| 00:00.01 | dennis00 | American cows go shadoozle afaik, but our cows go 'boe'. |
| 00:00.20 | LemensTS | exten => ${DIALPATTERN},1,Dial(${TRUNKTYPE}/${TRUNKNAME}/${DESTNUMBER}) |
| 00:00.20 | LemensTS | can I not use ${DIALPATTERN} variable in this position? |
| 00:00.26 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 00:01.01 | leifmadsen | LemensTS: is it a global variable? |
| 00:01.40 | LemensTS | Its the same as the other 3 variables, its a channel variable passed from an agi script. |
| 00:01.53 | leifmadsen | not sure if you can use a channel variable like that |
| 00:02.12 | leifmadsen | I think it has to be a global variable, but I haven't tried that in a long time |
| 00:02.21 | LemensTS | I figured on dialplan reload, that it would need to be a valid destination not a variable |
| 00:02.31 | *** join/#asterisk sjobeck (n=Adium@63.227.212.111) |
| 00:02.42 | *** part/#asterisk sjobeck (n=Adium@63.227.212.111) |
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| 00:12.59 | exothermc_ | how do you get asterisk to kick back a 302? |
| 00:16.30 | leifmadsen | exothermc_: Transfer() before any audio or Answer() |
| 00:16.37 | leifmadsen | (i.e. first step in the dialplan) |
| 00:17.37 | exothermc_ | leifmadsen: diversion header inserted? |
| 00:17.37 | exothermc_ | \ |
| 00:17.47 | leifmadsen | ya, something like that |
| 00:18.47 | carrar | diversion aren't usually a 302 |
| 00:18.57 | carrar | they are initiated calls |
| 00:20.05 | exothermc_ | carrar: ahh ok |
| 00:22.59 | *** join/#asterisk uluatu (n=uluatu@189.114.176.219.dynamic.adsl.gvt.net.br) |
| 00:23.48 | carrar | exothermc_, why do you want to generate a 302 from Asterisk? |
| 00:24.06 | exothermc_ | carrar: Actually was a bad idea |
| 00:24.13 | carrar | There is a patch out there for SIPRedirect, or it might already be in your system |
| 00:24.37 | exothermc_ | carrar: Since we couldn't do any accounting on the call. We are going to do a new call with diversion header. |
| 00:24.37 | Micc | I've got an aastra 6730i that worked fine on my network but on customer's network it can receive calls, but when making calls I don't see anything until I turn on sip debug and I see Correct auth, but based on stale nonce. Anyone have any idea how I can solve this on asterisk side? Can I make this peer not so picky about nonce? |
| 00:25.27 | carrar | exothermc_, that doesn't make any sense |
| 00:26.08 | Micc | They seem to be registering fine, but I do see we send them a 402 unauthorized. |
| 00:26.14 | exothermc_ | carrar: How would I account for a call if I'm no longer in the signaling path? |
| 00:26.38 | carrar | stay in the path |
| 00:26.55 | exothermc_ | carrar: Stay in the path, and reply with a 302? |
| 00:27.35 | carrar | 302 is a redirect |
| 00:27.40 | carrar | putting you out of the path |
| 00:27.58 | exothermc_ | carrar: I'm not sip expert, but I'm pretty sure those are mutually exclusive. |
| 00:28.59 | dennis00 | I am looking for change the language for Asterisk, I have downloaded the files, does anybody know how to let Asterisk use them? |
| 00:29.51 | jblack | someone was looking for me? |
| 00:30.19 | jblack | xa0z: Here I am |
| 00:30.50 | *** join/#asterisk exothermc_ (n=miles@74.85.89.233) |
| 00:31.06 | exothermc_ | sorry got knocked offline |
| 00:31.07 | p3nguin | (1538.22) -!- xa0z has left #asterisk [] |
| 00:31.13 | exothermc_ | not sure if I missed any insight |
| 00:31.30 | exothermc_ | carrar: I didn't know you could send a 302 and stay in the signaling path. |
| 00:33.15 | *** join/#asterisk xlp (n=secu@mail-out.lpcorp.fr) |
| 00:33.29 | xlp | hello |
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| 00:55.00 | dennis00 | are there any dutch people here? |
| 00:57.20 | p3nguin | I think you're like the third person I have seen. |
| 00:58.43 | p3nguin | (that has said they are Dutch, that is) |
| 01:00.37 | freetown2 | we need an ascii windmill in here :D |
| 01:05.45 | *** join/#asterisk hhkahya (n=hulusika@88.239.45.61) |
| 01:08.46 | *** join/#asterisk chendy (n=chatzill@204.152.211.137) |
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| 01:11.13 | bpgoldsb | Does the Directory application work with realtime voicemail? |
| 01:19.18 | freetown2 | just great. looks like I will have to do a weekly check for updates just to make sure the HQ sysadmin stays on top of security issues. p3nguin you still applying to work at ESF? |
| 01:19.57 | dennis00 | European Science Foundation? |
| 01:20.36 | freetown2 | nope, English Schools Foundation, HK. |
| 01:21.12 | freetown2 | just got a small joke going here from yesterday |
| 01:21.35 | dennis00 | I see. |
| 01:21.35 | freetown2 | or trying to get it going anyway |
| 01:21.40 | dennis00 | Can I join the joke? |
| 01:21.52 | freetown2 | sure. check out www.esf.edu.hk |
| 01:22.12 | dennis00 | I am also pulling a joke with p3nguin, but he is not responding. |
| 01:22.18 | dennis00 | But your joke is better. |
| 01:22.47 | freetown2 | he did say he was applying :D |
| 01:24.05 | freetown2 | if he did get in and push out a certain head - i'd get to rollout * in school...heck we'd probably get to rollout * in all the ESF schools :D |
| 01:27.19 | dennis00 | lol |
| 01:28.57 | dennis00 | http://pastebin.ca/1751806 |
| 01:29.06 | dennis00 | Do you see what goes wrong here? Is this an issue with my provider? |
| 01:29.44 | hardwire | I cannot for the life of me find the document that describes dundi metrics |
| 01:30.08 | hardwire | theres reserved ranges for certain things |
| 01:30.10 | hardwire | can't find it |
| 01:34.40 | *** join/#asterisk coppice (n=chatzill@106.202.17.210.dyn.pacific.net.hk) |
| 01:34.46 | freetown2 | dennis00, dunno...not at all familiar with sip. i see it tried to setup a direct connection between your x-lite and the other side, it got refused, tried to proxy and finally got a busy...no idea |
| 01:35.14 | freetown2 | hi coppice . at home or at work? |
| 01:35.54 | carrar | SIP/2.0 407 Proxy Authentication Required |
| 01:36.18 | *** part/#asterisk beek (n=klinebl@pdpc/supporter/bronze/beek) |
| 01:36.51 | dennis00 | I do not use a proxy. |
| 01:36.53 | dennis00 | And I do authenticate |
| 01:38.32 | freetown2 | i am be wrong but i think asterisk tried to bridge the call when the attempt to setup a direct link failed... |
| 01:39.09 | p3nguin | dennis00: Looking for 0107142866 in phones |
| 01:39.31 | p3nguin | dennis00: Add the exten => 0107142866,...... in the appropriate place. |
| 01:39.44 | dennis00 | You already did that, right? |
| 01:39.46 | p3nguin | Is that for a DID? |
| 01:39.48 | p3nguin | no |
| 01:40.22 | p3nguin | I don't recognize the number, so I don't know what it is meant for. |
| 01:40.35 | p3nguin | I hope it's not an internal phone/device. |
| 01:40.45 | p3nguin | I would hate to have to remember that is your internal extension. |
| 01:41.37 | freetown2 | it's just ten numbers long |
| 01:42.06 | p3nguin | That's about six more than I care to press in for an internal phone. |
| 01:42.51 | freetown2 | uber large org. :D |
| 01:43.08 | p3nguin | for sure |
| 01:43.08 | *** join/#asterisk uluatu (n=uluatu@189.114.176.219.dynamic.adsl.gvt.net.br) |
| 01:43.49 | *** join/#asterisk nsgn (n=nsgn@71.20.112.162) |
| 01:44.20 | nsgn | Digium TDM800p in Asterisk, Firmware 107, only getting half duplex on PSTN calls |
| 01:44.54 | nsgn | a few things i google up tell me that this is a firmware 107 issue but i've never screwed with a firmware update for these cards before. google surprisingly doesn't do much to assist with that. help much appreciated |
| 01:45.28 | nsgn | half duplex in this case meaning the effect that one person talking over the other cuts them off. i can't talk at the same time i'm listening to someone |
| 01:45.38 | nsgn | i must stop talking to hear any audio from their end again |
| 01:47.41 | freetown2 | I like walkie-talkies |
| 01:48.37 | nsgn | precisely. that's the effect. this appears to be a firmware issue with the hardware echocanceler i have. i need advice on getting it to run a newer firmware without great distruction to this system that's in operation |
| 01:48.49 | nsgn | or at least must be at 8am tomorrow |
| 01:49.20 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 01:49.36 | nsgn | is the firmware a part of each build of dahdi? |
| 01:50.14 | freetown2 | p3nguin, may i bother you for the script/tool that you used to do a routine check on www.esf.edu.hk? I going to blast the HQ sysadmin so I thought I might as well flame him for that box of his/HQ's. |
| 01:50.45 | p3nguin | freetown2: Sure: |
| 01:50.48 | p3nguin | freetown2: |
| 01:50.51 | p3nguin | dammit |
| 01:50.57 | bpgoldsb | Hmm, it appears the 'x' option for for the Record application is no longer working in 1.6.2 |
| 01:51.01 | p3nguin | freetown2: echo -e "HEAD / HTTP/1.0\n\n" | nc www.esf.edu.hk 80 |
| 01:51.44 | p3nguin | freetown2: Both apache and php are out-of-date. |
| 01:51.46 | freetown2 | gah. So snort would have done it I guess huh? |
| 01:52.05 | freetown2 | i'd just have to feed it that box's ip eh? |
| 01:52.09 | p3nguin | snort? That's an IDS. |
| 01:52.32 | carrar | everything is outof date |
| 01:52.37 | freetown2 | oh...it had to have a local client eh...been a while. nevermind :P |
| 01:52.40 | carrar | and close up ssh |
| 01:52.45 | carrar | and upgrade |
| 01:52.58 | p3nguin | freetown2: Seriously, grab a command line, type in "echo -e "HEAD / HTTP/1.0\n\n" | nc www.esf.edu.hk 80" and press enter. |
| 01:53.03 | freetown2 | close up ssh? ha! I had a conversation with him on that score. |
| 01:53.13 | carrar | well at least upgrade it then |
| 01:53.16 | *** join/#asterisk lordmortis (n=lordmort@203-206-67-161.dyn.iinet.net.au) |
| 01:53.24 | freetown2 | i said, either use ssh keys or I will block ssh to the moodle server they are putting in the school |
| 01:53.44 | freetown2 | since i am first line support, no way i am letting them do what they like |
| 01:54.07 | freetown2 | not surprised that ssh is open on their box. |
| 01:54.30 | freetown2 | p3nguin, done so. I see the zend version and so forth. thanks a lot. |
| 01:54.36 | p3nguin | I don't mind having a box facing the internet with SSH available, but make sure you have good passwords or NO passwords with keys only. |
| 01:54.43 | freetown2 | has zero exp in looking about apache. |
| 01:54.46 | nsgn | ..anyone on updating firmware for a digium card? i've never had to deal with this |
| 01:55.08 | p3nguin | Ideally, you'll have to use a VPN to reach the sshd. |
| 01:55.13 | freetown2 | i worked as a mta admin before but nothing on the webservers was my responsibility except the mail queue |
| 01:56.09 | freetown2 | p3nguin, agreed. which is why i closed ssh to all except verified HQ ips on the moodle server - thebuzz.bradbury.edu.hk |
| 01:56.20 | p3nguin | ACLs? Good job. |
| 01:56.28 | *** join/#asterisk tgunr (n=tgunr@cust-66-249-166-12.static.o1.com) |
| 01:56.41 | p3nguin | Most people can't grasp how to create a decent ACL at the firewall. |
| 01:57.02 | carrar | allow any any |
| 01:57.22 | freetown2 | yeah, especially if they had to use crap that create one big frigging RH-Firewall-1 chain. |
| 01:57.58 | p3nguin | I wouldn't expect you to use an RHEL or CentOS box as your firewall. |
| 01:58.09 | p3nguin | At least use a cheap ASA or PIX. |
| 01:58.13 | freetown2 | glad i got to learn something while I was at Outblaze ltd. (mail business now part of IBM - lotuslive) |
| 01:58.22 | carrar | go juniper |
| 01:58.56 | freetown2 | school is on a tight budget... |
| 01:58.58 | p3nguin | I'm not saying RHEL and CentOS are bad. I personally use a Linux box with iptables as a firewall at more than one location. |
| 01:59.30 | freetown2 | and i want control...i am not negotiating with ripoff ISP known as PCCW Netvigator for acccess to their router |
| 01:59.35 | carrar | build a firewall using openbsd |
| 01:59.50 | p3nguin | pf is awesome. |
| 01:59.58 | freetown2 | i have not yet got round to makign a briding firewall with openbsd |
| 02:00.11 | p3nguin | I use pf on my primary server which is internet-facing. |
| 02:00.13 | freetown2 | good for you |
| 02:00.27 | p3nguin | erm |
| 02:00.33 | freetown2 | i had a floppy based openbsd firewall on an old 486 some time back :D |
| 02:01.16 | freetown2 | right now, it is just one box...it can defend itself for the moment. |
| 02:02.37 | nsgn | pf is pretty nice |
| 02:02.45 | nsgn | capable, but clumsy in interface at times |
| 02:03.12 | nsgn | paying for echo cancelation to find it's screwed up on the tdm800 is really sucky, on the other hand. |
