimpMan: there is no need to run the database server on the same host as Asterisk. |
| 20:34.45 | WimpMan | remote as in not within the application. |
| 20:34.53 | *** join/#asterisk Chicago (n=Chicago@c-98-223-62-171.hsd1.in.comcast.net) |
| 20:34.57 | ManxPower | no, remote as in on a different machine and ip address |
| 20:35.22 | bradleyprice86 | I am trying to edit my voicemail menu so that when I go to listen to the voicemail of a certain individual it will have an extra option where they can go to another extension. Is there an easy method of doing this? |
| 20:35.31 | WimpMan | There should be no difference if its the same machine or another. |
| 20:35.33 | ManxPower | You understand that Realtime is basically a way to store the asterisk config in a database like MySQL or Postgress, etc |
| 20:35.57 | WimpMan | yes |
| 20:35.57 | ManxPower | bradleyprice86: You would have to modify the source code. |
| 20:35.59 | [netman] | ManxPower: like FreePBX does? |
| 20:35.59 | *** join/#asterisk SirThomas (n=tomc@mail.kendeco.com) |
| 20:36.19 | ManxPower | [netman]: Since I've never used one of those evil sissy guis, I would not know,. |
| 20:36.32 | bradleyprice86 | So, it wouldn't be something that could be done from configuration files? |
| 20:36.39 | xheliox | ManxPower: all the howtos show using switch => Realtime/context@familyname -- but I'll recheck. |
| 20:36.40 | ManxPower | I prefer to stick to the nice manly Asterisk PBX, thank you very much. |
| 20:37.01 | ManxPower | xheliox: the last time I used switch => there was no such thing as Realtime |
| 20:37.04 | WimpMan | :-) |
| 20:37.38 | WimpMan | Does realtime still make hints impossible? |
| 20:37.53 | ManxPower | quickly prints up 100 bumper stickers that say "Only girly men use Trixbox" |
| 20:38.11 | Chicago | I am trying to install Speech-Recognizer-SPX in order to play ZoIP/Zasterisk. I installed pocketsphinx-0.5 and sphinxbase-0.4 into /usr/local and here is the compile error from Speech-Recognizer-SPX -> http://pastebin.ca/1187835 |
| 20:38.12 | Nugget | "gvim" is the best asterisk gui. :) |
| 20:39.00 | *** join/#asterisk rasterix (n=IceChat7@host81-155-2-150.range81-155.btcentralplus.com) |
| 20:39.07 | Chicago | Can someone on Gentoo try to install ZoIP today and let me know if you get the same error. The ZoIP info is here -> http://uc.org/read/ZoIP |
| 20:42.52 | ManxPower | I almost feel sorry for the poor sod with an FXO card thinking that the line from "zap show channel" that says "Hookstate (FXS only): Offhook" meanxs exactly that. I.e. that the information is not accurate for FXO ports. I've told him that in an e-mail but he argued with me, so I left him to figure it out on his own. |
| 20:43.22 | ManxPower | Someone told him the problem is that he didn't have busydetect= set. |
| 20:43.32 | ManxPower | Rather funny, actually |
| 20:44.17 | Nugget | heh |
| 20:46.50 | tzafrir_laptop | ManxPower, so why print out a non-relevant field? It's a bug |
| 20:47.21 | ManxPower | tzafrir_laptop: I agree. |
| 20:47.25 | *** join/#asterisk jeffspeff2 (i=jeff@c-98-240-113-135.hsd1.ky.comcast.net) |
| 20:51.50 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 20:57.20 | xheliox | ManxPower: switch does work with Realtime, I was just using the wrong family name -- thanks though. |
| 21:03.08 | *** join/#asterisk jeffspeff (i=jeff@c-98-240-113-135.hsd1.ky.comcast.net) |
| 21:03.31 | ManxPower | Good to know. |
| 21:03.47 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 21:04.10 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 21:06.52 | *** join/#asterisk RecycleBin (n=roger@stjhnf0124w-142162082245.pppoe-dynamic.nl.aliant.net) |
| 21:07.10 | RecycleBin | Can i use a regular soft modem, to call out with asterisk ? |
| 21:07.17 | ManxPower | RecycleBin: no |
| 21:07.25 | RecycleBin | Didnt think so |
| 21:07.30 | RecycleBin | What sort of card do i need ? |
| 21:07.44 | ManxPower | FXO card or ATA w/FXO |
| 21:07.53 | RecycleBin | Ty, |
| 21:07.54 | ManxPower | or a channel bank with FXO + T-1/E-1 card. |
| 21:08.15 | RecycleBin | is asterisk overkill for wanting to autodial numbers for a reminder service ? |
| 21:08.52 | ManxPower | RecycleBin: I guess it depends on what else is out there. |
| 21:09.07 | RecycleBin | True. |
| 21:10.01 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
| 21:11.45 | *** join/#asterisk Zefk (n=user@89.42.229.21) |
| 21:12.37 | Zefk | Hi. Is it possible to extract the UUI field from a PRI call ? any examples ? |
| 21:13.22 | WimpMan | Zefk: Probably only via debug, i fear. |
| 21:14.13 | ManxPower | Zefk: I don't think so. |
| 21:14.51 | ManxPower | You only get a few things from a PRI. AOC is one, and of course DNID, RDNIS, ANI, etc |
| 21:15.19 | ManxPower | Actually I don't know if Asterisk supports AOC on PRI or only on MFC/R2. |
| 21:15.59 | lmadsen | http://bugs.digium.com/view.php?id=7494 |
| 21:16.05 | lmadsen | AOC related ^^^ |
| 21:17.23 | Zefk | but ... other solutions to extract this field ? |
| 21:19.47 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 21:19.47 | rasterix | evening people |
| 21:19.51 | ManxPower | lmadsen: One of my customers is considering a 9meg MPLS connection for a private WAN. |
| 21:19.54 | rasterix | evening manx :) |
| 21:20.04 | lmadsen | ManxPower: hawtness :) |
| 21:20.08 | lmadsen | hugs MPLS |
| 21:20.12 | ManxPower | lmadsen: No internet though. |
| 21:20.14 | lmadsen | very neat protocol |
| 21:20.18 | lmadsen | that sucks |
| 21:20.32 | ManxPower | Well, WE don't see MPLS, we just see it as a fast HDLC or MLPPP connection |
| 21:20.34 | lmadsen | did some MPLS protocol analysis in school |
| 21:20.56 | ManxPower | They currently have a 3meg internet via MPLS, but we just see it as MLPPP. |
| 21:21.17 | ManxPower | I just dont like now knowing if the far end is down or not, like I can with a point to point t-1 |
| 21:22.06 | mattx86 | hey guys, I've got a weird issue with a hard-wired analog handset. you lift the handset, get dialtone, dial '47' (maybe any two digits?) and then it goes to a busy/disconnected tone. I can dial into their context (SIP handset -> Asterisk --(IAX)--> Asterisk) and it works fine. Here's my config: http://pastebin.com/m499e23e8 |
| 21:22.48 | ManxPower | So that analog phone is connected to a card, an ATA, or a channel bank? |
| 21:23.34 | ManxPower | Tell the person to try a different analog phone |
| 21:23.53 | rasterix | lol simple solutions are the best sometimes |
| 21:25.18 | mattx86 | Channel Bank |
| 21:25.40 | rasterix | < taking a break from trying to define regex expressions to parse extensions.conf << depressed :( |
| 21:29.55 | *** part/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 21:30.00 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 21:31.26 | RecycleBin | If i get the FXO card will it work over a standard telephone line .. And or does anyone know if theres a voice enabled modem i could use with a bash script to dial out and play a msg ? |
