| 00:00.07 | *** join/#asterisk cmendes0101 (~cmendes01@wtnl.corp.tierra.net) |
| 00:04.57 | *** join/#asterisk robert___ (~hellspawn@objectx/robert) |
| 00:54.07 | *** part/#asterisk YoMomma (~YoMomma@71-95-164-90.static.mtpk.ca.charter.com) |
| 00:58.40 | *** join/#asterisk greenwolf (4257044e@gateway/web/freenode/ip.66.87.4.78) |
| 00:59.57 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 00:59.57 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 01:04.02 | *** join/#asterisk slicknick5181 (~slicknick@204.195.131.94) |
| 01:04.38 | slicknick5181 | I found a wake up call script online and I was wondering who could help me with a little caller id problem the code apparently has |
| 01:05.43 | slicknick5181 | http://pastebin.com/gZGkEaFF |
| 01:08.16 | [TK]D-Fender | slicknick5181: [Mar 19 21:00:08] WARNING[12809]: pbx.c:3877 func_args: Can't find trailing parenthesis for function 'CALLERID(nu'? <--- what do YOU think this should have referenced? |
| 01:08.32 | [TK]D-Fender | Looks a little short.... |
| 01:08.36 | slicknick5181 | Well I checked that Let me paste my code |
| 01:08.41 | [TK]D-Fender | "core show function CALLERID" |
| 01:08.45 | [TK]D-Fender | ^ |
| 01:09.56 | [TK]D-Fender | Take a close look at the syntax and your code... |
| 01:09.57 | slicknick5181 | [TK]D-Fender, I did that and it shows all the CALLERID info |
| 01:10.01 | *** join/#asterisk deo (~deo@222.127.13.226) |
| 01:10.14 | [TK]D-Fender | slicknick5181: Your CLI shows things have been chopped.... |
| 01:10.47 | slicknick5181 | http://pastebin.com/yTNmAsmT |
| 01:11.25 | slicknick5181 | [TK]D-Fender, It looked a little short so I checked the code to see if in fact I forgot the n in num and I didn't |
| 01:12.19 | [TK]D-Fender | slicknick5181: Looks like an escaping issue |
| 01:13.02 | slicknick5181 | [TK]D-Fender, what do you mean? |
| 01:13.29 | *** join/#asterisk deo (~deo@203.177.214.75) |
| 01:13.43 | [TK]D-Fender | that you have a ) in your function call and didn't escape it properly |
| 01:13.54 | [TK]D-Fender | System() |
| 01:16.19 | slicknick5181 | [TK]D-Fender, I couldn't find anything that would disable the correct capture of the caller id |
| 01:16.28 | *** join/#asterisk deo (~deo@122.53.72.218) |
| 01:20.36 | slicknick5181 | [TK]D-Fender, Maybe I'm miss understanding |
| 01:20.49 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 01:20.49 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 01:27.07 | *** join/#asterisk SeRi (~wtf@pdpc/supporter/professional/seri) |
| 01:27.51 | slicknick5181 | [TK]D-Fender, I'm very lost on what to do here all my other CALLERID settings I have in different parts of the dialplan work fine |
| 01:28.58 | [TK]D-Fender | Yes, but this is a system call and things need escaping |
| 01:29.09 | [TK]D-Fender | Anyway keep at it, I've got to head out for a bit |
| 01:29.17 | slicknick5181 | ok Thank you |
| 01:31.18 | *** join/#asterisk deo (~deo@222.127.13.226) |
| 01:33.05 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 01:33.56 | *** join/#asterisk deo (~deo@122.53.72.218) |
| 01:36.11 | *** join/#asterisk droemel (~droemel@p4FCAD4FE.dip.t-dialin.net) |
| 01:37.21 | *** join/#asterisk deo (~deo@222.127.13.226) |
| 01:42.13 | *** join/#asterisk deo (~deo@122.53.72.218) |
| 01:43.32 | *** join/#asterisk deo (~deo@222.127.13.226) |
| 01:43.36 | *** join/#asterisk Bradada (~Bradada@114-32-7-232.HINET-IP.hinet.net) |
| 01:43.54 | *** join/#asterisk deo (~deo@122.53.72.218) |
| 01:58.40 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 02:10.07 | poseidon | Anyone know of a php api which uses the asterisk http ami? I just want to grab simple information such as extensions on a call, how long, if it is inbound/outbound/transfer, etc |
| 02:39.49 | *** join/#asterisk bigDoggy (brads@2600:3c00::f03c:91ff:fe70:f0a9) |
| 02:40.08 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.33) |
| 02:49.06 | slicknick5181 | poseidon, This channel has much better times I've found mornings are better I am in estern time |
| 02:54.13 | poseidon | I am eastern time as well |
| 02:54.22 | poseidon | I just generally work at night because I have school in the morning |
| 02:58.54 | *** join/#asterisk LiuYan (~LiuYan@211.154.128.171) |
| 03:02.37 | *** join/#asterisk ghost75 (~trechber@dslb-088-066-171-177.pools.arcor-ip.net) |
| 03:03.53 | lorsungcu | sup yo |
| 03:05.54 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.55) |
| 03:13.12 | *** join/#asterisk Praise- (~Fat@unaffiliated/praise) |
| 03:13.35 | *** join/#asterisk JuStIcIa_ (~artur0@190.167.27.98) |
| 03:14.33 | *** join/#asterisk willryder (~tedryder@nc-184-3-103-138.dhcp.embarqhsd.net) |
| 03:15.05 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 03:17.02 | bigDoggy | is it easier to configure and IVR on freepbx via the gui? I am looking at the /extensions_additional.conf going 'yuck'. Never played with freepbx, only asterisk. |
| 03:17.27 | ChannelZ | "easier" is a relative point of view |
| 03:18.03 | ChannelZ | It's easier in that it's clicky-clicky, but perhaps not easier to make it do exactly what you want it to do. |
| 03:18.38 | bigDoggy | anything hancoded isn't going to necessarily reflect in the gui, will it..? |