| 02:03.12 | nsgn | http://forums.digium.com/viewtopic.php?p=131418&sid=cc06cb1c88d689f081e70a1581ddbf9f |
| 02:05.20 | freetown2 | pf clumsy? most people lose their minds reading the iptables man page... |
| 02:05.48 | freetown2 | or /etc/sysconfig/iptables :D |
| 02:06.08 | *** join/#asterisk rossand (n=aross@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 02:08.11 | *** join/#asterisk scardinal (n=supreme@90.184.100.119) |
| 02:11.02 | freetown2 | hmm...php 5.2 ain't RH provided i don't think... |
| 02:12.22 | *** join/#asterisk Tech_Travis (n=Administ@cpe-76-168-191-127.socal.res.rr.com) |
| 02:14.06 | nsgn | last call for any help with my tdm800 :( |
| 02:14.53 | freetown2 | sorry, not had a chance to use digium cards...and probably won't given the cheaper option available just across the border... |
| 02:15.21 | nsgn | this is the first time i've had a bad experience with them |
| 02:15.36 | freetown2 | can't you get support from them? |
| 02:15.56 | nsgn | not on as short a notice as i need now |
| 02:16.41 | freetown2 | downgrade firmware for the moment? |
| 02:17.12 | nsgn | i asked about how to upgrade/downgrade firmware in here cause i have no clue how on these things and nobody answered |
| 02:18.09 | freetown2 | maybe it is just bad timing |
| 02:18.25 | nsgn | yeah. i bypassed the hardware EC and i'm ok for now |
| 02:18.52 | nsgn | i'm just ticked i paid for that EC module that actually does sound really nice when it works |
| 02:19.01 | nsgn | but i have to shut it off cause of some compatibility issue |
| 02:19.32 | nsgn | ah well, still have another client to drive to tonight so i've got this one doing EC in software and all sounds ok. thanks and goodnight! |
| 02:23.07 | *** join/#asterisk darkskiez__ (n=dz@62-50-207-156.client.stsn.net) |
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| 02:37.04 | freetown2 | Oh goodie. I get to blast another vendor. HQ don't host www.esf.edu.hk themselves |
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| 02:45.38 | dlynes | Would there be a reason why a call from a sip endpoint that talks passthrough to an asterisk box, and then gets handed off across a common sip connection to another asterisk box get a username mismatch on the second asterisk box, when the second asterisk box tries to match the target that it's calling? |
| 02:46.14 | dlynes | When every other sip endpoint on the same asterisk box is able to pass calls through the same sip connection to the secondary asterisk server just fine? |
| 02:46.40 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 02:47.17 | dlynes | Now when it was iax2 connecting the two boxes, this username mismatch error never happened |
| 02:47.30 | dlynes | We switched it over to SIP recently because of call quality issues with iax2 |
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| 03:25.22 | drfreeze | Anyone know how I can log the amount of time a call is on hold? |
| 03:25.45 | drfreeze | We are using polycom phones. |
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| 04:16.22 | LemensTS | Do channel variables take much resources? |
| 04:20.46 | *** join/#asterisk xuser (n=xuser@unaffiliated/xuser) |
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| 04:26.35 | ChannelZ | memory |
| 04:27.49 | hardwire | Leak |
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| 04:31.06 | *** join/#asterisk audified (n=aurora12@124.155.196.226) |
| 04:31.31 | audified | hi guys i need some help |
| 04:31.32 | dlynes | Is it just me, or is iax2 not terribly reliable? |
| 04:31.51 | dlynes | audified, instead of saying you need help, just state the problem |
| 04:31.55 | audified | Rejected connect attempt from , request '@default' does not exist |
| 04:32.06 | audified | this happens when a user dials out |
| 04:32.09 | audified | using iax2 |
| 04:32.19 | dlynes | audified, can you paste the complete error message? |
| 04:32.46 | audified | ok |
| 04:33.06 | dlynes | audified, if it takes more than three lines, please use pastebin |
| 04:33.08 | dlynes | ~pb |
| 04:33.08 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 04:33.41 | audified | Rejected connect attempt from <ip address>, request '<number that the user dialed@default' does not exist |
| 04:33.47 | audified | this is the message |
| 04:33.59 | audified | the user is using an ata.. |
| 04:34.03 | ChannelZ | It says what it does, it does what it says |
| 04:34.20 | dlynes | audified, you have unauthenticated traffic trying to send you calls |
| 04:34.37 | audified | ok but i've registered the ata with my server |
| 04:34.46 | dlynes | audified, no, you haven't |
| 04:35.06 | dlynes | audified, or if you have, you don't the user defined correctly |
| 04:35.15 | audified | that is weird because on the ata config page, it alr shows registered |
| 04:35.26 | dlynes | audified, then you don't have the user defined correctly |
| 04:35.49 | dlynes | audified, this is an iaxy? |
| 04:36.05 | audified | ok, im used the web gui to add the iax2 account for the user |
| 04:36.10 | audified | as in i used* |
| 04:36.19 | dlynes | web gui? |
| 04:36.23 | dlynes | what web gui? |
| 04:36.49 | dlynes | on the ata? and as i asked, is the ata an iaxy? |
| 04:37.09 | audified | yes |
| 04:37.18 | dlynes | yes to which question? |
| 04:37.23 | dlynes | i'm not a mind reader |
| 04:37.28 | audified | ata to an iaxy |
| 04:37.50 | dlynes | ok, and the web gui you're talking about is on the iaxy device? |
| 04:38.00 | audified | yup |
| 04:38.02 | dlynes | ok |
| 04:38.04 | audified | correct |
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| 04:42.13 | audified | [Jan 15 12:41:12] NOTICE[3137]: chan_iax2.c:8317 socket_process: Rejected connect attempt from <ip>, request '1234567@default' does not exist |
| 04:42.20 | audified | this is the full one |
| 04:42.24 | *** join/#asterisk chilicuil (n=chilicui@unaffiliated/chilicuil) |
| 04:42.27 | audified | except tt i edited the numbers |
| 04:43.12 | dlynes | that's more like it, although I figured something out similar to that previously...but it's always better to do a complete paste like the last paste you did |
| 04:43.27 | audified | sorry |
| 04:43.43 | dlynes | audified, can you pastebin the iax2.conf file as well please? (entire file) |
| 04:43.55 | dlynes | audified, scrub the passwords before you pastebin it |
| 04:44.47 | dlynes | audified, and for the matching peer if you're paranoid about the user/peer names, replace the matching one with '<ip>', so that i know it's the same as the one from the log message above |
| 04:45.09 | dlynes | erm matching user i mean |
| 04:45.48 | audified | ok hold on |
| 04:47.39 | audified | http://pastebin.com/mdd5666e |
| 04:47.44 | audified | my iax.conf is just like tt |
| 04:47.55 | audified | those tt are not commented out |
| 04:48.22 | voipmonk | iax has no secret, eh? |
| 04:48.54 | voipmonk | and peers dont need contexts |
| 04:49.01 | voipmonk | only friends or users |
| 04:49.14 | audified | no secret |
| 04:49.18 | voipmonk | and the top half isnt labelled.. |
| 04:49.18 | hardwire | fusers. |
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| 04:56.33 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.0 (2009/12/18), Asterisk 1.6.1.12 (2009/12/18), 1.6.0.20 (2009/12/18), 1.4.28 (2009/12/18), *-Addons 1.6.2.0 (2009/12/18), 1.6.1.2 (2009/12/02), 1.6.0.4 (2009/12/02), 1.4.10 (2009/12/02), dahdi-linux 2.2.0.2 (2009/07/23), dahdi-tools 2.2.0 (2009/06/24), Libpri 1.4.10.2 (2009/10/20) -=- Related channels: #asterisknow #switchvox #asterisk-bugs #asterisk-gui |
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| 06:13.48 | soman | Hi, Good morning all. |
| 06:15.24 | soman | I am using asterisk 1.6 with TE121 PRI card, and facing the problem "PRI Error on span 0: We think we're the CPE, but they think they're the CPE too" |
| 06:15.47 | soman | I was told to consult the telco and remove the loopback on the interface. |
| 06:16.38 | soman | But, Actually the same PRI connection is working fine with the other server, which is using TE110P dual span card with asterisk 1.2 |
| 06:17.33 | soman | Im getting that error only on new server (asterisk 1.6 + TE121 card). Can any one suggest me what else could be the problem |
| 06:20.07 | soman | ChannelZ: TSM: [TK]D-fender: tzafrir_laptop or can anyone help me how to sort out this "warning". |
| 06:20.48 | tzafrir_laptop | so, there's no loop? |
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| 06:20.59 | tzafrir_laptop | The driver isn't somehow in some loop mode? |
| 06:22.25 | ChannelZ | sorry I know next to nothing about PRI |
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| 06:37.59 | soman | tzafrir_laptop: how to check whether the driver is in loop mode. I tried re installing dahdi-linux-complete.. and started asterisk.. still the same issue |
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| 06:40.59 | soman | any ideas,,,, |
| 06:48.51 | ChannelZ | does dahdi_scan perhaps reveal anything out of place? |
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| 07:00.27 | soman | tzafrir_laptop: ChannelZ: I am trying to reinstall everything from scratch. while installing asterisk1.6, in ./configure outpur I see this line "checking for mISDN_open in -lmISDN... no" . Does this mISDN library has got something to do with that error? |
| 07:00.57 | tzafrir_laptop | no |
| 07:01.12 | soman | ok |
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| 07:06.35 | Jenna | hiyall, any asterisk + sipXecs guru around. |
| 07:06.42 | Jenna | ? i.e. |
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| 07:13.33 | sidd | hi, i have noticed that sip show peers shows a lot of unconnected peers. how do i remove them without deleting them from the file? |
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| 07:16.43 | p3nguin | sidd: Comment them out instead of deleting them. Then run "sip reload" from the CLI. |
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| 07:25.33 | soman | tzafrir_laptop: ChannelZ: I have re-installed everything.. but still the same error.. here is the output of dahdi_scan http://pastebin.com/m3f534dc4 |
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| 07:35.45 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.0 (2009/12/18), Asterisk 1.6.1.12 (2009/12/18), 1.6.0.20 (2009/12/18), 1.4.28 (2009/12/18), *-Addons 1.6.2.0 (2009/12/18), 1.6.1.2 (2009/12/02), 1.6.0.4 (2009/12/02), 1.4.10 (2009/12/02), dahdi-linux 2.2.0.2 (2009/07/23), dahdi-tools 2.2.0 (2009/06/24), Libpri 1.4.10.2 (2009/10/20) -=- Related channels: #asterisknow #switchvox #asterisk-bugs #asterisk-gui |
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| 07:37.55 | donatas | Have anyone seen such warning: [Jan 15 09:27:55] NOTICE[2075]: chan_sip.c:5506 process_sdp: No compatible codecs, not accepting this offer! ? |
| 07:38.13 | Jenna | soman, what is the error ? |
| 07:38.39 | soman | Jenna: PRI Error on span 0: We think we're the CPE, but they think they're the CPE too. |
| 07:39.04 | sidd | p3nguin: what is happening is that only 2-3 clients are able to register on my sip server. what i found that if 4-5 users share the same internet connection behind a natted adsl modem, they have the same ip and the server says all are alive while only one may be at that time |
| 07:39.25 | sidd | am wondering if this could cause the server to deny connections. |
| 07:39.29 | p3nguin | sidd: Can you give them different ports? |
| 07:39.49 | sidd | p3nguin: you mean fixed ports? |
| 07:40.27 | p3nguin | I mean add things like port=5061 port=5062 port=5063 in the sip peer definitions. |
| 07:41.05 | sidd | p3nguin: ok, will try that and see. can the ports be reused for other users who will not share the ip of these customers? |
| 07:41.28 | p3nguin | I think that will be okay. |
| 07:41.37 | sidd | p3nguin: thanks, will try that and see. |
| 07:43.10 | kaldemar | soman: you need to configure one end as net and the other as cpe. so, in chan_dahdi.conf, set parameter signalling as either pri_net or pri_cpe. |
| 07:44.17 | Jenna | soman, u need to sort this one with ur telco. its seems ur configurations dont sync |
| 07:44.41 | soman | kaldemar: I have set the signalling to pri_cpe... |
| 07:45.49 | soman | Jenna, I have another server with asterisk 1.2 and TE110P card.. and there i am not getting any errors... the same configurations i am using for the new server too |
| 07:47.25 | kaldemar | soman: it can't be pri_cpe in BOTH ends |