| 21:31.36 | Yourname | Hi, how can I do assisted transfers in a scenario where an agent is logged into a queue by AgentLogin()? |
| 21:32.22 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 21:32.49 | WimpMan | RecycleBin: That surely can be dome. I used to do voicemail with a modem way back then. |
| 21:33.48 | RecycleBin | Hmm, interesting |
| 21:34.00 | ManxPower | Yourname: um, press the transfer key? |
| 21:34.00 | RecycleBin | i hate decisions |
| 21:34.33 | *** join/#asterisk CanWood (n=chatzill@24.108.64.80) |
| 21:34.45 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 21:34.46 | RecycleBin | Asterisk will work with a standard telephone line though eh ? |
| 21:34.59 | WimpMan | RecycleBin: Maybe it gets easier when you check for available software :-) |
| 21:35.18 | WimpMan | RecycleBin: Sure. Anything you get an interface for, |
| 21:35.41 | RecycleBin | Cool |
| 21:35.43 | lesouvage | I heard a program about branding with a tune or piece of music in the way that a tune is linked with a brand like Nokia (the Nokia tune). I think Asterisk is doing a great job if it comes branding by using a special tune. |
| 21:35.48 | rasterix | recyclebin: why do you need to use a bash script why not use the fxo card to dial out and play the msg? |
| 21:36.42 | RecycleBin | rasterix, tbh i really don't have much knowledge in this area, im researching the best way to go about it |
| 21:37.05 | rasterix | how many phone lines do you have? |
| 21:37.11 | RecycleBin | Its just going to be one |
| 21:37.16 | WimpMan | Not and, or. User Asterisk and an FXO card or a modem and a chat script. |
| 21:37.32 | WimpMan | Thet's the question I read. |
| 21:37.53 | RecycleBin | All im trying to do, is automaticly dial out a list of numbers play a recorded msg and hang up |
| 21:38.05 | ManxPower | Asterisk does not support modems |
| 21:38.20 | RecycleBin | Right, so i need the FXO card for asterisk |
| 21:38.20 | ManxPower | Well, doesn't support using a modem. |
| 21:38.22 | rasterix | recyclebin: sounds like a simpsons episode i once saw |
| 21:38.38 | RecycleBin | Heh heh |
| 21:38.45 | WimpMan | ManxPower: 1. it used to do so, and 2. that would be the solution without Asterisk, I guess. |
| 21:39.12 | RecycleBin | So the right modem would do what i want ? |
| 21:39.25 | WimpMan | Sure. |
| 21:39.37 | Yourname | ManxPower: Is that assisted? |
| 21:39.57 | WimpMan | The question is if you'd rather spend money on an FXO card or time on finding the right software to use a modem. |
| 21:39.57 | RecycleBin | Thats probably best. Im not looking for anything complicated at this point. Originally a PBX was something i was concidering in the future |
| 21:40.08 | RecycleBin | Thats a good point |
| 21:40.27 | RecycleBin | ebay shows cheap FXO for 10-20 bux |
| 21:40.35 | Yourname | ManxPower: Also, this is on AgentLogin() where the agent is on a persistant call. |
| 21:41.07 | RecycleBin | There are even some cheap asterisk appliances |
| 21:41.27 | rasterix | recyclebin: why not set it up using asterisk... the knowledge u gain will be useful later |
| 21:41.35 | rasterix | and you will have much more control |
| 21:42.03 | WimpMan | yup |
| 21:42.08 | RecycleBin | Sounds like a good plan |
| 21:42.18 | RecycleBin | Appreciate your help |
| 21:42.18 | *** join/#asterisk ylea (i=eval@pc-123-52-239-201.cm.vtr.net) |
| 21:42.30 | *** part/#asterisk ylea (i=eval@pc-123-52-239-201.cm.vtr.net) |
| 21:42.59 | ManxPower | RecycleBin: those are not recommended |
| 21:43.30 | RecycleBin | The cheap ones ? |
| 21:43.34 | *** join/#asterisk Levonk (n=lk@adsl-76-237-14-117.dsl.lsan03.sbcglobal.net) |
| 21:43.35 | [xpl] | [Aug 28 17:40:55] WARNING[8479]: channel.c:2734 set_format: Unable to find a codec translation path from alaw to g729 |
| 21:43.35 | [xpl] | <PROTECTED> |
| 21:43.35 | [xpl] | [Aug 28 17:41:04] WARNING[15067]: translate.c:175 framein: no samples for alawtoulaw |
| 21:43.45 | Corydon76-dig | "Spend as if your job depended upon it" |
| 21:44.03 | rasterix | manxpower: its true because you have no support if things go wrong... and you dont know if any problems are your fault or a problem with the "cloned" card |
| 21:44.03 | [xpl] | who can help me ? |
| 21:44.12 | RecycleBin | Could someone point me to a link to a decent FXO card to hook into a standard phoneline |
| 21:44.16 | [TK]D-Fender | [xpl]: First, do not spam in here again, use PASTEBIN. Second did you install licenses and the G.729 codec? |
| 21:44.19 | [TK]D-Fender | ~pb |
| 21:44.20 | jbot | [~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 |
| 21:44.21 | [TK]D-Fender | ^^^^^^^^^^^^^^ |
| 21:44.30 | RecycleBin | Thats likely good advice |
| 21:44.31 | [xpl] | no |
| 21:44.42 | [xpl] | ok sorry [TK]D-Fender |
| 21:44.58 | [TK]D-Fender | [xpl]: G.729 is not free. Go to digium.com to purchase licenses for your server if you want to transcode it |
| 21:45.12 | rasterix | some of the cloned cards are absurdly cheap... but i wouldnt go there |
| 21:45.23 | WimpMan | RecycleBin: There are lots of stores for voip equipment. Take a look there. |
| 21:45.29 | RecycleBin | K |
| 21:45.47 | rasterix | recyclebin: where are you located |
| 21:45.54 | RecycleBin | Canada |
| 21:46.05 | RecycleBin | Eastern |
| 21:46.06 | rasterix | ah... uk here so cant help |
| 21:46.11 | RecycleBin | Np |
| 21:46.14 | rasterix | all suppliers i use are uk based |
| 21:46.21 | WimpMan | RecycleBin: Then Sangoma would be nearest, I guess. |
| 21:46.25 | [TK]D-Fender | RecycleBin: How many lines? |
| 21:46.41 | RecycleBin | Just one to start |
| 21:46.48 | [TK]D-Fender | RecycleBin: What kind of usage? |
| 21:46.48 | RecycleBin | Possibly more in the future |
| 21:46.59 | rasterix | no experience of sangoma analogue but pri cards are great |
| 21:47.06 | RecycleBin | I just want to auto dial numbers in a list and play a msg to start |
| 21:47.22 | rasterix | recyclebin: then what? |
| 21:47.30 | WimpMan | rasterix: Some people say thei're the best. |
| 21:47.31 | [TK]D-Fender | RecycleBin: http://www.canadianvoipstore.com/home.php |
| 21:47.33 | *** part/#asterisk orkid (n=orkid@unaffiliated/orkid) |
| 21:47.35 | *** join/#asterisk IsUp (n=nocturne@88.255.77.200) |
| 21:47.39 | IsUp | hiya hiya hi |
| 21:47.47 | IsUp | Sangoma fixed my Zaptel problem :> |
| 21:47.48 | RecycleBin | Hang up, really. If things work out i want them to press a number to connect to a live person |
| 21:47.53 | [TK]D-Fender | RecycleBin: Sangoma A200d + 1 FXO module (2ports) |
| 21:47.56 | RecycleBin | Ty |
| 21:48.01 | [xpl] | [TK]D-Fender : i just buy 20 channels |
| 21:48.11 | *** join/#asterisk `paul (n=paul@125.252.68.126) |
| 21:48.20 | rasterix | wimpman: had a digium card before, switched to sangoma and have had comments that audio is better from ppl that didnt know we switched |
| 21:48.24 | [xpl] | [TK]D-Fender : and i have the registration code ... and now who can i registred and instaled the codec? |
| 21:48.35 | WimpMan | And one more customer satisfied with sangoma support :-) |