| 03:19.07 | ChannelZ | dunno, I don't use FPBX |
| 03:19.48 | ChannelZ | I'll guess no. |
| 03:20.30 | slicknick5181 | If you start using the .conf files you must keep track of it yourself for the most part |
| 03:21.45 | bigDoggy | guis are the scurge of satan. |
| 03:41.24 | *** join/#asterisk apb1963_ (~apb1963@174.134.117.244) |
| 03:52.37 | *** join/#asterisk Bradada (~Bradada@59-120-137-34.HINET-IP.hinet.net) |
| 04:09.02 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 04:09.02 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 04:10.50 | *** join/#asterisk deo_ (~deo@112.198.79.53) |
| 04:19.54 | *** join/#asterisk FireAndIce (~FireAndIc@123.201.2.96) |
| 04:36.26 | *** join/#asterisk Bradada (~Bradada@59-120-137-34.HINET-IP.hinet.net) |
| 04:42.04 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 04:54.31 | *** join/#asterisk radic (~radic@dslb-178-002-209-159.pools.arcor-ip.net) |
| 05:01.10 | *** join/#asterisk willryder (~tedryder@nc-184-3-103-138.dhcp.embarqhsd.net) |
| 05:03.11 | *** join/#asterisk mwally (~mwally@mwally.opencnam.com) |
| 05:03.27 | drmessano | I'm playing around with using the iptables recent module to (possibly) limit sip attacks.. Found a forum post on it, but I am concerned about only matching based on IP address and not IP source port |
| 05:03.38 | drmessano | Any IPTABLES gurus around? |
| 05:06.35 | ChannelZ | Not sure about guru but hit me |
| 05:07.04 | andross | well requests could come from any source port |
| 05:07.54 | ChannelZ | yeah don't you mean destination port (you)? |
| 05:08.59 | drmessano | My concern is that, suppose I set a hitcount of -recheck -second 60 and a hitcount of 5. If I have 5 phones with multiple extensions that decide to reregister after the internet has been down for 2 hours at a remote location. Won't that be much greater than 5 from 1 IP? |
| 05:09.26 | drmessano | (If we're purely going by IP and not the source port) |
| 05:11.26 | carrar | You could block on file not found too |
| 05:11.43 | carrar | or things like that |
| 05:11.53 | carrar | if they are polling their configs |
| 05:12.05 | ChannelZ | well in that case yes I suppose you'd have to hardcode all the source ports to differentiate them, assuming they don't change. But if you know their IP, why not just whitelist them completely? |
| 05:12.17 | andross | seems weird to filter traffic from a location that you have control over tho |
| 05:12.36 | andross | like why not just allow whatever sip/rtp from that ip |
| 05:12.40 | ChannelZ | I mean you either want to trust them or you don't.. or perhaps your case is more subtle. |
| 05:12.59 | carrar | softphones can come from anywhere |
| 05:13.24 | andross | softphones from abroad are always an issue |
| 05:13.53 | drmessano | I can't whitelist every location where there may be multiple endpoints due to dynamic IP addresses, and users do roam. |
| 05:13.58 | carrar | If they register on 1st try have to give them the benfite of the doubt |
| 05:14.08 | carrar | unless you know they are just are false |
| 05:15.00 | drmessano | But this hitcount thing seems flawed if I am limited to 20, and have to make the window so wide as to account for multiple legit devices hitting the box from one IP too fast |
| 05:16.39 | carrar | Could just do it, then exclude blocks that are issues but is real traffic :) |
| 05:17.20 | carrar | or just not bother with it |
| 05:17.25 | ChannelZ | if the intention is to try and throttle, let them fail and try again. |
| 05:17.42 | drmessano | I thought about that too |
| 05:18.07 | ChannelZ | Like if your limit is 5 and you get 6 requests, one guy loses but more than likely will try again after a timeout. |
| 05:18.16 | drmessano | Ah |
| 05:18.28 | ChannelZ | Minor inconvenience I suppose |
| 05:18.32 | carrar | unless it's cronic |
| 05:18.39 | drmessano | I think thats the part I missed.. I was hung up on "But wont they all.. " |
| 05:18.40 | carrar | and then hits random users |
| 05:18.46 | drmessano | No, they wont all.. just the missed ones |
| 05:18.54 | ChannelZ | Where it gets squirrely is if they all start making calls at once, you have a rapid exchange of individual SIP packets for the call setup |
| 05:19.04 | drmessano | Yeah |
| 05:19.22 | carrar | What is the issue? |
| 05:19.27 | ChannelZ | So there's really no good means to deal with it just using the packet limit thing. |
| 05:19.29 | carrar | that you are trying to solve |
| 05:19.37 | ChannelZ | You'll eventually get false positives |
| 05:19.51 | carrar | Is there a real probably happening? |
| 05:19.55 | carrar | problem |
| 05:20.52 | carrar | or just playing? |
| 05:20.52 | drmessano | I have a couple boxes that are getting murdered by SIP attacks... I can reproduce it with SIPp pretty easily.. Box gets exhausted, Asterisk crashes, restarts |
| 05:20.54 | ChannelZ | Most of the BS traffic I ever get is drive-by registration tries, or trying to place calls.. I trap all that mess with fail2ban and then just block that IP |
| 05:21.26 | carrar | so you get lots of failed attempts on your console/logs? |
| 05:21.39 | carrar | with a IP in there someplace? |
| 05:21.48 | drmessano | Yeah, and they're like 30+ calls a second and BOOM |