| 07:48.06 | soman | kaldemar: I tried with pri_net too.. then i am getting "we think we are the network, but they think they are the networrk too" |
| 07:48.25 | kaldemar | soman: what are you connecting to? |
| 07:48.47 | kaldemar | are you using a loopback cable on the interface? |
| 07:49.52 | donatas | how to solve problem between asterisk and client behind NAT ? |
| 07:49.54 | soman | Kaldemar: I am connecting to the PRI cable from the telco operator |
| 07:51.00 | kaldemar | soman: doesn't sound like it. show your /etc/dahdi/system.conf and /etc/asterisk/chan_dahdi.conf and tell which port is connected to what. |
| 07:52.15 | soman | kaldemar: here they are http://pastebin.com/m5b37cd5e |
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| 07:54.59 | ChannelZ | ~sipnat |
| 07:55.00 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 07:55.34 | sidd | p3nguin: one more thing, since most of my customers are on adsl, they keep getting new IPs all the time. the server remembers these ip addresses and shows them on the asterisk console. how do i clear this list? |
| 07:57.07 | p3nguin | no clue |
| 07:57.43 | p3nguin | If the clients still get phone calls on the new IP addresses, it sounds like a non-issue. |
| 07:57.53 | soman | kaldemar: any ideas |
| 07:59.38 | sidd | p3nguin: as of now i have a problem of more than 2-3 clients not connecting. |
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| 07:59.53 | kaldemar | soman: that really doesn't make sense unless there's a loopback cable in the port or the driver module is loaded with loopback enabled. |
| 08:00.07 | p3nguin | sidd: Because of the new IP addresses? |
| 08:00.46 | soman | kaldemar: how to check whether the driver module is loaded with the loopback, and how to disable that |
| 08:00.49 | sidd | p3nguin: am not sure. am looking at all avenues and because of this list am unable to see who is registered and who is not. |
| 08:00.58 | kaldemar | soman: iirc, there's a loopback parameter for the module, you can check that with "modinfo wcte12xp". |
| 08:00.58 | sidd | lot of ip addresses |
| 08:00.58 | p3nguin | sidd: If it is because of the changing IP addresses, reduce your timeout value so that the client will be told to "check in" with the server more often, updating the IP address sooner. |
| 08:01.11 | sidd | p3nguin: ok, will try that too. |
| 08:01.24 | kaldemar | soman: the module doesn't get the loopback parameter by default, so if you did a clean install, that shouldn't be it. |
| 08:03.06 | soman | kaldemar: here is the output of "modinfo wcte12xp" http://pastebin.com/m3efa6ffc |
| 08:04.31 | sidd | p3nguin: sorry, which timeout value should i change? |
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| 08:05.25 | p3nguin | sidd: maxexpiry, minexpiry, defaultexpiry |
| 08:05.34 | sidd | ok. |
| 08:09.13 | soman | kaldemar: I did a clean install of the dahdi and libpri.... and still its the same problem |
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| 08:16.13 | ChannelZ | splat |
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| 08:16.54 | benngard | split |
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| 08:58.45 | soman | Kaldemar: How can i disable the loopback now.. |
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| 09:11.27 | sidd | p3nguin: i am playing with the maxexpiry timeout. what could be a good value? putting 180 logs out the client often |
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| 09:22.18 | donatas | i have Dlink DPH-120S voip phone and it doesn't work with DTMF |
| 09:22.26 | donatas | maybe anyone have such problem? |
| 09:22.39 | jamicque | Hi can anyone please help me confgiuring MWI with Lisnkys SPA phones? I'm running an asterisk 1.6.1.12 and I seem to have a problem in acomplishing it. |
| 09:24.04 | soman | Kaldemar: Now I have configured my TE121 card as T1. and now i am not getting that errros....but, dahdi_scan shows that "alarms=RED?REC". any guess on what could be wrong here |
| 09:25.02 | sidd | i am getting unspecified for clients that are connected. what can i do? |
| 09:25.24 | sidd | it happens every few minutes. on restarting i get them back as ok. |
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| 09:26.34 | soman | tzafrir_laptop: any ideas... |
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| 09:32.04 | donatas | who can say me, why if i place a call with voip phone i got this: http://p.defau.lt/?o6mjztL59Ksia7cdP1WMiA , but if i call with X-Lite i got Ringing and not Called.. |
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| 09:36.02 | sidd | what is 503 error sent by asterisk to my client? |
| 09:41.01 | fenrus | Service unavailable" |
| 09:41.01 | fenrus | sidd, http://www.voip-info.org/wiki/view/SIP+response+codes |
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| 09:42.07 | sidd | fenrus: hi, my server keeps dropping clients. and every second i see the client requesting and the server denying with code 503. any ideas? the client drops after 2 minutes |
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| 09:43.51 | fenrus | as i said, the 503 message means "Service unavailable" |
| 09:43.51 | fenrus | why this happens i do not know. |
| 09:43.51 | fenrus | it drops all the clients, or just some specific ? |
| 09:44.45 | sidd | fenrus: right now, there is only one. it connects, stays alive for 2 minutes and then sip show peers says unspecified and the client becomes UNKNOWN |
| 09:45.37 | fenrus | what kind if phone is this ? |
| 09:45.41 | fenrus | is it configured to reregister |
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| 10:11.35 | E-bola | Anybody here uses any type of TAPI privder with asterisk? Like AstTAPI or SIPTAPI ? |
| 10:13.21 | Jenna | is there a way to exit the shell of asterisk without shutting it down ? |
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| 10:15.40 | E-bola | Jenna: umm type quit? |
| 10:17.31 | Jenna | E-bola, obviously you haven't used quit on the asterisk shell before. |
| 10:18.06 | garymc | Yo! Anyone know if i can make asterisk play my telco error messages instead of hearing the server error message. eg when i dial an incorrect number? cos it is starting to do my head in. I get the same message for all errors. I want to hear my telco messages as they are more understandable |
| 10:18.50 | E-bola | Jenna: No comments, i guess you should read the manual |
| 10:19.01 | Gido-E | Jenna, it is quit |
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| 10:19.51 | Jenna | I get this on typing "quit "at asterisk cli. No such command 'quit' (type 'help quit' for other possible commands) |
| 10:20.04 | sidd | fenrus: yes, it is configured to reregister. infact, it is a client i have made. what may be happening is that when the server sends a keep alive, the client responds. then the client sends registration request but the server responds with a 503 |
| 10:20.16 | garymc | control c helps quit the cli |
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| 10:20.37 | sidd | is there somewhere i can see the entire client server interaction the way it is supposed to be? |
| 10:20.54 | Jenna | garymc, it also kill the asterisk deamon |
| 10:21.01 | sidd | who sends what first? i see the server trying to see if all the clients are connected |
| 10:21.11 | sidd | it does this every 2 seconds |
| 10:21.17 | sidd | is that normal? |
| 10:21.40 | garymc | yes |
| 10:21.48 | garymc | jenna what are you trying to quit |
| 10:22.10 | garymc | sidd are you running sip debug if so yes |
| 10:22.21 | sidd | garymc: yes. ok. |
| 10:23.24 | sidd | my client disconnects after 2 minutes. what should i be looking for? |
| 10:23.25 | garymc | to stop that happening sidd you need to stop the sip debug. I think you type "sip debug stop" or something along those lines |
| 10:23.33 | sidd | sip debug off |
| 10:23.47 | garymc | yep thats it |
| 10:23.59 | garymc | im not sure what you should be looking for |
| 10:24.33 | sidd | but when i do a tcpdump, i still see a lot of packets going to all the clients, whether they are connected or not |
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| 10:33.45 | angryuser_ | When multiple sip accounts dialed, i.e Dial(Technology/ext Technology/ext) how the logic goes if i have 3 sip Devices (FXO gateways) and i need to dial the only free one ? i can set call-limit to 1 , the question is wil i dial all free FXO's or the only one ? and if not how to effectively choose only one sip account ? I know that i can do it with group, and group count, but is there any faster way ? |
| 10:34.58 | angryuser_ | maybe unclear: i ahve 3 sip accounts which are 3 FXO's in reality, i need to choose the free one, or get Busy if all 3 are used, thanks |
| 10:35.43 | angryuser_ | Set Group, and then use Group_count can do it , but maybe there is some thing faster |
| 10:35.56 | Gido-E | angryuser_ i would advice that. |
| 10:36.12 | Gido-E | It is technolegy independent. |
| 10:36.17 | *** part/#asterisk Tech_Travis (n=Administ@cpe-76-168-191-127.socal.res.rr.com) |
| 10:37.26 | angryuser_ | But when i dial multiple devices (sip) at the only one get dialed or all of them get dialed and the first one to answer get conencted ? |
| 10:37.54 | angryuser_ | But when i dial multiple devices (sip) the only one get dialed or all of them get dialed and the first one to answer get conencted ? |
| 10:38.38 | angryuser_ | there is also a callgroup |
| 10:38.41 | angryuser_ | hmm |
| 10:39.58 | angryuser_ | hmm, the group count is the only way |
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| 11:38.50 | TommyBotten | Using asterisk 1.6.2.0 and spandsp 0.0.6pre12, I always get the "transmit: Transmission error"-message at the end of a fax transmission. The fax goes through fine, but still this message triggers a hangup. |
| 11:39.04 | TommyBotten | I am not using T.38 |
| 11:39.53 | donatas | how to setup asterisk, that it would transfer a call to random user (not bussy) ? |
| 11:40.25 | TommyBotten | donatas: Using queues is probably what you are looking for |
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| 11:41.17 | donatas | thanks |
| 11:42.43 | *** join/#asterisk ChanServ (ChanServ@services.) |
| 11:42.43 | *** mode/#asterisk [+o ChanServ] by irc.freenode.net |
| 11:43.04 | TommyBotten | donatas: Leif Madsen wrote a comprehensive guide on queues. It's interesting reading: https://issues.asterisk.org/file_download.php?file_id=24471&type=bug |
| 11:43.10 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-218-155-147.cablep.bezeqint.net) |
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| 12:10.40 | donatas | hmz, i set callcounter=yes, but i don't see in core show hints that it is being InUse |
| 12:10.45 | donatas | only Idle |
| 12:13.37 | *** join/#asterisk uluatu (n=uluatu@189.114.176.219.dynamic.adsl.gvt.net.br) |
| 12:16.55 | *** part/#asterisk puzzled_ (n=foobar@puzzled.xs4all.nl) |
| 12:21.12 | Akiraa | Is there an inherent advantage to the British (BT) quirky telephony socket design? |
| 12:23.23 | *** join/#asterisk infobot (i=ibot@rikers.org) |
| 12:23.23 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.0 (2009/12/18), Asterisk 1.6.1.12 (2009/12/18), 1.6.0.20 (2009/12/18), 1.4.28 (2009/12/18), *-Addons 1.6.2.0 (2009/12/18), 1.6.1.2 (2009/12/02), 1.6.0.4 (2009/12/02), 1.4.10 (2009/12/02), dahdi-linux 2.2.0.2 (2009/07/23), dahdi-tools 2.2.0 (2009/06/24), Libpri 1.4.10.2 (2009/10/20) -=- Related channels: #asterisknow #switchvox #asterisk-bugs #asterisk-gui |
| 12:30.50 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 12:34.56 | TommyBotten | donatas: Have you set it up to subscribe to a context where the hints are defined? |
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| 13:05.27 | *** join/#asterisk Kchehab (n=kchehab@212.98.141.202) |
| 13:07.02 | Kchehab | <PROTECTED> |
| 13:09.17 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:09.17 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 13:14.47 | *** join/#asterisk [TK]D-Fender (n=chatzill@216.191.106.163) |
| 13:16.30 | Tim_Toady | Kchehab as cha_sip.c says: Implementation of RFC 3261 - without S/MIME, and experimental TCP and TLS support |
| 13:19.55 | *** join/#asterisk HenrikJott (n=info@d83-183-134-141.cust.tele2.se) |
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| 13:25.40 | donatas | TommyBotten: what do you mean subscribe ? I have set exten => 111,hint,SIP/111 |
| 13:25.40 | *** join/#asterisk hluesea (n=hulusika@88.247.127.66) |
| 13:25.42 | donatas | only this |