| 21:48.37 | [xpl] | [TK]D-Fender : can you help me please ? |
| 21:48.50 | rasterix | lol tbh sangoma support was fast |
| 21:48.54 | rasterix | but not helpful |
| 21:49.23 | [TK]D-Fender | [xpl]: 20? thats a lot... |
| 21:49.29 | IsUp | its helpful i think |
| 21:49.35 | IsUp | they connected to my server with SSH |
| 21:49.37 | `paul | can i run a script after a call has ended? (set somthing in the DB then call then after call update the id in the DB) |
| 21:49.38 | WimpMan | knows people who are very satisfied with their support. |
| 21:49.39 | [xpl] | yes i wanna buy 2 |
| 21:49.39 | IsUp | fixed all problems etc.. |
| 21:49.43 | [TK]D-Fender | [xpl]: Go read the install guide |
| 21:49.44 | [xpl] | and i buy 20 ....it's okay |
| 21:49.51 | Yourname | http://lists.digium.com/pipermail/asterisk-dev/2003-November/002320.html -> I have a question that looks just like this. |
| 21:50.03 | rasterix | isup: had an interupt issue with a dell server... they couldnt resolve it |
| 21:50.21 | [xpl] | i most download the registred and chmod 755 registred ./registred ? |
| 21:50.24 | [xpl] | and that is it ? |
| 21:50.26 | rasterix | isup: i resolved it by throwing the nasty dell server in the bin and buying an hp |
| 21:50.40 | fish-bulb | WimpMan: how is stating that someone noticed a difference in using their product have anything to do with the quality of their support? |
| 21:50.51 | IsUp | it's just about dell lol |
| 21:50.53 | ph8 | i've bought one IAX outbound trunk - i'm right in thinking that only allows one channel out right? |
| 21:50.54 | IsUp | i am using HP |
| 21:50.57 | ph8 | might be a silly question :p |
| 21:51.11 | [xpl] | http://downloads.digium.com/pub/register/ |
| 21:51.16 | WimpMan | fish-bulb: Nothing. Why do you think so? |
| 21:51.18 | rasterix | isup: im never buying dell again... end of |
| 21:51.31 | angryuser | ph8: depends on you tarif plan |
| 21:51.37 | fish-bulb | WimpMan: I guess I just misread what you replied with back there |
| 21:52.33 | WimpMan | also knows people not in this channel. Well actuallu I don't really know anyone here. |
| 21:52.44 | ph8 | angryuser: So one IAX2 trunk is capable of multiple channels? |
| 21:52.51 | jblack | ph8; yes. |
| 21:53.01 | ph8 | that's interesting |
| 21:53.05 | jblack | That's one of iax2's benefits that sip doesn't have. Trunking. |
| 21:53.11 | rasterix | last week i phoned dell support... was told the contact id been dealing with was outside having a smoke and would call back |
| 21:53.46 | rasterix | phoned back twenty minutes later... was told he had been on holiday the last week and would be back in a weeks time... i spoke to him the day before! |
| 21:53.51 | *** join/#asterisk awannabe (n=brad@ip24-251-147-29.ph.ph.cox.net) |
| 21:53.53 | ph8 | haha that's awesome |
| 21:53.54 | lowtek | Yea, that was me, sorry. I spilled coffee on my script that day too. It's rough living ove here in India, we only get a few breaks a day. |
| 21:53.57 | ph8 | i appear to be able to make multiple calls on the one |
| 21:53.58 | ph8 | thanks |
| 21:54.07 | awannabe | Hello, has anyone used the GXW-4004 or GXW-4008 gateways? |
| 21:54.14 | fish-bulb | rasterix: One time I called Mediacom support and was upset about a three week service outage at my home, and asked to speak to their manager/supervisor and they told me "No." |
| 21:54.43 | rasterix | fish-bulb: lol i love support lines... i try to think of them as a game |
| 21:55.21 | ph8 | what sort of kit should i be looking at if I want to provide my own trunks? Presumably i need a contract with BT (UK) to connect to the PSTN? |
| 21:55.21 | jblack | works hard to be nice when calling customer support. When customer support doesn't return the favor by being helpful, it's time to replace the company |
| 21:55.36 | fish-bulb | ph8: I recommend reading up on the "trunk=" option for IAX2, it greatly reduces the bandwidth needed when making multiple calls |
| 21:55.37 | lowtek | jblack: What company? |
| 21:55.42 | rasterix | tip for all uk dell customers: claim your a large company not a small business and you get the ireland call centre not the indian centre |
| 21:55.50 | jblack | I try to be nice to every company. |
| 21:56.12 | ph8 | thanks fish-bulb - i got bored reading the pdf, i find them really hard to concentrate on, the book should arrive tomorrow or monday :-) |
| 21:56.12 | viraptor | is there any benefit in setting audiobuffers > 2 for meetme when I already pass all the calls through Local channel with dejittering enabled? |
| 21:57.05 | [xpl] | [TK]D-Fender : who can i see now if the license is enabled ? |
| 21:57.09 | rasterix | i dont want to knock indian support centres... but since the "smoke break turning into a two week vacation" incident i have no time for them (multiple problems before this too) |
| 21:57.17 | fish-bulb | ph8: lol, yeah I understand. The book is a great way to start |
| 21:57.28 | jblack | rasterix: Yeah. Time to replace the company. |
| 21:57.28 | [xpl] | [TK]D-Fender : i just put the license . but the error is the same ..:( |
| 21:57.46 | rasterix | jblack: time to blow up dell headquarters :) |
| 21:57.51 | jblack | rasterix: It's your money, and someone probably wants it more than you. |
| 21:57.57 | jblack | Pardon, wants it more than they do. |
| 21:58.21 | rasterix | lol i was gonna say... no one wants my money more than me :) |
| 21:58.36 | fish-bulb | it's an odd change that there for a while I was getting more people calling from India for support than US or UK residents |
| 21:58.51 | jblack | rasterix: If that were true, you'd be running around naked, with a fistfull of dollars. |
| 21:59.23 | rasterix | jblack: luckily my webcam is not on |
| 21:59.41 | jblack | You also wouldn't have a webcam. |
| 22:00.21 | jblack | or a house, or a computer, or internet, or food. |
| 22:00.29 | ph8 | what sort of kit should i be looking at if I want to provide my own trunks? Presumably i need a contract with BT (UK) to connect to the PSTN? |
| 22:01.54 | bkw_ | file: http://gallery.me.com/brian.west#100157/100_1174&bgcolor=black |
| 22:02.30 | jblack | I get pissed off when my daughter's english teacher's handouts are loaded with gammar errors. |
| 22:02.40 | bkw_ | jblack: AHAHHAHAH |
| 22:02.42 | bkw_ | classic |
| 22:02.44 | rasterix | lol @ gammar errors |
| 22:02.44 | *** join/#asterisk jeffspeff2 (i=jeff@c-98-240-113-135.hsd1.ky.comcast.net) |
| 22:03.26 | bkw_ | jblack: what did the teacher say? |
| 22:04.01 | rasterix | "those are not grammar errors" |
| 22:04.08 | jblack | Among others, "One of the most important papers - the research paper will be worth more". That _should_ read 'One of the most important papers, the research paper, will be worth more' |
| 22:04.30 | jblack | To add to the irony, this is the AP english teacher |
| 22:04.53 | rasterix | preferably with a full stop to end the sentence |
| 22:05.04 | jblack | An english teacher that makes grammar mistakes is like a math teacher that can't reliably add or subtract. |