| 05:22.06 | carrar | So what about autofwd or fail2ban? |
| 05:22.28 | carrar | You can only take in 1 second and your box explodes? |
| 05:22.41 | drmessano | I thought about fail2ban, but I keep reading more and more about how under an attack scenario that Asterisk dies before fail2ban even has a chance to react |
| 05:22.46 | ChannelZ | yeah that seems to be not-good |
| 05:22.48 | drmessano | Not 1 second |
| 05:23.01 | drmessano | 30+ calls a second over some period of time |
| 05:23.12 | drmessano | 20 seconds? |
| 05:23.27 | carrar | autofwd works great |
| 05:23.46 | carrar | assuming there is a IP in the log |
| 05:24.02 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.42) |
| 05:24.06 | drmessano | I am checking out autofwd |
| 05:24.28 | drmessano | I guess there's another angle too |
| 05:24.40 | carrar | rate limiting seems like a bad idea |
| 05:25.50 | ChannelZ | yeah it's just too generic in your case |
| 05:26.22 | drmessano | That's where I was stuck.. I couldn't make sense of a way to implement it and not screw something legitimate |
| 05:26.40 | carrar | beefire box might help also to extend the amount of time you can take a hit for something to kick in and block it |
| 05:26.45 | carrar | err beefier |
| 05:27.41 | ChannelZ | I'm more curious why/how it can kill * |
| 05:27.42 | drmessano | It's a VM.. and I guess I could pay for something bigger, but it seems kinda silly if I am doing it just to stave off attacks, when I don't need the resources otherwise |
| 05:27.47 | carrar | yeah |
| 05:27.51 | carrar | resource issues |
| 05:28.15 | carrar | so it's not doign any RTP? |
| 05:28.38 | carrar | or is it |
| 05:28.40 | carrar | must be |
| 05:28.43 | drmessano | You mean at night when the bots come? |
| 05:28.50 | carrar | heh |
| 05:29.05 | carrar | THEY COME IN THE COVER OF NIGHT |
| 05:29.26 | andross | what if you had softphone users with dynamic ips vpn in first |
| 05:29.32 | ChannelZ | Well yea - are these legit calls (do you accept anonymous?) or are they just lots of requests that get rejected? |
| 05:30.11 | carrar | I think your vm is undersized |
| 05:30.28 | drmessano | I do accept anonymous. I could shut that off, but that kinda pisses me off... |
| 05:31.27 | drmessano | Maybe so |
| 05:31.31 | ChannelZ | so these calls do connect? |
| 05:31.45 | ChannelZ | ls |
| 05:31.47 | ChannelZ | oops |
| 05:31.58 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-byfmhjedojdcgkga) |
| 05:32.11 | ChannelZ | What's your test args with sipp? I wanna see if I can kill mine! |
| 05:32.19 | drmessano | I drop the ones to non-existant extensions with a hangup |
| 05:32.37 | drmessano | I just fire it up and keep hitting + until I get the BOOM email from monit |
| 05:33.01 | drmessano | (+ increases the concurrent calls) |
| 05:33.36 | ChannelZ | yeah but what do I tell this thing to start.. just the host/ip? |
| 05:33.43 | drmessano | Correct |
| 05:34.07 | drmessano | 'sipp ipaddress' |
| 05:34.40 | drmessano | There's TONS of arguments and tweaks, but you can get it running without going nuts on the switches |
| 05:35.56 | ChannelZ | hmm. I have this going like a bat out of hell but nothing is breaking. But I don't have a catchall extension, I just reject them.. |
| 05:36.06 | ChannelZ | "Call from '' (192.168.1.1:5061) to extension 'service' rejected because extension not found" etc |
| 05:54.11 | *** join/#asterisk Bradada (~Bradada@114-32-7-232.HINET-IP.hinet.net) |
| 05:56.51 | *** join/#asterisk Bradada (~Bradada@114-32-7-232.HINET-IP.hinet.net) |
| 06:09.32 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 06:27.37 | *** join/#asterisk evilman_work (~evilman@87.244.6.228) |
| 06:27.51 | *** join/#asterisk Bradada (~Bradada@114-32-7-232.HINET-IP.hinet.net) |
| 06:28.37 | drmessano | carrar: |
| 06:33.53 | *** join/#asterisk bigDoggy (brads@2600:3c00::f03c:91ff:fe70:f0a9) |
| 06:35.32 | bigDoggy | I am using freepbx and I can dial out and I can see/hear the inbound call hit asterisk but cannot get it into my ivr. ?? http://pastie.org/6635880 |
| 06:45.03 | *** join/#asterisk fling (~fling@fsf/member/fling) |
| 06:47.43 | WIMPy | Try in #freepbx. It's not supported here. |
| 06:49.26 | *** join/#asterisk apb1963__ (~apb1963@174.134.117.244) |
| 06:49.37 | *** part/#asterisk bigDoggy (brads@2600:3c00::f03c:91ff:fe70:f0a9) |
| 07:01.44 | *** join/#asterisk v0lZy (~Thunderbi@mail.silk-group.net) |
| 07:08.44 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 07:12.16 | *** join/#asterisk santa0536 (~santa@cn-bgp-nat.portaone.com) |
| 07:15.04 | *** join/#asterisk deo_ (~deo@203.177.214.75) |
| 07:17.45 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
| 07:21.05 | *** join/#asterisk mihamina (~mihamina@ip-41-190-237-66.orange.mg) |
| 07:25.17 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 07:25.17 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 07:39.27 | *** join/#asterisk HmdP_Mobile (~HmdP_Mobi@D9799130.cm-3-2c.dynamic.ziggo.nl) |
| 07:42.16 | *** join/#asterisk Rac-on (jasper@bambi.rac-on.nl) |
| 07:42.38 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 07:44.04 | *** join/#asterisk mirela666 (~Thunderbi@212.200.146.253) |