| 13:30.33 | HenrikJott | Hi all! when i execute DeadAGI in my dialplan i get this output: http://www.pastebin.org/76706 |
| 13:31.37 | HenrikJott | It seems like it executes the php-script but then tries to find it as an extension for some reason... the php-file contains almost nothing and just creates another file for test purposes. |
| 13:32.06 | *** join/#asterisk voipmonk (n=shido6@67.204.37.228) |
| 13:34.46 | *** join/#asterisk iCEBrkr (n=icebrkr@72.251.206.106) |
| 13:36.19 | [TK]D-Fender | HenrikJott: Show us that the file is in the right place and the script itself |
| 13:38.21 | jamicque | Can anyone help me with MWI in Asterisk? Phone subscribes to recieve a notify when new voicemail is aviable. However nothing happens and on my SPA phone the red inidicator lights all the time signaling that there is a new voicemail, even if there isn't ant |
| 13:42.29 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 13:42.34 | voipmonk | jamicque: this spa - is it subscribed to only one system? |
| 13:42.40 | jamicque | yes |
| 13:43.06 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 13:44.12 | jamicque | asterisk don't seem to send any information. I use 1.6.1.12 |
| 13:45.25 | *** join/#asterisk etfonhomey (n=etfonhom@ip-64-32-192-35.iad.megapath.net) |
| 13:52.29 | voipmonk | how many line appearances do you have on your device? |
| 13:57.42 | jamicque | 6 (spa 962), but the main new voicmailindicator igihts up. It's not a linie indicator. It's a big led ion top of device |
| 13:59.16 | *** join/#asterisk RobH_ (n=robh@cpe-173-169-30-118.tampabay.res.rr.com) |
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| 14:09.36 | HenrikJott | [TK]D-fender: I´m sorry i solved it! it was a windows/unix-error. The file was in windows-format (different line breaks) and * didn´t like that =) thanks anyway! |
| 14:09.42 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:09.42 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 14:12.00 | *** part/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 14:12.03 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 14:17.57 | angryuser_ | Tryed to use Android with Sipdroid over openvpn client, works! |
| 14:18.00 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 14:18.26 | *** join/#asterisk ChkDigit (n=mike@static24-72-71-175.regina.accesscomm.ca) |
| 14:19.11 | angryuser_ | Anyone using it ? |
| 14:19.52 | *** join/#asterisk smooth_penguin (n=smoove@59.95.27.150) |
| 14:20.40 | leifmadsen | angryuser_: sounds cool :) I don't have an android phone, using Nokia currently |
| 14:21.43 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:22.10 | creativx | leifmadsen: n900? |
| 14:22.19 | leifmadsen | creativx: heh, E71 :) |
| 14:22.24 | leifmadsen | I'm still not sold on the N900 yet |
| 14:22.32 | leifmadsen | I think it's promising, but I'm waiting for 2nd gen |
| 14:22.39 | creativx | mkay |
| 14:22.42 | leifmadsen | (same with Nexus One though too) |
| 14:22.48 | creativx | i was tempted to try sip on mine |
| 14:23.04 | leifmadsen | the E series has been around long enough that they've got it right. The N900 looks too big for me in its current state |
| 14:23.09 | leifmadsen | SIP works great on the E71 |
| 14:25.19 | angryuser_ | i prefer motorola milestone (eur) / Droid |
| 14:26.06 | creativx | well after the maemo update yesterday things got a wee bit better on the n900 |
| 14:26.17 | angryuser_ | leifmadsen, i got sip working with fring, other clients were somehow not friendly to use the every day |
| 14:26.25 | *** join/#asterisk TheDavidFactor (n=chatzill@fw1.safedataisp.net) |
| 14:26.37 | angryuser_ | leifmadsen, i mean with my old nokia n95 |
| 14:27.03 | leifmadsen | angryuser_: ya, I've gotten it to work with both fring and with the built-in SIP client. The built-in is certainly tricky (not very intuitive) but once you know what it wants, then it works great. |
| 14:28.01 | angryuser_ | leifmadsen, i got it working with both, but when you change hot spot frequently, you need to define them again and again, and to mod profile every time |
| 14:28.01 | *** join/#asterisk Victor_Yure_ (n=victor@unaffiliated/victoryure/x-837844) |
| 14:29.02 | angryuser_ | leifmadsen, n900 is certanly geeks phone, i am not ready to run torrents on my phone yet, or else |
| 14:29.18 | angryuser_ | leifmadsen, and it is really big |
| 14:29.20 | leifmadsen | angryuser_: ah ya, that's true. It works great if you're only going through a subset of hot-spots and you can define them all, because then it switches automatically so effortlessly, but if you're going between a lot of unknown hot-spots, it can be annoying |
| 14:29.28 | creativx | transmission actually worked very well on the n900 |
| 14:30.08 | *** join/#asterisk RobH (n=robh@rob.tech.wikimedia.org) |
| 14:30.56 | *** join/#asterisk shamelessn00b (n=chatzill@58-65-172-114.nayatel.pk) |
| 14:31.37 | shamelessn00b | hi guys, I just installed asterisk using apt-get, but instead of following the dialplan in extensions.conf its following from the extensions.ael, how do I change it |
| 14:31.38 | shamelessn00b | ?? |
| 14:31.50 | *** join/#asterisk muiro (n=muiro@unaffiliated/muiro) |
| 14:32.21 | leifmadsen | shamelessn00b: it's using both actually -- disable pbx_ael.so in modules.conf |
| 14:32.25 | *** join/#asterisk hluesea (n=hulusika@88.247.127.66) |
| 14:32.32 | shamelessn00b | ok |
| 14:32.36 | shamelessn00b | thanks |
| 14:32.38 | [TK]D-Fender | shamelessn00b: go change extensions.ael . Also the only reason it'd use that is because you made something point to one of its contexts |
| 14:33.50 | shamelessn00b | leifmadsen: my modules.conf file doesnt contain any entry like pbx_ael.so |
| 14:34.05 | leifmadsen | shamelessn00b: that's because you have to add it |
| 14:34.17 | [TK]D-Fender | shamelessn00b: noload => pbx_ael.so |
| 14:34.20 | leifmadsen | shamelessn00b: notice the other noload lines -- use that as a reference |
| 14:34.23 | shamelessn00b | ok |
| 14:34.24 | *** join/#asterisk lordmortis (n=lordmort@203-206-86-190.dyn.iinet.net.au) |
| 14:34.24 | leifmadsen | [TK]D-Fender: don't just give him the answer :) |
| 14:34.27 | shamelessn00b | :) |
| 14:34.34 | [TK]D-Fender | leifmadsen: :p |
| 14:34.52 | leifmadsen | I don't want to know your name! I just want... |
| 14:34.53 | [TK]D-Fender | leifmadsen: It's Role-Reversal Fridays! |
| 14:34.57 | [TK]D-Fender | leifmadsen: ! ! ! |
| 14:35.12 | leifmadsen | I don't want relationship! I just want... |
| 14:35.26 | leifmadsen | I don't want to meet your mom! I just want... |
| 14:35.30 | shamelessn00b | lol |
| 14:36.18 | lordmortis | anyone have a ubuntu upstart script? |
| 14:36.20 | lordmortis | (for asterisk) |
| 14:37.15 | angryuser_ | lordmortis, use debian's |
| 14:37.46 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
| 14:37.46 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 14:39.00 | *** join/#asterisk highvoltz (i=rogers@bling.bling.org) |
| 14:39.01 | lordmortis | it doesn't run safe_asterisk though does it? |
| 14:39.04 | lordmortis | or astcanary? |
| 14:40.05 | highvoltz | Setting up a new asterisk box w/freepbx behind a firewall. In/Out trunk is a SIP. opened 5060 tcp/udp, and 10000-20000 udp. I can call out, and call in and make a connection, but cannot hear any talking. What might be the problem? |
| 14:40.50 | lordmortis | highvoltz: is your firewall forwarding 10000-20000 to your freepbx? |
| 14:41.32 | highvoltz | yeah ports are being forwarded to the server |
| 14:41.34 | lordmortis | also is canreinvite set on your in/out trunk? |
| 14:42.02 | highvoltz | let me look |
| 14:42.31 | highvoltz | its not set, so whatever default must be |
| 14:42.39 | [TK]D-Fender | highvoltz: you must set to "no" |
| 14:42.43 | leifmadsen | canreinvite=yes is default |
| 14:42.51 | shamelessn00b | leifmadsen: still same issue |
| 14:43.06 | leifmadsen | shamelessn00b: you have to restart asterisk |
| 14:43.10 | shamelessn00b | I did |
| 14:43.14 | leifmadsen | then you typed something wrong |
| 14:43.30 | highvoltz | ok let me set and retest |
| 14:43.41 | shamelessn00b | noload => pbx_ael.so |
| 14:43.46 | highvoltz | do I set it for both incoming and outgoing? |
| 14:43.51 | [TK]D-Fender | shamelessn00b: PASTEBIN <- |
| 14:43.57 | leifmadsen | shamelessn00b: ls /usr/lib/asterisk/modules/*ael* and add those to modules.conf with the noload operation |
| 14:44.06 | shamelessn00b | ok |
| 14:46.11 | shamelessn00b | noload => res_ael_share.so |
| 14:46.22 | shamelessn00b | still same issue |
| 14:46.27 | [TK]D-Fender | shamelessn00b: PASTEBIN <- |
| 14:46.53 | shamelessn00b | pastebin dialplan and modules ?? |
| 14:46.55 | leifmadsen | if you're doing it right, the AEL modules won't load, and you won't get extensions.ael content |
| 14:47.04 | *** join/#asterisk Defraz (n=Defraz@c72co-edge-router.fuzecore.com) |
| 14:48.23 | shamelessn00b | leifmadsen: now this is uber weird |
| 14:48.28 | [TK]D-Fender | shamelessn00b: the config file you just modified and "ls -la /usr/lib/asterisk/modules" |
| 14:48.32 | shamelessn00b | I STOPPED asterisk |
| 14:48.35 | shamelessn00b | the process |
| 14:48.39 | shamelessn00b | then restarted it |
| 14:48.43 | shamelessn00b | didnt work |
| 14:48.51 | shamelessn00b | then I typed in the command reload in cli |
| 14:48.54 | shamelessn00b | and now its working |
| 14:49.26 | *** join/#asterisk centoslinux (n=linuxcen@rhelbox.uio.no) |
| 14:49.29 | *** join/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/zeeek) |
| 14:50.13 | highvoltz | No go adding canreinvite=no to outgoing/incoming settings on the trunk. still no voice |
| 14:51.00 | leifmadsen | highvoltz: you've setup externip and localhost in sip.conf too right? |
| 14:51.20 | leifmadsen | highvoltz: the other end is probably sending to a private IP as mentioned in the SIP headers |
| 14:51.33 | *** join/#asterisk high-freq (n=hfreq@adsl-99-41-142-239.dsl.ksc2mo.sbcglobal.net) |
| 14:51.54 | highvoltz | let me verify |
| 14:52.13 | *** join/#asterisk Skarmeth (n=Skarmeth@201.57.179.27) |
| 14:52.26 | *** join/#asterisk linuxcentos (n=linuxcen@rhelbox.uio.no) |
| 14:53.57 | highvoltz | right now for incoming settings im just listing host= with the external ip |
| 14:54.26 | highvoltz | so I should be listing externip= and localhost=intenralip? or 127.0.0.1 |
| 14:54.49 | Katty | yawny. |
| 14:54.50 | leifmadsen | highvoltz: your local network (i.e. 192.168.1.0/24) |
| 14:55.08 | leifmadsen | highvoltz: see sip.conf.sample for more information |
| 14:55.28 | highvoltz | ok thanks! |
| 14:55.43 | Katty | it's a beautiful in this neighboorhood |
| 14:55.47 | Katty | a beautiful day for a neighbor |
| 14:56.03 | Zeeek | {{{Katty}}} |
| 14:56.14 | Katty | ohhhhhhh won't you be |
| 14:56.18 | Katty | my neighbor :> |
| 14:56.23 | Katty | hugs Zeeek |
| 14:56.31 | Zeeek | we're all neighbors in cyberspace |
| 14:56.56 | Zeeek | routers are out fences |
| 14:57.05 | Zeeek | PC our fireplaces |
| 14:57.15 | Zeeek | cats our MacBooks |
| 14:57.40 | fenrus | routers are our friends |
| 14:57.59 | Nugget | and IRC is like that annoying neighbor whose HAM radio interferes with your television. |
| 14:58.44 | Zeeek | Nugget: yes! |
| 14:58.45 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
| 14:58.45 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 14:58.50 | Zeeek | is a big ham |
| 14:58.53 | Katty | tinker's with Nugget's tv. |
| 14:59.08 | Katty | WELL DIS HERE thingy goes to that there doohicky |
| 14:59.09 | Zeeek | ... .... .. _ |
| 14:59.12 | Katty | so what we gonna do is |
| 14:59.16 | Katty | just ripp'er out |
| 14:59.21 | Katty | n'that'll get er done |
| 14:59.31 | Katty | ^- Southern Missouri Stereotype. |
| 15:00.17 | *** join/#asterisk grolloj (n=chatzill@h-68-166-73-162.nycmny83.static.covad.net) |
| 15:00.36 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 15:00.37 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:03.32 | Katty | pays bills. |
| 15:03.51 | Katty | i sure do have a lot of bills. |
| 15:04.06 | *** join/#asterisk moy (n=moy@bas1-unionville55-1177733883.dsl.bell.ca) |
| 15:04.11 | Katty | hi moy |
| 15:05.23 | moy | hi Katty, how's it going |
| 15:06.02 | Katty | good good. |
| 15:07.21 | *** join/#asterisk skymeyer (n=skymeyer@mailout.itconnect.be) |
| 15:09.28 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 15:10.03 | Katty | hi jaytee |
| 15:10.08 | jaytee | hi Katty |
| 15:10.20 | *** join/#asterisk cjk (n=cjk@vodsl-10270.vo.lu) |
| 15:10.52 | cjk | hi, what characters can be used in an extension 0-9*# -_ A-Z. anything else? |