| 22:05.15 | *** join/#asterisk jeffspeff2 (n=jeff@c-98-240-113-135.hsd1.ky.comcast.net) |
| 22:05.51 | [xpl] | Presario*CLI> show g729 |
| 22:05.51 | [xpl] | No such command 'show g729' (type 'help' for help) |
| 22:06.01 | rasterix | jblack: can i offer you some valium? |
| 22:06.29 | jblack | Thanks, but I don't like valium. |
| 22:06.48 | *** part/#asterisk bradleyprice86 (n=bradleyp@fw.datafax.net) |
| 22:07.07 | *** join/#asterisk beernuts (n=mattfox7@123.211.219.199) |
| 22:08.06 | rasterix | did you try putting her handout through the vigorous grammatical check that is microsoft word? |
| 22:08.20 | rasterix | maybe you can recommend she buys a copy |
| 22:08.36 | jblack | Nah. I know better than to go Nazi on an english teacher. |
| 22:08.45 | Chicago | Anybody here using the voip overlay on Gentoo? |
| 22:08.56 | rasterix | save that for the german language teacher eh |
| 22:08.58 | rasterix | brb |
| 22:10.57 | rasterix | interesting that poor grammar should come up since im doing battle with trying to do lexical analysis on extensions.conf |
| 22:11.42 | rasterix | mainly doing battle with my own inability to type in a regex without making a typo though |
| 22:12.23 | rasterix | one problem im having is a string within a string |
| 22:12.58 | rasterix | i.e. ${myvariable:${someothervariable}} |
| 22:13.07 | rasterix | grrr smileys |
| 22:13.36 | rasterix | ${myvariable: ${someothervariable}} |
| 22:14.10 | lmadsen | rasterix: most of us won't see the smiley |
| 22:14.14 | lmadsen | I don't enable them in IRC |
| 22:14.19 | lmadsen | (or anywhere for that matter...) |
| 22:14.31 | rasterix | ah ok im new to irc |
| 22:14.42 | rasterix | my client shows smileys everywhere |
| 22:14.46 | rasterix | >< |
| 22:14.52 | lmadsen | :) |
| 22:15.09 | jeev | leif! |
| 22:15.29 | rasterix | but the problem is im trying to break it down into tokens |
| 22:15.39 | lmadsen | Leif's not here man |
| 22:15.50 | jeev | is it his beautiful wife? |
| 22:15.52 | rasterix | and i have a variable token within a variable token... i cant think of another language where that could happen |
| 22:16.10 | lmadsen | Leif doesn't believe in marriage |
| 22:16.56 | rasterix | < carries on rambling about his token problem even though noone is listening |
| 22:16.59 | jeev | i have exten => *82,1,Dial(IAX2/iaxaccount/${CHANNEL:4:4}${EXTEN}) set up.. all works well, but i need to set up exten => *67,1,Dial(IAX2/iaxaccount/${CHANNEL:4:4}${EXTEN}) and i dont know how ot make it pass the *67 part because *67 will temporary set caller id but i know it's not *67, i'll change it later |
| 22:17.42 | lmadsen | rasterix: I'm listening, but have no input |
| 22:17.44 | jeev | so the *82 and stuff is cool, my other box accept sit.. but the sending iax for *67 doesn't have an extension, it can't find it. i need to pass *67 |
| 22:18.24 | [xpl] | who can i see if the license of the g729 codec is valid and works ? |
| 22:18.32 | rasterix | lmadsen: do you think im right in thinking having a variable within a variable hapens in the asterisk language but no other programming language? |
| 22:18.56 | lmadsen | rasterix: possibly... in PHP I think that means something weird |
| 22:19.05 | jeev | scratches self |
| 22:19.29 | rasterix | i just wish they had used a substr() function like every other language... *sigh* |
| 22:19.50 | lmadsen | eh? |
| 22:20.37 | rasterix | ${myvariable:5} = substr(${myvariable},5) |
| 22:21.19 | *** join/#asterisk EI5GTB (n=EI5GTB@193.120.192.61) |
| 22:23.43 | rasterix | i guess noone apart from me thinks its an issue |
| 22:24.44 | *** join/#asterisk Rajpoot (n=Rajpoot@75-48-229-227.lightspeed.prtgmi.sbcglobal.net) |
| 22:24.56 | rasterix | *lonely* |
| 22:25.27 | Rajpoot | i have a grandstream handytone-486 voip box. i would like to use it with a voip service to process credit cards. is this possible, and what codec should i use? |
| 22:26.08 | rasterix | grandstream *shudders* |
| 22:26.50 | rasterix | i have a grandstream voip phone... great door stop :) |
| 22:27.09 | aliver | The grandstream gpx2000 isn't that bad. |
| 22:27.14 | aliver | I've had good luck with those. |
| 22:27.29 | aliver | In fact, I'd say they work better for me in terms of administration than the polycom IP320 |
| 22:27.33 | rasterix | that might be the one i have... |
| 22:27.36 | aliver | but the audio quality isn't quite as good. |
| 22:27.41 | rasterix | does it have blue keys? |
| 22:27.50 | aliver | Very light blue |
| 22:27.55 | Rajpoot | if i got that one, could i use it for data, or is there any way to use the box i have already for data? |
| 22:27.55 | aliver | kind of grey blue |
| 22:27.56 | viraptor | [xpl]: did you load it? have you got codec_g729a.so in your lib/modules dir? |
| 22:27.56 | rasterix | yeah |
| 22:28.51 | aliver | if I get the G729 codec all licensed from our friends at Digium do I have to rencode any of my sound files? Asterisk will transcode right? |
| 22:29.21 | jblack | right. |
| 22:29.38 | aliver | cool |
| 22:29.44 | jblack | If you wait another few years, you can have g729 for free |
| 22:29.44 | rasterix | aliver: yah thats what i have... tbh it wasnt as bad after the last firmware upgrade i applied |
| 22:29.49 | [xpl] | viraptor : yes |
| 22:29.56 | [xpl] | viraptor : i have the codec instaled... |
| 22:30.01 | bkw_ | jblack: more like 10 years |
| 22:30.06 | rasterix | aliver: that was probably over a year ago |
| 22:30.06 | aliver | rasterix yeah, before the firmware update it was NOT a good phone. |
| 22:30.07 | [xpl] | viraptor : but i can see the canneles |
| 22:30.15 | [xpl] | viraptor : but i can see the *channels...:( |
| 22:30.15 | jblack | I thought the patents expire in 2014/ |
| 22:30.29 | [xpl] | Presario*CLI> show g729 |
| 22:30.30 | aliver | I wonder why Speex hasn't caught on? |
| 22:30.33 | [xpl] | No such command 'show g729' (type 'help' for help) |
| 22:30.40 | aliver | "show audio codecs" |
| 22:30.48 | *** join/#asterisk mog (n=mogorman@c-68-62-219-86.hsd1.al.comcast.net) |
| 22:30.48 | *** mode/#asterisk [+o mog] by ChanServ |
| 22:30.50 | lmadsen | *CLI> module load codec_g729.so |
| 22:30.52 | aliver | Does speex just suck or something? |
| 22:30.54 | [xpl] | <PROTECTED> |
| 22:30.55 | jblack | lack of inertia, maybe too flexible. |
| 22:30.59 | rasterix | aliver: i think it has a high processor overhead |
| 22:31.17 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 22:31.28 | aliver | rasterix I see. That would preclude it from working on low-CPU phones, then. |
| 22:31.31 | bkw_ | speex and g729 can be about the same in terms of CPU depending on quality settings |
| 22:31.45 | *** join/#asterisk mog (n=mogorman@c-68-62-219-86.hsd1.al.comcast.net) |
| 22:31.45 | *** mode/#asterisk [+o mog] by ChanServ |
| 22:31.47 | aliver | yeah, I thought it had both narrow and wideband modes. |
| 22:31.54 | bkw_ | and ultrawideband |
| 22:31.59 | bkw_ | but 32k audio is a hack |