| 07:48.03 | *** join/#asterisk hebber (~hebber@node-14pg.pool-125-25.dynamic.totbb.net) |
| 07:49.03 | hebber | Hi, I have a nut to crack - when I transfer using features, I loose track of peername and accountname. What variable can I use to find the peername and accountname after a transfer? |
| 07:53.41 | *** join/#asterisk n8ideas (~joshelson@216.241.45.82) |
| 07:54.11 | n8ideas | Gah... freaking disaster tonight. Can anyone tell me if "qualifygap" actually does what it's supposed to? |
| 08:00.20 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 08:04.04 | *** join/#asterisk soFh (~soFh@182.186.85.206) |
| 08:05.12 | *** part/#asterisk soFh (~soFh@182.186.85.206) |
| 08:11.50 | *** join/#asterisk Sorcier_FXK (~nssystem@unaffiliated/sorcierfxk) |
| 08:13.21 | jzaw | google-fu failing me |
| 08:13.28 | jzaw | can anyone point me to some info on |
| 08:13.31 | jzaw | [Mar 20 05:57:15] WARNING[13118]: chan_sip.c:17900 check_via: Could not resolve socket address for '`��ī :47298' |
| 08:13.32 | jzaw | [Mar 20 05:57:19] ERROR[13118]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo("`��ī ", "47298", ...): Name or service not known |
| 08:14.20 | wdoekes | jzaw: someone is sending you crap in the via header |
| 08:14.45 | wdoekes | I suspect a broken ALG |
| 08:14.55 | wdoekes | sip set debug on |
| 08:15.14 | deo | hi guys need some advice.. is it possible in asterisk to generate cdr automatically based on extension? is there some module to do that? thanks |
| 08:15.46 | wdoekes | deo: ? cdrs are generated for calls.. |
| 08:16.00 | deo | yes wdoekes .. |
| 08:16.31 | deo | what i mean is that is it possible to generate a csv file from the cdr per extension? |
| 08:16.41 | wdoekes | if you want them grouped by *calling* device, set an accountcode |
| 08:16.49 | deo | what i can see in the /var/log/asterisk/cdr-XXX/Master.csv |
| 08:17.09 | deo | is the whole Master.csv.. |
| 08:17.26 | deo | where to set that w? |
| 08:17.29 | deo | wdoekes: ? |
| 08:17.42 | wdoekes | if you want them grouped by dst (extension, *not* device), you'll have to search through the file and split it by dst |
| 08:17.49 | *** join/#asterisk Sorcier_FXK (~nssystem@unaffiliated/sorcierfxk) |
| 08:18.08 | wdoekes | accountlogs=yes ; create separate log file for each account code. Default is "yes" |
| 08:19.14 | wdoekes | you'll have to set up accountcode's for each device (e.g. accountcode=deo in sip.conf) |
| 08:19.30 | deo | you mean per extension wdoekes ? |
| 08:19.35 | wdoekes | NO |
| 08:19.40 | wdoekes | extension != device |
| 08:19.46 | deo | sorry |
| 08:20.04 | wdoekes | you need to know the difference |
| 08:20.41 | *** join/#asterisk areski (~areski@80.174.128.6.dyn.user.ono.com) |
| 08:20.56 | wdoekes | you have devices (sip accounts, etc..), which can call |
| 08:21.05 | wdoekes | you have extensions (points in your dialplan), which can be called |
| 08:21.18 | wdoekes | from the dialplan, you can call the devices |
| 08:22.44 | wdoekes | but you can do other stuff in the dialplan too, like go-to VoiceMail, for which you'd also get a cdr: for the 'dst' extension which was called |
| 08:23.19 | bulkorok | deo: you can handle CDR via databases... |
| 08:23.37 | deo | bulkorok: by scripting ? |
| 08:23.48 | bulkorok | there are modules I think |
| 08:23.58 | deo | thats what im planning to do, using php to query the db |
| 08:24.07 | deo | bulkorok: i hope there are modules |
| 08:24.16 | bulkorok | ew... no don't do that... it's invented already... |
| 08:24.17 | jzaw | thanks wdoekes ... looking now |
| 08:24.41 | bulkorok | I'm using one... let me check |
| 08:24.49 | hebber | deo: check cdr_adaptive_odbc.conf |
| 08:24.55 | deo | thanks bulkorok |
| 08:25.03 | deo | hebber: thanks will do |
| 08:25.16 | bulkorok | right... thats waht I use with mysql... |
| 08:25.52 | deo | bulkorok: what module is it/ |
| 08:25.54 | deo | ? |
| 08:26.14 | bulkorok | cdr_adaptive_odbc.so |
| 08:26.31 | deo | hmmnn i dont quite get it hehe |
| 08:26.43 | *** join/#asterisk classix (~salven@silenceisdefeat.com) |
| 08:27.04 | deo | hebber: do i dont find any cdr_adaptive_odbc.conf |
| 08:27.18 | deo | am i lacking in module to be recompile? :( |
| 08:27.28 | bulkorok | check menuselect... |
| 08:27.42 | wdoekes | install unixodbc-dev |
| 08:28.26 | hebber | deo: it was a pain to install odbc, but when I got it right the adaptive cdr works great |
| 08:29.06 | deo | hmmn ok hebber will try to check it |
| 08:29.07 | bulkorok | right... in menuselect under "Call Detail Recording" (surprise!) |
| 08:29.10 | *** join/#asterisk FireAndIce (~FireAndIc@123.201.2.96) |
| 08:29.16 | bulkorok | but depends on res_odbc |
| 08:29.18 | deo | bulkorok: ? |
| 08:29.25 | deo | ahhhh |
| 08:29.44 | bulkorok | and res_odbc Depends on: generic_odbc(E), ltdl(E) |
| 08:29.47 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 08:29.52 | wdoekes | what hebber said: make sure you get isql (command line tool) to access your database before continuing |
| 08:29.58 | *** join/#asterisk Wiretap (~wiretap@unaffiliated/wiretap) |
| 08:37.58 | hebber | deo: make sure mysql connection work, then install odbc, make sure it works, then menuselect and compile |