| 15:12.18 | jaytee | you could use a tilde, ~. Tilde's are cool! |
| 15:12.29 | cjk | ok thanks |
| 15:12.31 | cjk | i will use tilde |
| 15:12.35 | jaytee | I was kidding! |
| 15:16.46 | cjk | hmm |
| 15:16.51 | cjk | im looking for a seperator |
| 15:17.08 | [TK]D-Fender | cjk: As Tiger Woods.... I'm sure he could spare a few |
| 15:17.30 | cjk | do you have his number? :) |
| 15:21.34 | *** join/#asterisk bsaxon (n=bsaxon@12.68.234.174) |
| 15:24.42 | *** join/#asterisk lordmortis (n=lordmort@203-206-86-190.dyn.iinet.net.au) |
| 15:31.00 | leifmadsen | Asterisk 1.4.29, 1.6.0.21, 1.6.1.13, and 1.6.2.1 are now available! For more information, see the release announcements at http://www.asterisk.org |
| 15:31.41 | *** topic/#asterisk by leifmadsen -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.1 (2010/01/15), Asterisk 1.6.1.13 (2010/01/15), 1.6.0.21 (2010/01/15), 1.4.29 (2010/01/15), *-Addons 1.6.2.0 (2009/12/18), 1.6.1.2 (2009/12/02), 1.6.0.4 (2009/12/02), 1.4.10 (2009/12/02), dahdi-linux 2.2.0.2 (2009/07/23), dahdi-tools 2.2.0 (2009/06/24), Libpri 1.4.10.2 (2009/10/20) -=- Related channels: #asterisknow #switchvox #asterisk-bu |
| 15:34.41 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
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| 15:52.24 | Zeeek | By the way people, the author of the book Hacking VoIP is our guest on VUC today, see http://vuc.me for info and come over to #vuc and say hello to the community. You can call in via SIP or even Skype (for asterisk) |
| 15:53.05 | Zeeek | The VUC happens in about one hour from now and goes on long after, at least 2 hours. |
| 15:53.15 | Naikrovek | what is vuc |
| 15:54.04 | Kobaz | vuc you! |
| 15:54.16 | Kobaz | sounds like some sort of internet radio |
| 15:54.59 | *** join/#asterisk KavanS (n=KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
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| 16:00.02 | momelod | Greetings channel |
| 16:00.35 | *** join/#asterisk GNU\colossus (i=anonymou@truschnigg.info) |
| 16:01.04 | momelod | is there any application for doing surveys? call a list of numbers, ask a set of questions, and record the answers? |
| 16:02.14 | Naikrovek | asterisk is use-agnostic. if you want it to do something, you'll have to script it up |
| 16:02.23 | Naikrovek | whether or not someone else has already done this, i don't know |
| 16:03.06 | carrar | However it sounds like a pretty easy thing to write momelod |
| 16:03.25 | *** join/#asterisk edwin_quijada (n=macaruch@200.26.172.50) |
| 16:03.25 | momelod | i wanted to play a prank on my office.. have each handset ring, ask some silly questions and then later play the recorded answers over our speaker system.. (its friday...) |
| 16:03.46 | carrar | Looking to get fired uh? |
| 16:03.55 | momelod | nah, my office is kewl like that |
| 16:04.06 | Naikrovek | hehe |
| 16:04.15 | Naikrovek | yeah that wouldn't be hard to write i don't think |
| 16:04.20 | momelod | everyone would appreciate a good laugh |
| 16:04.49 | momelod | alright, well ill get to work on it.. just thought maybe someone already had done it |
| 16:05.50 | jaytee | this is why America is not as productive and competitive in the global market any more |
| 16:06.11 | edwin_quijada | How can I do a call from my AGI to another extension |
| 16:06.22 | momelod | actually ill save it till next friday.. today i put up an add on criagslist advertising a giveaway of puppies.. i put up an DID number and forwarded all those calls to someone :) |
| 16:06.36 | momelod | jaytee, im not in american |
| 16:06.48 | momelod | and when was america ever productive? |
| 16:06.53 | *** join/#asterisk Gugge (n=gugge@vlan2.dlxhosting.dk) |
| 16:06.55 | momelod | i thought they were a consumer market :P |
| 16:07.14 | *** join/#asterisk norus (n=n0rus@217.26.192.224) |
| 16:07.50 | momelod | fone fun fridays :) |
| 16:07.59 | bmoraca_work | america can certainly be productive and competitive...biggest problem is the huge number of regulations that corporations have to jump through...but that's a topic for another day :P |
| 16:08.06 | bmoraca_work | er...another channel |
| 16:08.48 | jaytee | momelod, ah, you're in Canada? |
| 16:09.53 | momelod | i think a fun workplace is more productive anyway, people are happier which results in better output, and u get a higher class of people because the hiring pool is that much bigger when your company is fun, more people apply. |
| 16:10.20 | momelod | jaytee, yupp, canadian eh |
| 16:11.07 | jaytee | momelod, down here the attitude is more like, "The floggings will continue until morale improves!" |
| 16:11.10 | Zeeek | vuc is the VoIP Users Conference, http://voipusersconference.org - it is a live, international group that meets every Friday at this time. Wde have a great time, give away hardware and books and learn things. What more can you ask of life? |
| 16:11.36 | momelod | jaytee, sounds oppressive |
| 16:11.48 | Naikrovek | no one uses "flogging" anymore. the way I hear it is "employment terminations will continue until morale improves" |
| 16:12.04 | Naikrovek | or |
| 16:12.15 | Naikrovek | "extended work hours will continue until morale improves" |
| 16:12.20 | *** join/#asterisk muiro (n=muiro@unaffiliated/muiro) |
| 16:12.42 | coppice | "investment in product development will continue to be cut until sales improve" seems to be the real one |
| 16:13.53 | muiro | MeetMe question. What I'm attempting to do is mute all audio to/from one person on the call. I was attempting to use MeetMeAdmin with option M for this, but this only mutes audio coming from the person. I want to mute audio from going to that person. How can I do this? |
| 16:14.21 | edwin_quijada | How can I do a transfer from my AGI to another extension |
| 16:15.04 | Kobaz | edwin_quijada: use the set context/extension/priority commands from AGI and then exit your agi application |
| 16:15.12 | *** join/#asterisk afink (n=afink@204.26.87.226) |
| 16:15.18 | momelod | anyways, no matter where u work, or how backwards their policies are, i dont see any harm in good natured fun, as long as its done in good taste and with no disruption to "productivity" |
| 16:15.33 | momelod | but maybe my jokes are getting a little too elaborate |
| 16:15.38 | momelod | i just cant help myself |
| 16:15.58 | Zeeek | over to #vux now, see you in 45 minutes |
| 16:16.01 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 16:16.10 | Zeeek | or maybe it's #vuc |
| 16:16.16 | Zeeek | anyway... |
| 16:16.24 | *** part/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/zeeek) |
| 16:16.57 | Kobaz | ah, i noticed CEL is in... that's some cool stuff |
| 16:17.05 | Kobaz | er, wrong window |
| 16:17.27 | afink | [TK]D-Fender: Mind if I pm you? I have some questions about your consulting work |
| 16:20.38 | [TK]D-Fender | afink: shoot |
| 16:23.42 | edwin_quijada | Kobaz : I mean $AGI->exec('Dial','SIP/2031'); ?? |
| 16:23.56 | *** join/#asterisk casix (n=casix@xenpbxedifici.adamvozip.es) |
| 16:24.02 | Kobaz | edwin_quijada: that works |
| 16:24.18 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 16:24.18 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 16:24.22 | edwin_quijada | Kobaz: Thks |
| 16:24.45 | Kobaz | edwin_quijada: think of AGI as a wrapper around asterisk dialplan |
| 16:24.53 | Kobaz | edwin_quijada: you can do anything in AGI that you could ith extensions.conf |
| 16:26.54 | casix | hola |
| 16:26.56 | casix | hello |
| 16:30.24 | muiro | is there a way I can set a MeetMe conference user to Talk Only mode some time during the conference, as opposed to only when they first enter? |
| 16:32.04 | *** join/#asterisk telnettech (n=telnette@cpe-71-74-91-116.insight.res.rr.com) |
| 16:35.08 | *** join/#asterisk uluatu (n=uluatu@189.58.22.227.dynamic.adsl.gvt.net.br) |
| 16:35.32 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 16:36.38 | *** join/#asterisk huey23 (n=pleasedu@65.111.241.212) |
| 16:36.47 | Naikrovek | talk only? |
| 16:36.55 | Naikrovek | you want them not to hear the conference? |
| 16:37.21 | p3nguin | Can MeetMeAdmin() alter it after someone joined? |
| 16:38.06 | Naikrovek | i think there are codes users can themselves enter to mute themselves or whatever, right? |
| 16:38.18 | Naikrovek | maybe something could be done that way |
| 16:38.24 | Naikrovek | i don't effing know |
| 16:40.22 | huey23 | i found an old zapmicro zma400p card laying around, does anyone know anything about this manufacturer |
| 16:41.56 | [TK]D-Fender | huey23: cheap knockoff |
| 16:42.14 | [TK]D-Fender | huey23: Probably work about as well as the original... which is several revisions old as it is |
| 16:42.18 | huey23 | i kind of figured that seeing as how their website doesn't work |
| 16:42.22 | *** join/#asterisk KavanS (n=KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 16:43.36 | huey23 | i'll play around with it to see if it's worthy; if not, i guess i will have to replace with "the original" |
| 16:45.22 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.157) |
| 16:45.53 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 16:51.04 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 16:51.04 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 16:52.05 | *** join/#asterisk saghul (n=saghul@ip3e830637.speed.planet.nl) |
| 16:53.38 | garymc | Yo! Anyone know if i can make asterisk play my telco error messages instead of hearing the server error message. eg when i dial an incorrect number? cos it is starting to do my head in. I get the same message for all errors. I want to hear my telco messages as they are more understandable |
| 16:54.30 | garymc | so when i dial an incorrect number I get the BT message instead of "All circuits are busy" |
| 16:55.08 | [TK]D-Fender | garymc: Answer in your channel.... |
| 16:55.19 | garymc | ok |
| 16:55.19 | Kobaz | garymc: and don't use the 'r' option to dial |
| 16:55.32 | garymc | what option should i use? |
| 16:55.32 | *** join/#asterisk jtodd (i=m2amrf88@asterisk/community-director-and-tie-dye-shirt-lover/jtodd) |
| 16:55.32 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 16:55.36 | muiro | Naikrovek: yeah, I want to set one user so that they can not hear the conference. MeetMeAdmin() can do some stuff, but I'm not sure if there's a way to do what I'm trying to do |
| 16:55.48 | Kobaz | garymc: whatever other options you want, but not 'r' |
| 16:56.13 | Naikrovek | i don't understand why you want someone to be able to speak into, but not hear, the conference |
| 16:56.38 | garymc | kobaz so where can i find what options there are and what they do? |
| 16:56.42 | muiro | Naikrovek: well, I don't want them to be able to speak in to -or- hear the conference |
| 16:56.50 | Naikrovek | uh |
| 16:56.56 | p3nguin | An Answer() in the channel is going to prevent the telco messages from being heard, isn't it? |
| 16:57.24 | Kobaz | garymc: core show application Dial |
| 16:57.40 | Kobaz | p3nguin: no... that will allow early media to come through |
| 16:57.44 | [TK]D-Fender | Kobaz: Not appicable. |
| 16:57.47 | Naikrovek | muiro: presumably you want them connected to the conference normally for some amount of time before or after they're silenced and deafened? |
| 16:57.53 | [TK]D-Fender | Kobaz: leave this one.. trust me... |
| 16:57.58 | Kobaz | heh |
| 16:58.05 | garymc | I got option, my asterisk dial command is tr |
| 16:58.21 | Kobaz | but if you are using r, it's going to ring, and hang up when it gets an error |
| 16:58.22 | [TK]D-Fender | garymc: NOT APPLICABLE. Stop now. |
| 16:58.31 | Kobaz | [TK]D-Fender: why not? |
| 16:58.37 | p3nguin | muiro: If you don't want the caller to be able to talk nor hear the conference EVER, then don't allow him to call the conference. |
| 16:58.38 | [TK]D-Fender | Kobaz: PRI <---- OOB |
| 16:58.38 | muiro | Naikrovek: I need to give the admin the ability to, on command, silence and deafen one user. Then, later, bring them back in |
| 16:58.40 | garymc | ok |
| 16:58.44 | Kobaz | oh, this is pri? |
| 16:58.53 | [TK]D-Fender | Kobaz: OH.. NOW you're going to ask? :p |
| 16:58.56 | Kobaz | hah |
| 16:58.57 | Kobaz | sorry |
| 16:59.01 | Naikrovek | it's also freepbx |
| 16:59.56 | Naikrovek | but is it telnet? *waits for nugget* |
| 16:59.56 | Nugget | telnet is eeeeeeevil! |
| 16:59.58 | Naikrovek | yup |
| 17:00.08 | Kobaz | we need a deragatory misrepresentation of the word freepbx |
| 17:00.16 | Naikrovek | no we don't |
| 17:00.17 | garymc | pisspbx? |
| 17:00.24 | garymc | ;) |
| 17:00.36 | Naikrovek | freepbx is fine, just not in here |