| 22:33.18 | `paul | when using manager api direct action... in the channel option should i just write ex. SIP/100?? or the whole SIP/100-xxxxxxx?? |
| 22:33.29 | rasterix | aliver: have you tried linksys spa-942? |
| 22:33.42 | viraptor | [xpl]: does 'module load codec_g729a' do anything interesting? |
| 22:34.45 | aliver | rasterix no but I've tried the 922 |
| 22:35.00 | [xpl] | yes |
| 22:35.38 | rasterix | aliver: and you prefer the grandstream? |
| 22:35.48 | aliver | only because of the cost |
| 22:35.56 | bkw_ | 922? |
| 22:35.56 | aliver | and the built in webserver config is faster to admin |
| 22:36.02 | aliver | SPA-922 |
| 22:36.05 | bkw_ | ah ok |
| 22:36.20 | rasterix | spa has a built in webserver? |
| 22:37.24 | bkw_ | yes |
| 22:37.28 | bkw_ | for the admin gui |
| 22:37.34 | rasterix | yeah i know |
| 22:37.34 | viraptor | [xpl]: what does 'yes' mean? :D did it load? did you get any messages? |
| 22:37.44 | aliver | yeah, it's prettier but not as functional as the grandstream's |
| 22:37.50 | [xpl] | http://pastebin.com/dc69ada4 |
| 22:38.02 | [xpl] | viraptor : http://pastebin.com/dc69ada4 |
| 22:38.08 | jblack | wow. music encoded in gsm sounds _terrible_ in speex |
| 22:38.38 | *** join/#asterisk angryuser (n=angryuse@95.250.146.195.dynamic.adsl.abo.nordnet.fr) |
| 22:38.51 | viraptor | [xpl]: no - I meant after running "module load codec_g729a" |
| 22:39.00 | rasterix | aliver: well i had a lot of audio quality problems on the grandstream... and he handsfree was almost useless |
| 22:39.10 | [xpl] | [Aug 28 18:38:59] WARNING[16924]: loader.c:649 load_resource: Module 'codec_g729a' could not be loaded. |
| 22:39.17 | [xpl] | [Aug 28 18:38:59] WARNING[16924]: loader.c:363 load_dynamic_module: Error loading module 'codec_g729a': /usr/lib/asterisk/modules/codec_g729a.so: undefined symbol: ast_register_translator |
| 22:40.00 | aliver | rasterix yes, the polycom's have MUCH better hand-free/intercom quality and the audio quality can be a bit better. Additionally, sometimes the grandstream's have hardware problems on the handset. |
| 22:40.32 | *** join/#asterisk knarfly (n=knarfly@c-75-74-155-198.hsd1.fl.comcast.net) |
| 22:40.42 | aliver | But for $60 bucks a phone they are cheap to replace (gpx-2000) |
| 22:40.54 | aliver | We usually have 3-4 extras on hand. |
| 22:40.58 | [xpl] | ? |
| 22:41.04 | rasterix | aliver: yes but even when you replace them they are still rubbish :P |
| 22:41.20 | aliver | true, but cheap, working rubbish. |
| 22:41.31 | aliver | I'm not the poor sap who has to talk on them all day. |
| 22:41.44 | viraptor | [xpl]: are you sure that you're using right binaries for your version of asterisk? |
| 22:41.51 | rasterix | aliver: < doesnt like the poor sap's moaning at him about the bad audio quality |
| 22:41.52 | [xpl] | i think so |
| 22:42.12 | [xpl] | Asterisk 1.4.18 currently running |
| 22:42.17 | [xpl] | i have |
| 22:42.17 | aliver | rasterix I understand that, I hear some of that too. However, until recently, I heard the same thing about the IP320 polycoms. |
| 22:42.34 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 22:42.35 | rasterix | aliver: "so we replaced our old pbx for a new one that everyone moans about the quality..." |
| 22:43.45 | viraptor | [xpl]: just in case -> check if this one works: http://downloads.digium.com/pub/telephony/codec_g729/asterisk-1.4/x86-32/codec_g729a_v34_i686.tar.gz |
| 22:43.51 | voxter | IP320's shouldnt have any quality problems, i would point fingers at your connectivity to the PSTN in that case |
| 22:44.02 | viraptor | (unless it's x86-64, then change the dir) |
| 22:44.18 | r0land | hello all.. |
| 22:44.32 | aliver | We trunk our lines out to an internet provider now. The quality went WAY down from when we used to do PRI |
| 22:44.48 | [xpl] | viraptor : ok |
| 22:45.10 | aliver | I guess I need to get the G729 codec, too. |
| 22:45.15 | r0land | any1 has an advice about sipphone->asterisk-> sipura... having calls overlapping with eachother! in other words, if i call sipura, and recieve a call at the same time on another sipura.. the lines overlaps and both calls gets merged.... |
| 22:45.29 | jblack | ouch. the enzyte guy got 25 years in jail? |
| 22:46.07 | r0land | jblack any advice about the overlapping issue ?! |
| 22:46.21 | rasterix | aliver: using those grandstreams probably doesnt help... there is enought problems with voip trunks without your handset letting you down a the end... |
| 22:46.25 | aliver | sipuras are pure unadulterated crap. |
| 22:46.26 | knarfly | Help my asterisk message log is showing someone is trying to login from a remote location |
| 22:46.37 | [xpl] | Presario*CLI> show g729 |
| 22:46.37 | [xpl] | 0/0 encoders/decoders of 20 licensed channels are currently in use |
| 22:46.37 | [xpl] | Presario*CLI> |
| 22:46.43 | aliver | rasterix it's bad with our polycoms too. |
| 22:46.45 | knarfly | they are not getting in but there are hundreds of attempts happening just today |
| 22:46.49 | [xpl] | thank you verry verry much viraptor |
| 22:46.49 | jbot | [xpl]: pas de quoi |
| 22:46.49 | r0land | aliver tending to think so.. |
| 22:47.03 | viraptor | [xpl]: no problem |
| 22:47.06 | rasterix | aliver: linksys!!! |
| 22:47.07 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 22:47.15 | r0land | knarfly some1s brutforcing his way into ur box :) |
| 22:47.26 | aliver | rasterix do you have a bunch of linksys? |
| 22:47.28 | viraptor | [xpl]: just add it to your /etc/asterisk/modules.conf if you want it to load every time automaticaly |
| 22:47.36 | rasterix | about 40 |
| 22:47.39 | r0land | aliver ive got 9 sipura... |
| 22:47.50 | aliver | rasterix like them? |
| 22:47.53 | knarfly | well they are trying from an IP address that's in of all places my home town of Dallas, TX which 1,500 miles away from here. |
| 22:47.53 | rasterix | and had no problems with any of them |
| 22:47.58 | rasterix | yes very much so |
| 22:48.06 | jblack | poor jamaica |
| 22:48.06 | aliver | hmm. might have to get some of those, then. |
| 22:48.11 | rasterix | all spa-942s |
| 22:48.26 | knarfly | no one I know hacks at pbx stuff like me so I don't think it's anyone I know |
| 22:48.31 | r0land | rasterix whts the default codec ur using on ur spa! |
| 22:48.35 | aliver | r0land I've had sipura FXS's do all kind of loony shit. |
| 22:48.51 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 22:49.01 | r0land | aliver well currently, whenever i speak to some1, if the other end spoke at the smae time! the sound quality deteriorates.. |
| 22:49.50 | r0land | knarfly i spent like 2 hours to get FAX go through sipura 3102, even though as they mentiond its set to work by default.. default my a44... |
| 22:49.57 | [xpl] | viraptor : now its work fine but i can her nothing ... and i have this error[Aug 28 18:48:55] WARNING[18061]: translate.c:175 framein: no samples for alawtolin |
| 22:51.20 | knarfly | I'm trying to get a sip phone on Asterisk #2 use the X101P on Asterisk #1 in my LAN but I keep getting "rejected because extension not found" error |