| 08:38.28 | deo | hebber: wdoekes bulkorok thanks for all your suggestions.. i will try that.. thanks ;) |
| 08:39.58 | *** join/#asterisk threesome (~threesome@customer-79-127-150-148.net.angelnet.cz) |
| 08:40.04 | *** join/#asterisk ujjain (ujjain@unaffiliated/ujjain) |
| 08:40.45 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 08:45.30 | *** join/#asterisk threesome (~threesome@customer-79-127-150-148.net.angelnet.cz) |
| 08:54.29 | *** join/#asterisk mwally (~mwally@mwally.opencnam.com) |
| 08:55.20 | *** join/#asterisk rolandow (~roland@s559700f1.adsl.online.nl) |
| 09:04.14 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 09:16.37 | *** join/#asterisk tamiel (~tamiel@197.158.64.3) |
| 09:28.40 | *** join/#asterisk HmdP_Mobile (~HmdP_Mobi@D9799130.cm-3-2c.dynamic.ziggo.nl) |
| 09:37.30 | *** join/#asterisk sedeki (~textual@unaffiliated/sedeki) |
| 09:40.42 | hebber | Hi, does anyone have experience of atxfer is working on only SIP to SIP, but not from a DADHI call? |
| 09:41.01 | hebber | the atxfer is defined in features.conf |
| 09:46.32 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 09:49.19 | *** join/#asterisk bogrd__ (~asteriskp@115.119.115.26) |
| 09:54.24 | bogrd__ | Hello, I am trying to integrate the already setup asterisk system with the hardware to handle calls from hardphones, I already have the card. Please let me know if this card here can serve the purpose for me. http://www.x100p.com/products/FXO.php |
| 09:58.59 | kaldemar | bogrd__: http://www.x100p.com/voip/fxo_fxs.php |
| 09:59.17 | kaldemar | what kind of hardphones? |
| 10:00.06 | bogrd__ | kaldemar: VOIP |
| 10:00.41 | kaldemar | you don't need any telephony hardware to use voip phones. |
| 10:00.55 | kaldemar | are you trying to connect your asterisk box to an analog line? |
| 10:01.39 | bogrd__ | I am getting a PRI line so that it can handle calls to PSTN from SIP phones |
| 10:02.29 | kaldemar | then you need PRI hardware. analog interfaces (FXO or FXS) won't do you any good. |
| 10:02.53 | kaldemar | there's also the possibility to use an ITSP to connect to PSTN. |
| 10:03.46 | bogrd__ | Since that I already have this hardware and PRI line, I am experimenting on them |
| 10:04.29 | bogrd__ | Can I use this hardware to connect to a single channel normal telephone line ? |
| 10:05.21 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 10:05.42 | *** join/#asterisk areski (~areski@81.184.35.151.dyn.user.ono.com) |
| 10:06.16 | *** join/#asterisk HmdP_Mobile (~HmdP_Mobi@D9799130.cm-3-2c.dynamic.ziggo.nl) |
| 10:06.27 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 10:07.54 | kaldemar | bogrd__: "normal" as in analog? FXO is used for that, yes. |
| 10:08.06 | bogrd__ | kaldemar: yes analog |
| 10:08.51 | bogrd__ | kaldemar: so can this card serve my purpose, Im using astersisk 11.2.1 ? |
| 10:10.27 | kaldemar | which purpose? |
| 10:11.16 | bogrd__ | I should be able to make and receive calls from analog phones |
| 10:11.32 | kaldemar | no. FXO is for connecting to a line, not a phone. |
| 10:12.12 | kaldemar | ~book |
| 10:12.12 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 10:14.15 | bogrd__ | kaldemar: ok, Can make a call to analog phones using VOIP phones connected to asterisk ? |
| 10:15.52 | kaldemar | not with FXO. |
| 10:16.54 | kaldemar | you can not connect analog phones to an FXO interface. |
| 10:17.49 | *** join/#asterisk sedeki (~textual@unaffiliated/sedeki) |
| 10:17.55 | bogrd__ | kaldemar: ya.. i mean.. i'll be using a sip softphone to connect to asterisk and then call a analog phone via the telephone line connected to FXO port of asterisk.. this is possible right? |
| 10:19.20 | kaldemar | of course. it does not matter what the phone behind PSTN is, you only interface with the phone line. |
| 10:22.08 | bogrd__ | The product specification says that it is compatible with Zaptel drivers. i read that Zaptel is renamed to DAHDI sometime back. As I am using Asterisk 11.2.1, are Zaptel drivers comptible yet ? |
| 10:23.21 | kaldemar | zaptel is dead. use DAHDI. |
| 10:24.01 | bogrd__ | is this hardware comptible with DAHDI ? |
| 10:24.25 | *** join/#asterisk Phoebus_ (~Phoebus@96.127.133.54) |
| 10:26.41 | kaldemar | bogrd__: ask the manufacturer. the zaptel->dahdi transision happened over 4 years ago. |
| 10:27.54 | bogrd__ | kaldemar: thank you .. il enquire the manufacturer about it .. |
| 10:30.45 | kaldemar | if you already have the card, you could just try it yourself. the wcfxo driver in DAHDI should be compatible. |
| 10:32.06 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 10:33.01 | *** join/#asterisk Phoebus (~Phoebus@pdpc/supporter/active/phoebus) |
| 10:35.13 | *** join/#asterisk rhce7320 (~rhce7320@59.167.200.141) |
| 10:38.07 | *** join/#asterisk Quintana (~sylvain@aghnar.doowan.net) |
| 10:40.21 | bogrd__ | kaldemar: il try that .. |
| 10:57.26 | *** join/#asterisk evilsk4ter (~evilsk4te@187.60.66.11) |
| 10:59.54 | *** join/#asterisk Boichev (~Boichev@31.13.221.124) |
| 11:14.51 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 11:14.51 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 11:17.10 | sedeki | when I call from my softphone to my cell phone, i cannot hear anything in my softphone but i can hear in my cell phone |