| 17:01.04 | Naikrovek | nothing wrong with it per se, just not supported in here. trixbox otoh |
| 17:01.08 | p3nguin | garymc: If you see r in the Dial() command, consider removing it. |
| 17:01.19 | Naikrovek | p3nguin: dial options of tr is default on freepbx |
| 17:01.29 | garymc | yes ive got tr |
| 17:01.37 | garymc | dont know what it means though? |
| 17:01.52 | Naikrovek | well the |
| 17:01.52 | Naikrovek | r |
| 17:01.52 | [TK]D-Fender | garymc: Stop asking across multiple channels |
| 17:01.56 | Naikrovek | yeah it's confusing |
| 17:01.59 | p3nguin | ~r |
| 17:02.00 | infobot | rumour has it, r is The "r" option to Dial will override any sounds you should be hearing and provide a fake ringing sound to the caller. You generally want the caller to hear the sounds they are supposed to hear, not a fake ringing sound. The caller will hear ringing without the "r" option. Using the "r" option is an edge case and should not normally be used or needed. |
| 17:02.03 | Naikrovek | hehe trying to follow both conversations |
| 17:02.11 | garymc | hehe ok ill stop |
| 17:02.20 | garymc | everyone come over to freepbx :P |
| 17:02.48 | jarrod | asterisk > freepbx |
| 17:02.50 | jarrod | *duck* |
| 17:03.07 | p3nguin | That's kind of a retarded comparison. |
| 17:03.46 | garymc | bloody hell how do i stop sip debug as "sip debug off" isnt working |
| 17:03.53 | p3nguin | Similar to "steering wheel > automobile" |
| 17:03.57 | *** part/#asterisk freckle_work (n=jon@87.127.248.145) |
| 17:04.07 | p3nguin | sip set debug off |
| 17:04.08 | *** join/#asterisk bcnyc (n=bill@208.79.183.212) |
| 17:04.09 | Kobaz | garymc: look at the help for sip debug |
| 17:04.19 | p3nguin | That's the opposite of sip set debug, by the way. |
| 17:04.57 | garymc | p3nguin thanks |
| 17:05.21 | jarrod | heh |
| 17:07.01 | muiro | okay, I think I might have a way to silence and deafen one user. MeetMeAdmin() has the option M which I can use to mute audio coming -from- the user, and it also has option "u" which is "lower one user's listen volume." However, I'm not sure if "u" works at all. Perhaps I'm not invoking it correctly, but it seems to do nothing |
| 17:07.55 | muiro | I love reading source to try to figure out how an app works, lol |
| 17:09.04 | Kobaz | muiro: once you're done reading, feel free to submit updated documentation |
| 17:09.19 | Kobaz | it's all community effort |
| 17:11.08 | muiro | Kobaz: I'll consider it |
| 17:21.30 | *** join/#asterisk RobH (n=robh@cpe-173-169-30-118.tampabay.res.rr.com) |
| 17:22.15 | leifmadsen | muiro: or file a bug so it can be fixed |
| 17:22.48 | *** join/#asterisk rossand (n=aross@red-gw42.cs.toronto.edu) |
| 17:24.27 | *** join/#asterisk vally (i=kvirc@ip-92-50-113-186.unitymediagroup.de) |
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| 17:33.51 | p3nguin | katty: http://imagebin.org/80047 |
| 17:34.47 | Naikrovek | muiro: please file bug report if it is indeed broken. |
| 17:34.51 | Naikrovek | i emplore you |
| 17:34.55 | Naikrovek | not enough people do this |
| 17:40.35 | *** join/#asterisk paulc (n=paulc@unaffiliated/paulc) |
| 17:45.06 | *** join/#asterisk rossand (n=aross@sb-sys.cs.toronto.edu) |
| 17:45.24 | muiro | oh, no, it's not a bug. You can decrease someone's listen volume by up to a factor of 5 (whatever that stands for), but this mute them, just lowers it. Can not lower below -5. |
| 17:45.37 | muiro | *doesn't mute them |
| 17:46.27 | Kobaz | so it should probably note that in the documentation |
| 17:48.10 | muiro | yeah, I'll make a note to add that information when I'm not blitzkreig coding |
| 17:50.15 | *** join/#asterisk klochan (n=klochan@95-27-73-139.broadband.corbina.ru) |
| 17:50.21 | *** join/#asterisk Ad-Hoc (n=nimbus@62.1.142.230.dsl.dyn.forthnet.gr) |
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| 17:53.00 | Kobaz | where did app_meetme go in mantis |
| 17:53.03 | Kobaz | er |
| 17:53.19 | *** join/#asterisk puzzled_ (n=patrick@puzzled.xs4all.nl) |
| 17:53.27 | *** join/#asterisk dandre (n=daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 17:53.50 | voipmonk | we ate it |
| 17:54.39 | *** join/#asterisk aidinb (n=Aidin@24-176-216-154.dhcp.lnbh.ca.charter.com) |
| 17:55.06 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 17:57.02 | bmoraca_work | uhg...freakin receptionists. transfered trouble calls to my voicemail instead of to me. |
| 17:58.57 | klochan | is there some way to if sip-user is offline or there is no such user? (to playback different recordings) =) |
| 17:59.11 | klochan | *some way to know =) |
| 17:59.45 | bmoraca_work | yes |
| 18:00.11 | bmoraca_work | theck DIALSTATUS: http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS |
| 18:00.32 | klochan | m-m |
| 18:00.32 | bmoraca_work | or you can use CHANISAVAIL as well |
| 18:00.42 | bmoraca_work | which is the right one depends on exactly what you want to do |
| 18:00.47 | klochan | it will return CHUNUNAVAIL |
| 18:00.52 | klochan | os both variants |
| 18:01.17 | bmoraca_work | if all you need to do is determine whether or not the channel is available (including call limits, etc), then CHANISAVAIL might be simpler |
| 18:02.27 | klochan | i want smth else =) |
| 18:02.41 | bmoraca_work | not by what you described you don't :P |
| 18:02.50 | bmoraca_work | but DIALSTATUS does give you more options |
| 18:03.51 | *** join/#asterisk tgunr (n=tgunr@cust-66-249-166-12.static.o1.com) |
| 18:04.53 | klochan | in sip.conf we have some some users (e.g. 101,102,103... ) I want: if i call from aster to 101 and thats user is offline - i will get Playback(unavailable), but if i call 201 (there is no user in sip.conf) - i want to listen Playback(invalid) |
| 18:04.54 | *** join/#asterisk Chinorro (n=Chino@19.226.117.91.dynamic.mundo-r.com) |
| 18:04.58 | klochan | something like this |
| 18:05.30 | klochan | in both variants i will get code 20 (chununavail) |
| 18:05.47 | klochan | it's something like in gsm-operators |
| 18:06.04 | *** part/#asterisk andreas-- (n=andy@unaffiliated/slacky) |
| 18:06.13 | bmoraca_work | i'm not sure you can do that with just dialplan. |
| 18:06.56 | klochan | ok, have i do whis with AGI? |
| 18:07.43 | bmoraca_work | you could. parse "sip show peers" or your sip.conf file directly. or you could use RealTime. that'd be easiest. |
| 18:08.07 | bmoraca_work | one FUNC_ODBC function with RealTime and a couple Gotoifs would handle it |
| 18:08.09 | *** join/#asterisk [netman] (n=netman@17.Red-81-36-134.dynamicIP.rima-tde.net) |
| 18:08.24 | p3nguin | klochan: http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS |
| 18:08.37 | ManxPower-work | ~hangupcause |
| 18:08.38 | infobot | i guess hangupcause is Q.931 Hangup Causes can be found at http://www.quintum.com/support/xplatform/network/Q931_Disconnect_Cause_Code_List.pdf OR Q.931 <-> SIP Codes can be found at http://www.faqs.org/rfcs/rfc3398.html OR HANGUPCAUSE dialplan variable info (possibly outdated) at http://www.voip-info.org/wiki/index.php?page_id=569 |
| 18:08.54 | bmoraca_work | p3nguin, that doesn't distinguish between a peer that is not available and a peer that doesn't exist. |
| 18:09.01 | p3nguin | klochan: Take a look at how they created that macro. It could help you develop a dialplan that fits your need. |
| 18:09.54 | bmoraca_work | klochan, ManxPower-work is right. hangupcause will do what you need. |
| 18:09.57 | bmoraca_work | er |
| 18:09.58 | bmoraca_work | nm |
| 18:10.06 | bmoraca_work | maybe |
| 18:10.14 | bmoraca_work | never used it. does hangupcause work over SIP? :P |
| 18:10.50 | klochan | i've tried hangupcause |
| 18:10.57 | klochan | error code 20 in both variants |
| 18:11.05 | carrar | Why are you trying to dial SIP devices that don't exist in the first place |
| 18:11.43 | klochan | sorry, don't understand =) what u mean "in the first place"? |
| 18:12.00 | carrar | You want to know if a SIP devices exist or not |
| 18:12.09 | klochan | yes |
| 18:12.14 | ManxPower-work | Remember, without qualify=something you'll have to wait for whatever LOOOONNNNGGGGG timeout SIP has when a device does not responed. |
| 18:12.16 | carrar | Why would you dial one that doesn't exist in the first place |
| 18:12.21 | p3nguin | Yeah, just don't Dial() it at all. |
| 18:12.28 | *** join/#asterisk [netman] (n=netman@17.Red-81-36-134.dynamicIP.rima-tde.net) |
| 18:12.52 | klochan | sometimes we can mistake ) |
| 18:12.55 | ManxPower-work | You could try using ChanIsAvail or similar |
| 18:12.59 | carrar | fix those mistakes |
| 18:13.02 | carrar | someplace else |
| 18:13.44 | carrar | or us ChanIsAvail like max said |
| 18:13.46 | bmoraca_work | klochan, your context in extensions.conf should have a catch-all and your patterns should be specific enough that non-existent peers wouldn't match any of them except the catch-all. |
| 18:13.56 | carrar | use |
| 18:14.11 | bmoraca_work | klochan, i can see why you might want to do what you want to do, but it speaks of design flaws in extension.conf |
| 18:14.16 | ManxPower-work | bmoraca_work: Yes, that is the *correct* way do handle the issue |
| 18:14.18 | *** join/#asterisk ruben23 (n=AGENT@122.55.48.243) |
| 18:14.24 | bmoraca_work | carrar, doesn't distinguish between "not registered" and "doesn't exist" |
| 18:14.34 | carrar | go read it again |
| 18:14.43 | *** join/#asterisk lanning (n=lanning@208.87.235.224) |
| 18:15.42 | bmoraca_work | ahhh |
| 18:15.48 | bmoraca_work | AVAILSTATUS will do it |
| 18:16.15 | bmoraca_work | wasn't familiar with that, as "core show application chanisavail" doesn't detail the variables very well |
| 18:16.49 | klochan | i think, that's what i want (AVAILSTATUS i mean) |
| 18:17.00 | bmoraca_work | klochan, http://www.voip-info.org/wiki/view/Asterisk+cmd+ChanIsAvail |
| 18:17.44 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:17.58 | klochan | thank's a lot )) i'll try =) |
| 18:20.04 | *** join/#asterisk errr_ (n=errr@fedora/errr) |
| 18:24.33 | *** join/#asterisk Tim_Toady (n=moi@77.49.24.242.dsl.dyn.forthnet.gr) |
| 18:27.38 | klochan | bmoraca_work, "Chanisavail is not intended to detect if a phone is in use or not at all, it's only intended to check if asterisk could send the call there" =)) |
| 18:28.01 | *** join/#asterisk jaskew (n=jdaskew@netblock-66-159-217-102.dslextreme.com) |
| 18:31.35 | *** join/#asterisk freddyk (n=freddyk@host241-134-dynamic.51-82-r.retail.telecomitalia.it) |
| 18:31.40 | klochan | but i can trace by device_state |
| 18:31.45 | klochan | that is it |
| 18:32.17 | *** join/#asterisk benngard (n=benngard@90-230-92-67-no148.tbcn.telia.com) |
| 18:32.54 | *** join/#asterisk dmz (n=dmz@64.203.207.101.dyn-cm-pool-54.hargray.net) |
| 18:33.19 | p3nguin | klochan: If it can't send the call there, send it somewhere else. Should be just as easy as not Dial()ing an unused device in the first place. |
| 18:33.37 | *** join/#asterisk puzzled (n=foobar@puzzled.xs4all.nl) |
| 18:33.45 | klochan | no |
| 18:34.34 | p3nguin | I'm still leaning toward just not Dial()ing it at all. Seems the more sane approach. |
| 18:34.47 | klochan | e.g. the call can't be sent in two variants: 1) no such user at all 2) user is offline |
| 18:35.24 | p3nguin | Why is it so hard to NOT DIAL() THE DAMN THING AT ALL? |
| 18:35.25 | klochan | i'll have to tell clients that they were mistaken with diled number |
| 18:36.48 | p3nguin | Create patterns for the things you want dialed. Create matches for everything you don't want dialed, and use Goto() to send those to an "invalid" response. |
| 18:37.17 | klochan | u don't understand what i want |
| 18:37.42 | p3nguin | Or "u" don't understand what "you" want. |
| 18:37.57 | klochan | i'm understand exactly and now it works =) |
| 18:39.29 | dlynes | Is there a master document that tells exactly what is in each sound file, without having to listen to all the sound files and annotate them? |
| 18:40.00 | *** join/#asterisk moos3 (n=rgenthne@216.52.121.66) |
| 18:40.21 | moos3 | can mixmonitor email recordings once done? |
| 18:40.30 | dlynes | moos3, no, but asterisk can |
| 18:41.33 | moos3 | I'm trying to setup something to records calls in a queue that will record them and then email them to a super visor once the call between agent and client is done |
| 18:41.36 | moos3 | ideas on that? |
| 18:42.15 | dlynes | moos3, shell script |
| 18:42.24 | dlynes | moos3, system(...) |
| 18:42.53 | moos3 | ok I get how to handle the system() call how can my dailplan figure out if the call between them is done |