| 22:51.27 | aliver | r0land well you could try using QoS on your router for UDP on ports 1024-65535 for RTP |
| 22:51.33 | aliver | or whatever you are actually using |
| 22:51.39 | viraptor | [xpl]: alawtolin looks like pcma to wav translation... (but I might be wrong) are you trying to record something / use a conference? |
| 22:51.49 | [xpl] | yes |
| 22:52.02 | rasterix | roland: alaw |
| 22:52.21 | [xpl] | y want to use a conference |
| 22:53.14 | aliver | When I do a "sip show peer <my_upstream_internet_itsp>" do the codecs listed there come from their sip switch or do they come from what I've configured in sip.conf ? |
| 22:54.00 | aliver | I'm guessing from sip.conf |
| 22:54.19 | viraptor | [xpl]: run "module show like codec_" - if you don't have codec_alaw, just run "module load codec_alaw" |
| 22:54.52 | r0land | rasterix alaw= g711u !? |
| 22:55.12 | [xpl] | Module 'codec_alaw' already exists. |
| 22:55.38 | rasterix | g711a!!! |
| 22:55.39 | knarfly | how do I clear /var/log/asterisk/message file - mine is full of some phreaker attempt to get into my * server |
| 22:55.51 | r0land | aliver if u do sip show peer it shows registerd users at the moment |
| 22:56.17 | *** join/#asterisk Levonk (n=lk@adsl-75-62-143-71.dsl.lsan03.sbcglobal.net) |
| 22:56.21 | aliver | r0land not "sip show peers" I'm talking about "sip show peer <SIPID>" |
| 22:57.11 | r0land | aliver well still the same!! it gives u the info about the state of tht sipID! |
| 22:57.13 | aliver | Those give completely different output. |
| 22:57.29 | r0land | aliver it works for troubleshooting.. |
| 22:57.34 | aliver | Okay, but the state as set by the sip.conf or by the peer itself. |
| 22:57.42 | r0land | by the peer |
| 22:58.10 | r0land | aliver the way u could notice tht.. if in sip.conf youve set host as dymanic! in sip show peer sipID itll show u its current IP |
| 22:58.16 | r0land | thus, peer's state:) |
| 22:58.21 | r0land | logicaly speaking.. |
| 22:58.36 | RecycleBin | Where would asterisk take a list of numbers for auto dialing from.. a database or a flat file ? |
| 22:59.03 | RecycleBin | And could it take numbers from a database ? |
| 22:59.15 | rasterix | recyclebin: connect to a mysql server |
| 22:59.15 | aliver | r0land the same doesn't seem true about the audio codecs though. |
| 22:59.20 | RecycleBin | Nice |
| 22:59.27 | RecycleBin | Thats awsome |
| 22:59.55 | RecycleBin | i don't even have my hardware and i love it already |
| 23:00.04 | r0land | aliver well of course! since if u defvine the codec in [general] all peers that doesnt have any codec defined for them would take the default one (the one set in the default) |
| 23:00.07 | aliver | r0land pretty sure the codecs shown there are not actually supported by the peer. but I'll experiment and find out. |
| 23:00.18 | rasterix | recyclebin: or you can call agi |
| 23:00.25 | rasterix | and do it however you want |
| 23:00.30 | aliver | r0land okay that sounds reasonable. |
| 23:00.58 | Ryushin | I just upgraded a server that was mostly working fine except for a couple of issues. I upgraded the kernel, asterisk, zaptel, and wanpipe. I'm using pretty much the identical config at another location and they are working fine. When I make a call off the PRI, it just kinda hangs there not going anywhere. Incoming calls aren't coming down the PRI either. |
| 23:01.14 | Ryushin | I've called my provider of the PRI, and they are going to have some call me back that can do some tests. |
| 23:01.37 | r0land | rasterix am currently facing all sort of troubles with sipura! voice overlapping from different sipuras... lines breaking... lines breaking after different intervals of time! |
| 23:01.40 | r0land | rasterix its a mess |
| 23:01.43 | rasterix | ryushin... have you upgraded ur sangoma driver? |
| 23:01.46 | aliver | Is there any way to see what codecs are being used on an in-progress call? |
| 23:01.48 | Ryushin | I'm hesitant to think that it's the other side, but that I've messed something up. |
| 23:01.55 | Ryushin | rasterix: Yes, to the latest one. |
| 23:01.59 | rasterix | ahh ok |
| 23:02.12 | RecycleBin | whats agi ? |
| 23:02.15 | rasterix | what does your zaptel.conf have in it? |
| 23:02.21 | Ryushin | It seems to insert just fine, and wanrouter looks like it's tarting correctly. |
| 23:02.36 | rasterix | check to see if your dchannel is going up and down |
| 23:02.42 | r0land | aliver if u do sip show channels you get to see the channels connected |
| 23:03.14 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 23:03.17 | rasterix | do you have the line dchan= in your zaptel.conf? |
| 23:03.40 | Ryushin | http://www.pastebin.ca/1187947 |
| 23:03.41 | *** part/#asterisk `paul (n=paul@125.252.68.126) |
| 23:04.04 | r0land | aliver http://www.asteriskguru.com/tutorials/cli_cmd_14.html |
| 23:04.16 | Ryushin | Yes. This was a working server. I just upgraded everything to a newer version. I kept the same config files. |
| 23:04.16 | aliver | r0land I know that. but can you see what audio codc they are using. |
| 23:04.24 | rasterix | ryushin: i think dchan may need to be replaced by |
| 23:04.27 | rasterix | hardhdlc |
| 23:04.35 | aliver | sip show channels just shows you calls and channels |
| 23:04.48 | Ryushin | Okay, I'll try that. |
| 23:04.55 | rasterix | no no wait |
| 23:05.00 | rasterix | let me check i got that right! |
| 23:05.36 | rasterix | yes thats right |
| 23:05.56 | Ryushin | So I put hardhdlc = 48? |
| 23:06.00 | rasterix | yup |
| 23:06.10 | Ryushin | ztcfg -vvvv is showing an error. |
| 23:06.39 | Ryushin | But yea, I think you're right. |
| 23:06.40 | rasterix | u recompiled everything? |
| 23:06.48 | Ryushin | The newer server is showing the same syntax. |
| 23:06.51 | Ryushin | Let me try again. |
| 23:08.11 | *** join/#asterisk russellb (n=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 23:08.11 | *** mode/#asterisk [+o russellb] by ChanServ |
| 23:08.39 | aliver | sip show channels |
| 23:08.44 | aliver | that'll do it. shows the codec in use |
| 23:08.53 | r0land | :) |
| 23:08.59 | aliver | NOT "show channels" |
| 23:09.22 | aliver | ;P |
| 23:09.40 | Ryushin | rasterix: That solved it after restarting wanrouter. I wonder when the syntax changed. |
| 23:10.07 | rasterix | i am not sure |
| 23:10.22 | rasterix | but the funny thing is if you run the wanpipe install script |
| 23:10.33 | rasterix | it creates a zaptel with the old syntax... helpful! |
| 23:10.47 | Ryushin | The new server I had the wanrouter scripts build the zaptel.conf file. Though I would try it out and see if it would work. I never noticed that they changed a line. |
| 23:11.47 | rasterix | no and they dont exactly make a point of it on their support pages... i guess they like support calls to keep them company |
| 23:12.06 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
| 23:12.38 | rasterix | ryushin: you are my first succesfully solved asterisk support case :) |