| 11:18.07 | kaldemar | did you check the rtp debug? |
| 11:18.20 | sedeki | how do i check the rtp debug? |
| 11:18.52 | kaldemar | "rtp set debug on", make a call and check if rtp packets are coming from your provider and going to your softphone. |
| 11:20.08 | *** join/#asterisk deo_ (~deo@112.198.90.167) |
| 11:20.51 | sedeki | I get lots of packets like this |
| 11:20.52 | sedeki | Got RTP packet from 193.105.226.106:64862 (type 08, seq 000500, ts 080000, len 000160) |
| 11:20.53 | sedeki | Sent RTP packet to 178.78.205.116:8000 (type 03, seq 004840, ts 080000, len 000033) |
| 11:22.52 | sedeki | I guess that means that packets are coming to my softphone |
| 11:30.01 | kaldemar | sedeki: make sure a firewall doesn't block those on the softphone computer. |
| 11:30.20 | sedeki | should I allow several codecs or just one? |
| 11:31.36 | *** join/#asterisk vlad_starkov (~vlad_star@83.220.237.239) |
| 11:32.11 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 11:38.03 | *** join/#asterisk vlad_sta_ (~vlad_star@83.220.237.239) |
| 11:38.51 | *** join/#asterisk HmdP_Mobile (~HmdP_Mobi@D9799130.cm-3-2c.dynamic.ziggo.nl) |
| 11:42.21 | kaldemar | sedeki: that's up to you to decide. |
| 11:42.35 | sedeki | okay got it working by changing my softphone settings. |
| 11:56.41 | sedeki | kaldemar have you used starpy? |
| 11:59.01 | *** join/#asterisk mihamina (~mihamina@41.188.46.121) |
| 12:01.00 | kaldemar | sedeki: no. |
| 12:07.24 | *** join/#asterisk Ickmund (~Ickmund@cli-5b7e85ed.bcn.adamo.es) |
| 12:08.46 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 12:13.23 | Ickmund | I have a fresh installation of 11.2.1 where I'm not getting any console info. done core set verbose 10, logger show channels gives me 'Console Enabled - DEBUG NOTICE WARNING ERROR VERBOSE' |
| 12:14.20 | Ickmund | I see nothing. No registrations, no verbose messages, nothing from extensions, nothing. What can I be missing here? |
| 12:15.32 | Greenlight | Ickmund: Try starting it directly "asterisk -c" and look for any errors when loading up |
| 12:16.00 | *** join/#asterisk classix (~salven@silenceisdefeat.com) |
| 12:18.04 | Ickmund | Greenlight: no errors, but it started working... forgot the old reboot there... |
| 12:24.38 | *** join/#asterisk FireAndIce (~FireAndIc@123.201.2.88) |
| 12:28.17 | *** join/#asterisk kleszcz (~tick@linuxmafia.pl) |
| 12:30.22 | *** join/#asterisk elico (~Thunderbi@bzq-79-176-184-165.red.bezeqint.net) |
| 12:31.33 | *** join/#asterisk Sorcier_FXK (~nssystem@unaffiliated/sorcierfxk) |
| 12:34.12 | *** join/#asterisk [TK]D-Fender (~Joe@216-191-106-165.dedicated.allstream.net) |
| 12:43.25 | jmetro | dat netsplit |
| 12:47.02 | *** join/#asterisk Chotaire (chotaire@chotaire-home.vipri.net) |
| 12:53.45 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 12:55.09 | *** join/#asterisk serafie (~erin@nat/digium/x-ufkqkztkkjpsyhvb) |
| 13:03.17 | *** join/#asterisk mjordan (~mjordan@wsip-184-179-108-66.sd.sd.cox.net) |
| 13:03.18 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:04.27 | *** join/#asterisk jkroon (~jkroon@dsl-244-32-225.telkomadsl.co.za) |
| 13:05.07 | *** join/#asterisk apurvtwr (b64722fb@gateway/web/freenode/ip.182.71.34.251) |
| 13:13.21 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:13.21 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:19.23 | *** join/#asterisk vlad_starkov (~vlad_star@83.220.237.239) |
| 13:39.05 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 13:39.06 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 13:40.12 | eXcAliBuR | yay |
| 13:47.43 | malcolmd | ahh, morning |
| 13:50.09 | *** join/#asterisk blee (~blee@68.204.217.123) |
| 13:52.51 | *** join/#asterisk Sorcier_FXK (~nssystem@unaffiliated/sorcierfxk) |
| 13:58.12 | eXcAliBuR | i'm cold, does anyone have a blankey ? |
| 13:58.47 | eXcAliBuR | i was rather rude to digium support last night :( |
| 13:58.53 | eXcAliBuR | they referred me to your site |
| 13:59.05 | eXcAliBuR | I don't think they even read my question |
| 13:59.08 | *** join/#asterisk alami (~alami@unaffiliated/alami) |
| 13:59.31 | eXcAliBuR | I want to have a line key directly connected to an analog line |
| 14:04.18 | malcolmd | slastation, slatrunk. good luck, support doesn't assist with that. also, there's no immediate dial capability, so the phone won't take the analog line off-hook when the handset is raised or the handsfree engaged. |
| 14:05.58 | eXcAliBuR | i c |
| 14:07.17 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 14:09.01 | eXcAliBuR | can the line keys show when an analog line is busy? |
| 14:09.32 | *** part/#asterisk mjordan (~mjordan@wsip-184-179-108-66.sd.sd.cox.net) |
| 14:11.46 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 14:11.46 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 14:13.18 | *** join/#asterisk corretico (~luis@190.211.93.38) |
| 14:17.22 | [TK]D-Fender | eXcAliBuR, Yes. |
| 14:17.41 | [TK]D-Fender | eXcAliBuR, However forget real SLA. * does not support that. |