| 18:43.06 | dlynes | moos3, because Dial() returns??? |
| 18:43.25 | moos3 | k |
| 18:44.05 | dlynes | man is rhythmbox ever slow |
| 18:51.09 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 18:55.17 | *** join/#asterisk Micc_ (n=quassel@c-98-225-57-96.hsd1.wa.comcast.net) |
| 18:55.21 | moos3 | dlynes: how does I implement that into this http://pastie.org/779915 |
| 18:58.05 | *** join/#asterisk tgunr (n=tgunr@cust-66-249-166-12.static.o1.com) |
| 18:59.58 | [TK]D-Fender | moos3: All wasted ExecIf's in there. Could solve with 1 jump |
| 19:00.27 | moos3 | really? |
| 19:00.31 | moos3 | how |
| 19:00.42 | [TK]D-Fender | moos3: 1 GotoIf instead. |
| 19:01.26 | [TK]D-Fender | moos3: You also have no invalid handler |
| 19:01.35 | *** join/#asterisk e4 (n=e4@rrcs-76-79-59-194.west.biz.rr.com) |
| 19:02.18 | moos3 | [TK]D-fender: yeah I know and it goes to voicemail if you sit in the queue for 5 minutes with out anyone getting to you |
| 19:03.27 | moos3 | need to figure out how to auto record calls between the caller and agent in that queue |
| 19:04.27 | ruben23 | hi, nay recommended IP phones best for asterisk..easy to configure. |
| 19:05.10 | moos3 | ploycoms |
| 19:05.25 | p3nguin | I'm satisfied with Cisco 7900 series. |
| 19:05.29 | klochan | bmoraca_work, http://pastie.org/779928 it's an example what i want =) if it's interesting for u =) thanks |
| 19:07.47 | moos3 | i use plycoms in our office of 60 people or so |
| 19:08.09 | moos3 | no hacking need to get to sip to work :) |
| 19:10.18 | moos3 | [tk]-d-fender: how can i handle the auto call recording |
| 19:10.43 | *** join/#asterisk rizwank (n=rizwank@76.89.131.47) |
| 19:10.59 | ruben23 | p3nguin: what in particular model you have on cisco 7100 series..? |
| 19:11.37 | p3nguin | ruben23: I'm satisfied with 7940G/7960G as well as 7912G phones. I run SIP images on them all. |
| 19:12.42 | rizwank | My calling card company wants to provide free access for Hatians to call out to the US during the post-earthquake recovery. I can't seem to find any companies that have DIDs for sale -- can anyone recommend a resource? |
| 19:12.58 | p3nguin | But now I am considering "testing" an SCCP image on a 7940G and using chan_skinny just to see what happens. |
| 19:13.29 | p3nguin | rizwank: DIDs with numbers in what country? |
| 19:13.34 | rizwank | Haiti. |
| 19:14.30 | ruben23 | moos3:what polycom model is that..? |
| 19:14.41 | moos3 | 601s 301s |
| 19:14.53 | moos3 | we are going to update to the newer models soon |
| 19:15.07 | [TK]D-Fender | ruben23: For you.... Linksys SPA series. |
| 19:16.28 | moos3 | I have a IVR with options, 1, 2, 3, 4, 5 and theres some people with extensions that are 2xxx that are getting told invalid extension |
| 19:16.30 | p3nguin | rizwank: I guess Flowroute has Haiti DIDs. |
| 19:17.13 | *** join/#asterisk rcampbell2 (n=rcampbel@h83.140.89.75.dynamic.ip.windstream.net) |
| 19:17.18 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:17.18 | rizwank | They do? |
| 19:18.08 | rizwank | negatory - just called them. |
| 19:18.44 | *** join/#asterisk magronez (n=eusei@unaffiliated/magrao/x-2903) |
| 19:20.26 | [TK]D-Fender | moos3: Because it doesn't have extens to match |
| 19:20.34 | p3nguin | they have _one_ on their list. |
| 19:20.54 | *** part/#asterisk rcampbell2 (n=rcampbel@h83.140.89.75.dynamic.ip.windstream.net) |
| 19:21.30 | moos3 | no there users extensions is in the sip.conf |
| 19:21.40 | Naikrovek | why would you use a cisco phone on an asterisk system |
| 19:21.53 | Naikrovek | you still need to buy the call manager license i think... |
| 19:21.59 | Kobaz | hmm |
| 19:22.37 | Naikrovek | qwell wrote an open letter to a cisco higher up about it and he responded openly saying call manager license was required even if you don't use call manager or a cisco phone system |
| 19:22.58 | p3nguin | rizwank: Oh, sorry. I was given bad info. Area code 509 isn't Haiti. |
| 19:23.14 | rizwank | Can anyone suggest a forum/IRC Channel where I might be able to find someone knowledgeable about Haitian DIDs? (International calling code 509). |
| 19:23.21 | rizwank | Yeah, easy mistake. Thanks p3nguin. |
| 19:24.22 | [TK]D-Fender | moos3: No |
| 19:24.26 | Qwell | areacode? is Haiti even NANP? |
| 19:24.32 | [TK]D-Fender | moos3: extensions = extensions.conf |
| 19:24.32 | Kobaz | Naikrovek: that's retarted |
| 19:24.53 | [TK]D-Fender | loves tarts |
| 19:25.06 | Qwell | No, per wikipedia, Haiti isn't part of NANP. |
| 19:25.07 | p3nguin | qwell: I doubt it. I was just provided bad info, that's all. |
| 19:25.09 | Kobaz | retarded rather |
| 19:25.59 | moos3 | [TK]D-Fender: so what your saying is that I need to make a extension in extension that dails itself? |
| 19:26.11 | Kobaz | dials a device |
| 19:26.20 | p3nguin | qwell: The person mistook NANP area code 509 for Haiti's country code of 509. |
| 19:26.25 | [TK]D-Fender | moos3: stop calling SIP PEERS as EXTEnsiONS. |
| 19:26.42 | Qwell | p3nguin: ahh |
| 19:29.00 | Naikrovek | Kobaz: i agree, but apparently it's the way it is |
| 19:29.02 | moos3 | so if I hear you right I need a exten => 2xxx entry? |
| 19:29.42 | p3nguin | _2XXX |
| 19:29.49 | moos3 | ie this exten => _XXXX,1,Macro(stdexten,${EXTEN},SIP/${EXTEN},70) ;Catch all extensions not defined above |
| 19:29.49 | moos3 | exten => _XXXX,2,Congestion |
| 19:30.32 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 19:30.41 | hardwire | [TK]D-Fender: like ManxPower-work ? |
| 19:31.45 | [TK]D-Fender | hardwire: UCAN HAN HAZ CONTEXT? |
| 19:31.51 | hardwire | tarts. |
| 19:32.09 | hardwire | ducks |
| 19:33.40 | *** join/#asterisk luckyaba (n=lucky@ip72-194-215-55.sb.sd.cox.net) |
| 19:34.12 | moos3 | [tk]d-fender: whats the issue with my 2xxx extensions then? |
| 19:35.52 | [TK]D-Fender | moos3: Where do I see you showing me revised dialplan and a failed call? |
| 19:36.43 | moos3 | may dailplan is more then 2K+ lines |
| 19:37.03 | [TK]D-Fender | moos3: How about the context in question. |
| 19:37.11 | Naikrovek | it's not all applicable, show the correct area |
| 19:37.43 | hardwire | or gzip it :) |
| 19:38.21 | *** join/#asterisk Skeeter- (i=Skeeter@190-141.cgocable.ca) |
| 19:40.51 | ruben23 | <PROTECTED> |
| 19:41.31 | p3nguin | ruben23: You said two totally different subjects. Phones. Local extensions. Which one are you asking me about? |
| 19:42.08 | ruben23 | p3nguin:analog phones. |
| 19:42.52 | p3nguin | ruben23: Pick up a cheap phone at the store, such as Walmart for $10 and a $35 PAP2 ATA from an online store. |
| 19:44.14 | hardwire | anybody have a more than straightforward way to reassign CID based on trunk per peer? |
| 19:44.29 | hardwire | I suppose global variables would work |
| 19:44.33 | hardwire | or peer variables |
| 19:44.34 | hardwire | hmm. |
| 19:46.24 | carrar | uh |
| 19:49.44 | *** join/#asterisk Tech_Travis (n=Travis@mail.techglia.com) |
| 19:50.17 | moos3 | [TK]D-Fender: here you go http://pastie.org/779991 |
| 19:50.50 | [TK]D-Fender | moos3: and the rest/ |
| 19:50.58 | moos3 | workign on it |
| 19:54.44 | moos3 | [TK]D-Fender: http://pastie.org/private/khiwwxigc4ifoz74gcdcjq |
| 19:54.53 | *** join/#asterisk Chinorro (n=Chino@19.226.117.91.dynamic.mundo-r.com) |
| 19:56.54 | [TK]D-Fender | moos3: And the REST of what I asked for? |
| 19:57.14 | moos3 | yeah give me a seocnd to find a log |
| 19:57.24 | [TK]D-Fender | moos3: Live CLI, no logs |
| 20:00.35 | moos3 | here you go |
| 20:00.36 | moos3 | http://pastie.org/780024 |
| 20:00.40 | moos3 | I mask my cellphone |
| 20:00.45 | *** join/#asterisk sawgood (n=sawgood@173-13-158-25-sfba.hfc.comcastbusiness.net) |
| 20:02.36 | [TK]D-Fender | moos3: And what line do YOU think should match that? |
| 20:03.22 | moos3 | line 515 |
| 20:03.57 | moos3 | see I dailed 2003 |
| 20:04.17 | moos3 | its almost like its not waiting for the rest of the digits |
| 20:04.20 | carrar | I see 35 lines |
| 20:04.48 | carrar | where is the other 480 lines? |
| 20:04.49 | [TK]D-Fender | moos3: -- Invalid extension '003' in context 'incoming' on DAHDI/7-1 |
| 20:05.00 | [TK]D-Fender | moos3: No, I see you dialed 003 |
| 20:05.27 | carrar | ZERO |
| 20:07.20 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 20:08.53 | moos3 | yeah I can't figure out why some times it drops the first digit |
| 20:09.29 | moos3 | how can I drop a sip connection to a meetme |
| 20:11.00 | [TK]D-Fender | moo3What is a "sip connection", and what is it doing prior? |
| 20:11.52 | moos3 | some how its in a meetme that it should be |
| 20:12.54 | moos3 | shouldn't be |
| 20:13.16 | [TK]D-Fender | moos3: Why don't they hang up and try going where their supposed to again? |
| 20:13.22 | rizwank | Is AGI a language that can be used to replace the older extensions language -- i.e. used to handle incoming calls and route them? |
| 20:13.31 | moos3 | lol it was a stuck sip connection |
| 20:13.49 | moos3 | that had no one in there |
| 20:14.13 | [TK]D-Fender | moos3: then use "soft hangup [channel]" to kill it, and if that fails, an AMI redirect to an exten that will do the same |
| 20:14.25 | moos3 | ok cool thanks |
| 20:15.42 | moos3 | so how do I make it asterisk not think the first digit is the menu answer right away? and wait 1 second for more digits if not more digits then enter menu |
| 20:15.54 | moos3 | I think that should fix the problem i'm having |
| 20:16.17 | [TK]D-Fender | moos3: Set your diigit timeouts properly |
| 20:16.30 | moos3 | how do I do that? |
| 20:17.48 | *** join/#asterisk joako (n=ston3d@opensuse/member/joak0) |
| 20:18.13 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:19.00 | [TK]D-Fender | moos3: "core show function TIMEOUT" |
| 20:20.42 | moos3 | that doesn't seem like 5 seconds would be to fast |
| 20:24.38 | *** join/#asterisk rossand (n=aross@red-gw42.cs.toronto.edu) |
| 20:28.37 | p3nguin | moos3: MeetMe() also has options to pass DTMF "thru" the conference. Maybe that can help. |
| 20:29.50 | benngard | damn funny weakend i gonna have, we had a meetining with avaya (our pbx supplier) today, either we have to pay big $$$ to upgrade or go for an opensource solution, my boss "claims" me for an answer monday morning :( |
| 20:30.16 | voipmonk | well what features do you need to have, benngard |
| 20:30.48 | benngard | i think i have settled mi mind |
| 20:30.52 | benngard | but |
| 20:31.00 | voipmonk | drum roll |
| 20:31.02 | voipmonk | but ? |
| 20:31.11 | benngard | can i caount on u? |
| 20:31.15 | benngard | count |
| 20:31.17 | voipmonk | me? |
| 20:31.23 | voipmonk | to do what, benngard ? |
| 20:31.38 | benngard | when i ran into problems ofc |
| 20:31.52 | benngard | i gonna do it, for sure |
| 20:32.12 | Naikrovek | if you describe your problems properly and give us info we ask for to help us help you, then yes |
| 20:32.20 | Naikrovek | i don't see any reason why anyone in here wouldn't help |
| 20:32.27 | voipmonk | just drop in |
| 20:32.29 | voipmonk | and ask |
| 20:32.33 | benngard | thats the answer i was looking for! |
| 20:32.35 | Naikrovek | how many endpoints will the system have |
| 20:32.45 | benngard | around 500 |
| 20:32.49 | Naikrovek | nice |
| 20:32.54 | Naikrovek | single server? |
| 20:32.55 | voipmonk | how many simultaneous calls are you looking to utilize? |
| 20:33.22 | benngard | like 100 calls at a time |
| 20:33.30 | voipmonk | do you need to record any of them? |
| 20:33.36 | benngard | sec |
| 20:33.43 | Naikrovek | probably some |
| 20:34.29 | benngard | i have beenr running part of the company (without my boss knowledge) for some time, asterisk is working ;) |
| 20:34.42 | Naikrovek | niiiice |
| 20:35.18 | voipmonk | sounds like you're well on your way... |
| 20:35.37 | voipmonk | will you keep the avaya in play or transistion those users over to asterisk? |
| 20:35.42 | Naikrovek | i love the under-the-radar efforts to prove things to prejudiced bosses |
| 20:35.45 | voipmonk | migrate, rather |
| 20:36.13 | voipmonk | Naikrovek: I'm hoping he unplugs the avaya and sits it next to the bosses desk |
| 20:36.23 | Naikrovek | yeah :) |
| 20:36.26 | moos3 | I'm runing a quad core 2.6ghz with 4 gigs of ram and I handle 100 end points, 64 to 78 calls, and recording and doesn't phase the box fyi |
| 20:36.32 | voipmonk | put some glass on the top |
| 20:36.34 | benngard | i (if i decide) move extension per extension to * |
| 20:36.36 | voipmonk | and make it a desk |
| 20:36.37 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 20:36.38 | voipmonk | or a stand |
| 20:36.41 | voipmonk | stick a lamp on it |
| 20:36.44 | voipmonk | :) |