| 23:12.43 | Ryushin | The latest version of wanpipe made the correct config file with the correct syntax. Well, zaptel is for digium. But yea, Sangoma should have a better support faq or knowledge base. |
| 23:13.32 | Ryushin | rasterix: And it was a pretty advanced question at that. Good job! Thank you, and I really mean it. You saved me from hours of work. |
| 23:13.58 | rasterix | np |
| 23:14.19 | Ryushin | Well, on to the next task of replacing a failed hard drive. |
| 23:15.17 | *** join/#asterisk grantm (n=grant@68.142.138.4) |
| 23:15.32 | rasterix | < hands ryushin a screwdriver and goes to make a cup of tea |
| 23:15.50 | Ryushin | I guess what would have been best if the ztcfg or wanrouter would error out about the bad syntax. |
| 23:16.15 | rasterix | ryushin: that would be "sensible" |
| 23:19.28 | RecycleBin | Can asterix predict if a number is not connected, or if it got an answering machine etc ? |
| 23:20.02 | rasterix | recyclebin: asterisk the future of telephony |
| 23:20.10 | RecycleBin | i guess soo |
| 23:20.11 | RecycleBin | so |
| 23:20.28 | rasterix | you should get a copy of that |
| 23:20.33 | RecycleBin | OH |
| 23:20.40 | RecycleBin | i didnt realize it was a book |
| 23:20.47 | rasterix | there is an application for detecting answering machines |
| 23:20.57 | rasterix | i am not sure how effective it is |
| 23:21.07 | RecycleBin | what about numbers not in service ? |
| 23:21.34 | rasterix | the call wont be connected |
| 23:21.43 | rasterix | so yes you will be able to monitor for that |
| 23:21.46 | *** join/#asterisk juniowww (n=juniowww@201.86.13.74.adsl.gvt.net.br) |
| 23:21.47 | RecycleBin | Ty |
| 23:22.04 | beernuts | if you are connected by isdn you can get cause codes that indicate these conditions....search "isdn cause codes" |
| 23:22.05 | florz | RecycleBin: depends on your connection to the PSTN |
| 23:22.13 | florz | or to the destination, in general |
| 23:22.23 | RecycleBin | Regular landline |
| 23:22.27 | rasterix | he is connecting via analogue |
| 23:22.37 | juniowww | Hello, anyone can help-me? |
| 23:22.49 | florz | juniowww: nope, that's not possible |
| 23:22.58 | juniowww | why ? :o |
| 23:23.07 | rasterix | juniowww: i can order you pizza... will that help? |
| 23:23.16 | beek | juniowww: Don't ask if you can ask -- just ask. |
| 23:23.22 | RecycleBin | can you get some hamburger meat on that |
| 23:23.28 | beek | If someone can answer the question you will get your answer. |
| 23:23.31 | beernuts | oh well not then....human progress detection then |
| 23:23.54 | RecycleBin | Ty beernuts |
| 23:23.57 | juniowww | can i download of codec g729 trial or to test ? |
| 23:24.27 | florz | RecycleBin: generally, analog doesn't signal any error conditions, or progress information in general, in any easily machine-parseable way |
| 23:24.35 | RecycleBin | Ok, |
| 23:24.48 | beek | juniowww: That's licensed code, so I doubt it. It's only $10/channel though from Digium, so purchase one channel and test away. |
| 23:25.04 | RecycleBin | ISDN mite be an option then depending on price |
| 23:25.23 | florz | RecycleBin: pretty much anything but analog should do |
| 23:25.30 | juniowww | i pay and i test .. can i test before buy |
| 23:25.45 | RecycleBin | Cool |
| 23:25.51 | rasterix | recyblebin: a bri card is cheap... but you will need access to an isdn line (obviously) |
| 23:26.06 | beek | juniowww: Contact Digium. I don't think so -- but $10 is hardly an expensive test. |
| 23:26.11 | fish-bulb | juniowww: Digium doesn't offer any trial license, or anything of the like |
| 23:26.31 | fish-bulb | juniowww: what is it that you are interested in testing with it? |
| 23:26.38 | juniowww | ok.. because my card is not international .. im from brazil :o |
| 23:27.02 | fish-bulb | juniowww: What card are you talking about? |
| 23:27.10 | juniowww | credit card |
| 23:27.24 | rasterix | juniowww: Under patent law, it is a legitimate use to study or experiment with a patented technology without paying for a patent license. You only have to pay royalty fees if you sell a product based on the technology or use the technology in products internally (this includes using the codec in a commercial setting, even if you aren't selling your platform, but only a service). |
| 23:27.48 | florz | rasterix: where do you know brasil's patent law from? |
| 23:28.11 | rasterix | An opensource implementation of G.729 codec can be found at http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/. |
| 23:29.28 | juniowww | my server is amd :l |
| 23:29.47 | rasterix | i dont know patent law i read that on voip-info |
| 23:29.47 | juniowww | This code only runs on modern Intel processors that are supported by Intel IPP libraries |
| 23:30.07 | juniowww | i ll get another creditcard to buy this codec :) |
| 23:30.21 | rasterix | personally id just buy 1 copy to test it |
| 23:30.26 | rasterix | much less hassle |
| 23:31.38 | fish-bulb | juniowww: do you have someone in the US or with an international credit card that could purchase it for you? |
| 23:32.37 | Corydon76-dig | rasterix: you might want to consider talking to a lawyer about that |
| 23:32.50 | juniowww | fish-bulb no .. but tomorrow i ll get another creditcard |
| 23:33.02 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 23:33.02 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 23:33.27 | lmadsen | russellb: can I ask you what I need to install to build app_jack? I've install the jack libraries, and I install arts-devel (which seems to give me resample?) but I still don't seem to have the deps for it |
| 23:33.37 | Corydon76-dig | Some lawyers may disagree with that assessment, and in any case, answering a lawsuit is far more expensive than just buying a license |
| 23:33.53 | jblack | I finally got around to trying to change the ring that callers get with a gsm. Background says it can do it, but the I'm getting stuck on the background, and the dial isn't happening until backgrond is done |
| 23:33.57 | rasterix | corydon76: i agree |
| 23:34.06 | rasterix | like i said... id buy a license |
| 23:34.58 | lmadsen | http://www.pastebin.ca/1187973 |
| 23:35.08 | russellb | lmadsen: you probably don't have libresample ... |
| 23:35.15 | russellb | lmadsen: it's in the thirdparty repository |
| 23:35.23 | r0land | hello all! |
| 23:35.27 | voxter | what cool things can you do with app_jack? |
| 23:35.51 | russellb | voxter: connect asterisk channels to jack ports, or hook jack up in the audio path of a channel |
| 23:36.14 | voxter | <- not familiar with Jack, really. I'll go google. |
| 23:36.16 | russellb | so, use an external app to do whatever you would want to do with the audio in asterisk |
| 23:36.19 | russellb | ah ... |
| 23:36.21 | russellb | jackaudio.org |
| 23:36.21 | r0land | anyone got an advice for me about this matter please: sip * sip calls are over lapping... in other words: softphone > asterisk >sipura is overlapping with another softphone's phone call and the two calls gets merged into one with every1 hearing the other... |