| 14:17.58 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 14:17.58 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 14:18.20 | *** join/#asterisk FireAndIce (~FireAndIc@123.201.2.88) |
| 14:19.33 | *** join/#asterisk FireAndIce (~FireAndIc@123.201.2.88) |
| 14:21.43 | AlexForster | so, I tried this question last night, I'm going to try it one more time... |
| 14:21.50 | AlexForster | developers: other than call transfers, what are the scenarios where asterisk will masquerade a channel? I'm pushing a few thousand calls a day through an instance of asterisk 1.8, and 3-4 times each day, for only a few specific people, asterisk ends both legs of the call immediately after masquerading a channel as "AsyncGoto/{channel}<ZOMBIE>" |
| 14:22.00 | AlexForster | I'm running a mostly stock install of FreePBX 2.10, and I don't have a single instance of AsyncGoto in my dialplan |
| 14:22.11 | AlexForster | alternatively, does anyone have any tips on debugging this (keeping in mind that it's a prod system with high volume, so I'm trying to avoid just "logging everything")? the SIP trace doesn't give any helpful info, nor do any manager events I've seen, though I'm not subscribed to any diaplan application events |
| 14:22.36 | Boichev | from where can I see all $agi->request functions ? |
| 14:25.11 | eXcAliBuR | when i dial a 9 digit number and press dial, it works, but when i lift the handset and start to dial, after the 6 number i get busy signal and asterisk says unknown extension. |
| 14:25.31 | *** join/#asterisk leedm777 (~leedm777@nat/digium/x-vldotsqyjfhrmznt) |
| 14:25.40 | [TK]D-Fender | eXcAliBuR, Fix your phone's dialplan. |
| 14:25.40 | eXcAliBuR | why won't it let me finish dialing? |
| 14:26.02 | eXcAliBuR | would it be the digit_map ? |
| 14:26.22 | [TK]D-Fender | eXcAliBuR, What its name is, yes |
| 14:29.46 | *** join/#asterisk thecardsmith (~quassel@pdpc/supporter/student/thecardsmith) |
| 14:35.13 | eXcAliBuR | _NXXNXXXXXX why is there a second N ? and not a X ? |
| 14:35.43 | Boichev | in asterisk 1.8 how to get to the call lenght with AGI |
| 14:36.03 | igcewieling | because you can't have 212-055-1212 that is why |
| 14:36.21 | igcewieling | Boichev: same way as previous versions |
| 14:36.55 | igcewieling | eXcAliBuR: also that does not look like a phone digit map, that looks like an Asterisk extensions pattern. |
| 14:37.11 | eXcAliBuR | yuh, i think mine is broken |
| 14:37.16 | Boichev | igcewieling, I don't know how to get it in previous versions ... I added 1.8 because I saw in the manual that DeadAGI is depricated since 1.6 |
| 14:38.23 | igcewieling | Boichev: just before you run your AGI, run "dumpchan" that will show you all the channel variables, assuming the call was answered and ended somewhere else in the dialplan you'll see DIALEDTIME and ANSWEREDTIME or similar variables. |
| 14:38.38 | igcewieling | you can get those variables in your AGI |
| 14:38.39 | [TK]D-Fender | <eXcAliBuR> when i dial a 9 digit number and press dial, it works, but when i lift the handset and start to dial, after the 6 number i get busy signal and asterisk says unknown extension. <--- NXXNXXXXXX is a NANPA format. This does not match the NINE digits you said you were trying to dial. |
| 14:39.06 | [TK]D-Fender | Boichev, It's in CDR() as always |
| 14:40.03 | kaldemar | Boichev: find out how to read variables/functions from AGI and then look at fields duration and billsec in func CDR. |
| 14:41.53 | eXcAliBuR | this is my digit map ... there is something wrong wit it too, I just can't figure out what. digit_map=[0-8]xxxxx|911|9911|411|0|9411|9611|7777|9011xxx.T3|91xxxxxxxxxx|9[2-9]xxxxxx|*xxx.T3 |
| 14:42.04 | Boichev | kaldemar, [TK]D-Fender igcewieling I got it running with $agi->request['agi_uniqueid']; but don't have any idea how to access the CDR |
| 14:42.13 | eXcAliBuR | i'm gonna add more xx's |
| 14:42.20 | [TK]D-Fender | Boichev, "core show function CDR" <--- |
| 14:42.38 | igcewieling | Boichev: uniqueID is not the call length |
| 14:42.39 | kaldemar | Boichev: find out how to read variables with the library/framework you're using for AGI. |
| 14:42.57 | *** join/#asterisk ujjain (ujjain@unaffiliated/ujjain) |
| 14:43.11 | [TK]D-Fender | eXcAliBuR, So far I still don't see a pattern for 9 digits, or an explanation of where that is considered valid |
| 14:43.16 | Boichev | igcewieling, uniqueID was an example |
| 14:44.24 | igcewieling | Boichev: you can only get a VERY limited number of items from the request, best to do a $agi->get_variable("CDR(billsec)") like [TK]D-Fender said. |
| 14:44.43 | igcewieling | not knowing anything about your language or agi lib I can't be more specific. |
| 14:44.45 | eXcAliBuR | guess defaults isn't always good |
| 14:45.04 | igcewieling | eXcAliBuR: this is telecom, the defaults are NEVER good. |
| 14:45.09 | [TK]D-Fender | eXcAliBuR, Correct |
| 14:45.21 | igcewieling | eXcAliBuR: what country are you in? |
| 14:45.37 | eXcAliBuR | yay it's working now |
| 14:45.50 | eXcAliBuR | you won't judge me? |
| 14:45.52 | [TK]D-Fender | eXcAliBuR, digit_map=[0-8]xxxxx|911|9911|411|0|9411|9611|7777|9011xxx.T3|91xxxxxxxxxx|9[2-9]xxxxxx|*xxx.T3 <- not default for anyone... |
| 14:45.54 | eXcAliBuR | Canada |