| 20:36.44 | Naikrovek | lol |
| 20:37.06 | moos3 | whats the best way to remap extension for all my sip peers? |
| 20:37.08 | Naikrovek | moos3: wow almost save server here. 100 simultaneous calls, all recording, no sweat. 15% cpu or so |
| 20:37.21 | Naikrovek | has been playing witih sipp |
| 20:37.30 | Naikrovek | s/witih/with/ |
| 20:37.54 | voipmonk | make the sip peers use the same device name as an extension number and use something like exten => _XXX,1,Dial(SIP/${EXTEN}|20) |
| 20:38.01 | voipmonk | exten => _XXX,2,Voicemail..... |
| 20:38.03 | voipmonk | blah blah |
| 20:38.16 | benngard | the hard thing for me is: am i willing to take the "programming shit" to move all the users |
| 20:38.20 | moos3 | yeah we are getting ready to pick up a big gov client and will be recording all incoming and out going calls to there numbers so I'm assuming it might bounce to 25% load |
| 20:38.24 | voipmonk | or you can use a macro to do what u need and exchange ${EXTEN} with ${ARG1} |
| 20:39.36 | moos3 | ok, we just want to remap all of our users, to use a schema that makes sense instead of ok heres a random extension |
| 20:48.23 | *** join/#asterisk retentiveboy (n=pdugas@adsl-221-170-128.pns.bellsouth.net) |
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| 20:59.41 | *** join/#asterisk Grapsus (n=grapsus@che21-2-82-245-89-120.fbx.proxad.net) |
| 20:59.45 | Grapsus | Hi! |
| 21:00.09 | voipmonk | hi!!!!! |
| 21:00.45 | Grapsus | I have an asterisk server configured with webcalldirect for external calls but my caller id is always anonymous, how can I change it ? |
| 21:06.09 | *** join/#asterisk aidinb (n=Aidin@24-176-216-154.dhcp.lnbh.ca.charter.com) |
| 21:08.10 | *** join/#asterisk slima (i=slima@unaffiliated/slima) |
| 21:18.34 | bmoraca_work | this is awesome: http://www.theonion.com/content/video/more_american_workers_outsourcing |
| 21:18.45 | bmoraca_work | just so you know |
| 21:18.46 | *** join/#asterisk af_ (n=getsmart@88-149-240-85.dynamic.ngi.it) |
| 21:21.16 | *** join/#asterisk ttl- (n=patrick@d5153A420.access.telenet.be) |
| 21:24.40 | Katty | :< |
| 21:24.44 | Katty | NATIONAL EMERGENCY |
| 21:24.46 | carrar | haha |
| 21:24.49 | Katty | out of chapstick :< |
| 21:24.50 | carrar | that video rocks |
| 21:25.06 | bmoraca_work | i seriously need to consider that, lol |
| 21:26.14 | Katty | watches video |
| 21:26.31 | eppigy | YES |
| 21:26.35 | eppigy | WATCH THE VIDEO |
| 21:26.38 | carrar | oldie but googie |
| 21:28.31 | Katty | lol, that's hilsarious |
| 21:28.39 | [TK]D-Fender | checkout time, BBIAB |
| 21:28.45 | Katty | hilsarious? |
| 21:29.05 | Katty | wonders what happened between brain and fingers ^_- |
| 21:34.02 | af_ | yawn .... what's up? |
| 21:34.30 | eppigy | Katty: a delightful error |
| 21:34.33 | eppigy | i thought |
| 21:38.10 | *** join/#asterisk Qwell (i=north@pdpc/sponsor/digium/Qwell) |
| 21:38.10 | *** mode/#asterisk [+o Qwell] by ChanServ |
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| 21:42.40 | *** part/#asterisk rizwank (n=rizwank@76.89.131.47) |
| 21:45.21 | *** join/#asterisk Ad-Hoc (n=nimbus@193.92.83.140.dsl.dyn.forthnet.gr) |
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| 21:51.14 | PMantis | Hi guys. I have a TE121 in Asterisk from Ubuntu 9.10, fully updated. Using Ubuntu's packages, rather than tar or SVN this time. Asterisk cannot see the DAHDI channels, apparently. Here's my debug info. ideas? http://pastebin.com/d70a868cd |
| 21:51.41 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 21:53.21 | PMantis | Hey [TK]D-Fender. I just posted a Q, *then* you enter. Bad timing. :) |
| 21:54.36 | PMantis | * doesn't see my DHADI channels, http://pastebin.com/d70a868cd |
| 21:55.09 | bmoraca_work | PMantis, is the dahdi service started? |
| 21:55.34 | bmoraca_work | and did you restart asterisk after running dahdi_config |
| 21:57.16 | PMantis | bmoraca_work: Since the dahdi "service" really only modprobes the modules and runs dahdi_cfg, yes. And yes I did restart asterisk checked permissions, reloaded chan_dahdi.so, restarted asterisk again for good measure, etc. |
| 21:57.23 | [TK]D-Fender | PMantis: ; This is not intended to be a complete chan_dahdi.conf. Rather, it is intended; to be #include-d by /etc/chan_dahdi.conf that will include the global settings <-------- |
| 21:58.59 | PMantis | [TK]D-Fender: I knew it had to be something stupid. I had a late-night brain lapse, and symlinked chan_dahdi.conf to dahdi-channels.conf |
| 21:59.00 | PMantis | argh |
| 22:00.51 | *** part/#asterisk ruben23 (n=AGENT@122.55.48.243) |
| 22:02.10 | *** join/#asterisk DelphiWorld (n=Miranda@41.104.53.114) |
| 22:02.12 | DelphiWorld | hi |
| 22:02.23 | DelphiWorld | lol, the A2Billing developer is from algeria;à |
| 22:02.25 | DelphiWorld | ;) |
| 22:07.35 | *** join/#asterisk GameGamer43 (n=GameGame@h69-129-142-83.mdsnwi.tisp.static.tds.net) |
| 22:07.49 | *** part/#asterisk DelphiWorld (n=Miranda@41.104.53.114) |
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| 22:13.06 | *** join/#asterisk aidinb (n=Aidin@32.169.120.69) |
| 22:13.20 | Godfather_ | o/ |
| 22:13.35 | Kobaz | -- Added extension 's' priority 8 (CID match '') to invalidHandler (0x883ec18) |
| 22:13.38 | Kobaz | what's this cid match thing |
| 22:14.23 | [TK]D-Fender | Kobaz: How about showing us complete CLI output and cctual code |
| 22:14.36 | Kobaz | yeah, i can do that... but |
| 22:14.40 | Kobaz | this is something new |
| 22:15.04 | Kobaz | i was just wondering what cid match is, in dialplan now |
| 22:15.17 | Kobaz | this is trunk |
| 22:15.26 | Kobaz | it might be in 1.6.2, i'm not sure |
| 22:15.44 | [TK]D-Fender | Kobaz: CID matching is pre 1.0 |
| 22:16.03 | Kobaz | hmm |
| 22:16.08 | Kobaz | i never saw that in the dialplan reload output before |
| 22:16.15 | seanbright | the message is new |
| 22:16.21 | seanbright | the feature is not |
| 22:16.44 | Kobaz | http://pastebin.ca/1752897 |
| 22:17.54 | [TK]D-Fender | AEL.... nevermind |
| 22:17.57 | Kobaz | i thought maybe the cid match thing was my problem with the dialplan ael/extensions merging |
| 22:18.22 | [TK]D-Fender | Kobaz: Could be a bug in AEL interpretation |
| 22:18.30 | Kobaz | heh |
| 22:18.33 | [TK]D-Fender | Kobaz: one of the great reasons I have no intention of touching it. |
| 22:18.50 | Kobaz | well this is the entire ael file |
| 22:19.02 | Kobaz | there's a problem in trunk with merging extensions.conf and ael |
| 22:19.14 | [TK]D-Fender | Kobaz: Well..... whatever. The interpreter could be buggy even if your spec is not. |
| 22:19.14 | Kobaz | you have one or the other, it seems |
| 22:19.19 | Kobaz | could be |
| 22:20.35 | seanbright | though AEL has nothing to do with that message |
| 22:21.29 | Kobaz | yeah i didn't think so |
| 22:22.03 | Katty | throws things |
| 22:22.29 | p3nguin | kobaz: exten => 1234/5432,1,...... extension 1234, match CID 5432 |
| 22:23.01 | leifmadsen | I don't like that formatting |
| 22:23.09 | leifmadsen | so much easier to use CALLERID() function ot match |
| 22:23.19 | leifmadsen | makes no sense to me to add the matching on every line of the dialplan |
| 22:23.22 | Kobaz | p3nguin: oh okay |
| 22:23.28 | Kobaz | leifmadsen: yeah that's strange |
| 22:23.40 | Kobaz | https://issues.asterisk.org/view.php?id=16618 |
| 22:23.42 | Kobaz | well there's my bug |
| 22:23.47 | leifmadsen | I almost always try and get away from the matching as soon as possible and move to a static extension name |
| 22:23.58 | *** part/#asterisk PMantis (n=sswitzer@out.ewbc.com) |
| 22:24.14 | leifmadsen | save the value of ${EXTEN} to something like ${X} then Goto(some_static_extension,1) |
| 22:24.30 | Katty | goes through leifmadsen's pockets. |
| 22:24.37 | leifmadsen | it's too error prone to type the same complicated pattern match 100 times |
| 22:24.45 | leifmadsen | anyways, that's something to write into the book :) |
| 22:24.49 | leifmadsen | goes to help with dinner! |
| 22:25.07 | p3nguin | leifmadsen: It's not on every line of the dialplan. |
| 22:25.50 | p3nguin | leifmadsen: You duplicate the priority a couple times doing callerID matching, use a Goto() when it matches, then when it doesn't it proceeds down the rest of the dialplan. |
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| 22:37.19 | citywok | is there a way to find out when calls drop instead of just get hung up? |
| 22:41.35 | Kobaz | citywok: define the difference |
| 22:41.35 | citywok | when a call fails, like the packets stop flowing and the connection dies |
| 22:42.20 | citywok | or too many packets drop and it loses it |
| 22:42.25 | Kobaz | you'll get a hangup when the call goes away |
| 22:42.25 | citywok | but i dont have a way to tell if they hungup, or it died |
| 22:42.32 | Kobaz | what are you using to check |
| 22:42.33 | Kobaz | ami |
| 22:42.34 | Kobaz | ? |
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| 22:42.45 | citywok | currently nothing at all |
| 22:42.52 | citywok | since i dont know of any way to do it |
| 22:43.08 | Kobaz | ami would be a good way |
| 22:43.08 | citywok | if i could sit and watch the AMI i would (in code obviously) |
| 22:43.08 | Kobaz | listen for hangup events |
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| 22:43.44 | citywok | do you know if anything there will help identify? |
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| 22:47.38 | Kobaz | identify what? |
| 22:50.13 | citywok | clean vs unclean hangups? the whole did it drop, or was it acutally hung up |
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| 22:51.07 | Kobaz | there's the hangup codes |
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| 23:02.09 | timholum | Hello everyone |
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| 23:15.09 | dlynes | citywok, Try the following after telnetting to localhost 5038: |
| 23:15.09 | dlynes | citywok, http://pastebin.ca/1752955 |
| 23:16.57 | dlynes | citywok, make sure you have the username and password (login and secret) set up that are in the example, in your /etc/asterisk/manager.conf file |
| 23:16.57 | dlynes | Howdy tim |
| 23:17.25 | Erestar | Using the manager, if I send CoreShowChannels and then SipPeers, can I assume that it will send back all the events associated with CoreShowChannels before I get any response from SipPeers? |
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| 23:21.31 | timholum | does anyone have any experiance with voicemail in a mysql database? i keep getting the error Failed to obtain database object for 'asterisk'! |
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| 23:30.10 | citywok | dlynes: oh, that's helpful limiting it to not a million things |
| 23:32.20 | citywok | yea, i'm familiar with the ami |
| 23:32.33 | citywok | thanks! |
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| 23:33.55 | citywok | i'm getting quite a bit of this at my console, any ideas? http://pastebin.ca/1752964 |
| 23:36.24 | citywok | every once in a while it scrolls by for a minute |
| 23:36.27 | citywok | (hundreds of lines) |
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| 23:39.37 | timholum | has anyone successfully gotten asterisk to use mysql for voicemails, i keep getting Failed to obtain database object for 'asterisk'! |
| 23:40.20 | timholum | and I have only found 2 sites that show how to do it |
| 23:40.21 | timholum | neather of which are very detailed |
| 23:41.29 | Micc_ | timholum, I've used it before, but its a pain in the ass. when you type odbc show it should show if its connected. |
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| 23:44.37 | carrar | timholum, PostgreSQL works great for voicemail (and everything else) |
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| 23:45.07 | timholum | Micc_ it tells me its connected, and my extentions and sip accounts are all pulled from the same database |
| 23:47.09 | timholum | Micc_ I am guessing my issue has to do with eather a table name or structure, but I can not find any documentation on how that should be ( other then the two sources I found that that I can not get to work ) |
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| 23:48.11 | Micc_ | timholum, oh, well yeah it has to be setup according to the file that comes with asterisk. |
| 23:48.29 | Micc_ | timholum, there should be a txt in the docs section for mysql voicemail |
| 23:48.47 | Micc_ | If you used one from a website it could be old. |
| 23:48.48 | timholum | Ill have to look for that |
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