| 23:36.29 | juniowww | another question.. insidevox provider have a mobile application .. in application i type .. user pass my phone number and destination number. and the app send by gprs to server and the server make callback |
| 23:37.16 | juniowww | anyone know when i get same app |
| 23:37.35 | bkw_ | write it on a webpage |
| 23:37.43 | bkw_ | and access that via the phone |
| 23:38.14 | juniowww | but is the app not by browser |
| 23:38.26 | bkw_ | os you can do it in the browser easy |
| 23:38.36 | bkw_ | s/os/so/ |
| 23:38.50 | juniowww | by browser is easy.. in a2billing_customer |
| 23:39.36 | voxter | russellb: I was hoping to chat with you actually about a few specifics on IAX. I've always had a scenario in which IAX sounds "bad" and SIP sounds good, and is replicatable, and doesnt make much sense to me. I'm speaking on using IAX to interconnect at Astricon this year and figuring the underlaying cause of it out would be most helpful. |
| 23:40.18 | *** join/#asterisk dawalama (n=dawalama@wsip-68-105-250-226.sd.sd.cox.net) |
| 23:40.26 | russellb | voxter: I'm about to leave for dinner. Feel free to email me directly and we can chat. russell@digium.com |
| 23:40.31 | dawalama | Hello all |
| 23:40.46 | dawalama | I am trying to use java-asterisk |
| 23:40.46 | voxter | Oh, Jack looks cool. Low latency network audio. That could come in handy for QoS observation for remote PBX's, or remote barging/monitoring. |
| 23:40.52 | voxter | russellb: Will do! |
| 23:41.23 | r0land | so any1 has an advice about calls overlapping on asterisk ?! |
| 23:42.17 | EI5GTB | im whirling together plans in my head for a home automation system... is there any way to display a call (say from an external line) on the screens around the house? |
| 23:42.24 | *** join/#asterisk implicit- (n=bayan@unaffiliated/implicit) |
| 23:42.30 | dawalama | I am listening to events triggered from asterisk, LinkEvent .. seems to get triggered when the link is established... what event should I listen to if I want to find out when the receiver answers the phone ? |
| 23:42.43 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 23:42.56 | dawalama | I know this is possible because CDR records the duration of the call |
| 23:43.08 | dawalama | Just wondering if asterisk publishses the event |
| 23:44.00 | rasterix | ei5gtb: there are a lot of different ways that could be done |
| 23:44.24 | rasterix | ei5gtb: you could use the ami to monitor events |
| 23:44.55 | dawalama | What I am trying to do is start monitoring/recording a call. The record works fine when I use LinkEvent, but it also records as soon as the receiver starts ringing. I want it to only record on Answer . |
| 23:45.26 | dawalama | any tips :-) |
| 23:46.18 | EI5GTB | rasterix, aah, very good.. i have nothing inplimented yet, but im planning on making a linux home automation system....and iv been into phone systems for a while...only started reading about asterix this morning... im absoloutly blown away by its flexibility |
| 23:47.14 | rasterix | ei5gtb you can connect to the asterisk manager interface (AMI) via tcp/ip and then configure what asterisk events you want to monitor |
| 23:47.58 | rasterix | to some extent |
| 23:48.17 | EI5GTB | wow, nice |
| 23:48.41 | EI5GTB | would be very easy to integrate some kind of home automation program to read everything spose.. |
| 23:48.43 | EI5GTB | neat |
| 23:49.17 | *** join/#asterisk bbhenry (n=oper@wsip-68-105-250-226.sd.sd.cox.net) |
| 23:49.22 | rasterix | you can also control asterisk through the AMI |
| 23:49.47 | rasterix | so its not just for monitoring events... you could also place a call through the AMI for example |
| 23:49.50 | EI5GTB | you could have a recorded message of ordering pizza then....and i could have a buttin on my touch screen "order pizza" |
| 23:50.05 | EI5GTB | lol, the possibilities are endless |
| 23:50.08 | rasterix | yes lol... you could do that |
| 23:50.31 | EI5GTB | im on hols atm, im itching to get home and knock together an experimental astrix box |
| 23:50.40 | EI5GTB | ]just needa get an ata card and a few ip phones |
| 23:50.55 | rasterix | or soft phones... |
| 23:50.59 | rasterix | for a home system |
| 23:51.07 | EI5GTB | true, yea, never thought of those |
| 23:51.24 | EI5GTB | i have a few places where i would need phisical phones |
| 23:51.32 | juniowww | somebody helpme with callback by mobile?? |
| 23:51.34 | EI5GTB | but i spose i can play about with soft phones for now |
| 23:51.59 | rasterix | yup that would be cheaper than buying a load of ip phones... |
| 23:52.34 | rasterix | x-lite is pretty good as soft phones go |
| 23:52.48 | EI5GTB | ill keep that in mind |
| 23:53.18 | EI5GTB | i would like (eventually) to give relations on the other side of the world phones, that we could dial as an extension.... |
| 23:53.45 | rasterix | ei5gtb: http://www.asteriskguru.com/tutorials/xlite_softphone.html |
| 23:53.46 | EI5GTB | would be cheaper uing ip than pstn (abviously) |
| 23:53.47 | voxter | I am also enjoying Express Talk lately. |
| 23:53.51 | voxter | (for a soft phone) |
| 23:53.55 | juniowww | somebody helpme with callback by mobile?? |
| 23:53.59 | EI5GTB | ooh |
| 23:55.01 | EI5GTB | what kind of cost are you talking for viop to cell phones? |
| 23:55.18 | EI5GTB | im trying to cut down on the cost of calls to mobiles from my house phone... |
| 23:55.33 | EI5GTB | tries usung a gsm unit for a while, it was a pay as you go |
| 23:55.39 | EI5GTB | too much bothr to top up and all |
| 23:56.01 | rasterix | ei5gtb: get asterisk then you can block your kids calls to cell phones XD |
| 23:56.22 | EI5GTB | lul, i have an analog system atm |
| 23:56.40 | EI5GTB | blocking cell calls was...a good idea at the start...untill i needed to make them :P |
| 23:56.51 | juniowww | i want get a app to record callback by gprs |
| 23:56.52 | EI5GTB | it was annoying have to reconfigure the system to make a call |
| 23:56.59 | rasterix | so just block them from THEIR phones |
| 23:57.02 | rasterix | :) |
| 23:57.05 | EI5GTB | mm |
| 23:57.34 | EI5GTB | tempting......theres no real way to do it cheaply? |
| 23:57.45 | rasterix | or log the calls they make... and take it out their pocket money |
| 23:57.47 | rasterix | :) |
| 23:57.50 | EI5GTB | heh |
| 23:58.21 | rasterix | give them a weekly allowance to mobiles.. possibilities are endless |
| 23:59.03 | rasterix | i am not aware of a cheep way to call cell phones |
| 23:59.14 | rasterix | ask the blue people |
| 23:59.22 | rasterix | they are smarter than me :) |
| 23:59.26 | EI5GTB | atm were paying 30 euro ($45) a month for all alocal land line calls, and calls to british land lines |
| 23:59.37 | EI5GTB | i wonder is their a cheaper viop method.. |
| 23:59.48 | rasterix | are you in the uk? |
| 23:59.52 | EI5GTB | i saw a skype package, qw |
| 23:59.59 | EI5GTB | disregard\ |