| 14:46.07 | [TK]D-Fender | eXcAliBuR, Clearly modded.... |
| 14:46.24 | [TK]D-Fender | eXcAliBuR, And 1980 is calling .. they want their "Dial 9 prefix" back :p |
| 14:46.41 | eXcAliBuR | yuh i deleted it |
| 14:46.54 | igcewieling | eXcAliBuR: Ah, you must not be calling the PSTN |
| 14:47.37 | igcewieling | *sigh* If it wasn't for customers, this job would be much more fun. |
| 14:49.03 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 14:54.21 | Boichev | igcewieling, [TK]D-Fender kaldemar I have big problems keeping the script running after the call here is my dialplan and agi script http://pastebin.com/8wj7muXc |
| 14:55.40 | *** join/#asterisk domi (~Thunderbi@mail.tas.de) |
| 14:55.55 | igcewieling | Boichev: I don't see the "g" option on Dial. Also your AGI will only execute (assuming you have g on Dial) when the DESTINATION hangs up first. When the SOURCE hangs up first then exten => h is run instead. |
| 14:56.40 | Boichev | igcewieling, it executes and gives me the extension and the called number |
| 14:57.14 | [TK]D-Fender | igcewieling, Commented out. there is no dial |
| 14:57.22 | [TK]D-Fender | (at the front |
| 14:57.47 | [TK]D-Fender | Boichev, And you aren't showing us the call so we can what is actually happening. |
| 14:58.06 | igcewieling | Boichev: what excactly is the issue? |
| 14:58.23 | igcewieling | you know get_variable returns an array, right? |
| 14:59.01 | kaldemar | Boichev: why are you expecting your AGI to be running during the call? |
| 15:00.13 | Boichev | kaldemar, as I understand to get the call lenght I have to request it after the call has ended ( after the SIGHUP signal) |
| 15:00.18 | kaldemar | and what is the point of using agi for writing that data to a file? |
| 15:00.50 | [TK]D-Fender | Boichev, Your script is not just sitting there while you dial. |
| 15:00.56 | [TK]D-Fender | Boichev, * call processing is LINEAR |
| 15:01.05 | [TK]D-Fender | Boichev, You don't seem to understand the flow yet |
| 15:01.12 | [TK]D-Fender | ~book |
| 15:01.12 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 15:01.13 | [TK]D-Fender | ^ |
| 15:01.33 | Boichev | [TK]D-Fender, sorry first think with dialplans |
| 15:01.35 | kaldemar | Boichev: when dialplan enters the Dial, your agi has already exited. |
| 15:02.11 | Boichev | [TK]D-Fender, after the hangup I can't call the AGI ... .what is the way to do get info after the call has ended |
| 15:02.41 | [TK]D-Fender | Boichev, Start by reading the book. So far what you look like you're trying to do in your AGI isn't anything you can't do straight in the dialplan without it. |
| 15:03.05 | [TK]D-Fender | Boichev, the "h" Asterisk Standard Extension <---- you've been told this already |
| 15:03.17 | [TK]D-Fender | Boichev, Read your basics. You are skipping the important bits |
| 15:03.19 | *** join/#asterisk AkkerKid (~AkkerKid@23.31.20.201) |
| 15:03.32 | AkkerKid | HEY ALL! |
| 15:04.39 | AkkerKid | anyone know how hard it would be to turn a Cisco IAD2431 into a SIP to FXS converter for fax machines using T.38? |
| 15:05.23 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 15:08.10 | eXcAliBuR | is there someone in here named Daniel ? |
| 15:08.35 | [TK]D-Fender | eXcAliBuR, Perhaps dozens. Are you looking for one in particular or will any do? |
| 15:08.51 | leifmadsen | Daniel: quick! hide! |
| 15:09.25 | kaldemar | Boichev: what is the reason for writing this info into a file anyway? |
| 15:09.28 | eXcAliBuR | well i'm looking Daniel McElveen. |
| 15:11.17 | [TK]D-Fender | eXcAliBuR, And what would his nick be? Any reason to assume he'd be here? |
| 15:11.32 | AkkerKid | I'm Sparticus. |
| 15:11.45 | Greenlight | No, I'm Sparticus |
| 15:12.32 | Boichev | kaldemar, basicly I'm tring after completing the call to go to a webpage for example 127.0.0.7/log_the_call.php?extention=xxx&dialed_num=xxxx&lenght=xxxx and I have done all the other parts that will log the call in the crm but for the first time I'm working with dialplans and AGI at all..... the output to file is just for testing.... I just need the steps to acomplish this because google info on this is small and I don't want to read the whole book in |
| 15:12.32 | Boichev | <PROTECTED> |
| 15:12.50 | igcewieling | AkkerKid: If the 2431 supports T.38 it should be easy, if it doesn't then give up now. |
| 15:12.51 | *** join/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 15:13.09 | eXcAliBuR | well i'd think a good lad like mr. mcelveen would hang out here |
| 15:13.11 | kaldemar | Boichev: how about doing that with a single line from your dialplan? |
| 15:13.13 | [TK]D-Fender | Boichev, Use the "h" exten. Call "curl" or something to hit that page. Done. |
| 15:13.18 | [TK]D-Fender | ^ |
| 15:13.39 | AkkerKid | igcewieling: the documentation says it does... I'm just not sure yet about the command line stuff... |
| 15:13.52 | eXcAliBuR | waves to sir leifmadsen |
| 15:14.02 | igcewieling | AkkerKid: likely depends on the actual IOS version you have installed. |
| 15:14.06 | AkkerKid | igcewieling: I'm looking for a tutorial or something to help me get my feet wet |
| 15:14.13 | Boichev | [TK]D-Fender, I can't get to the lenght of the call that is my problem |
| 15:14.28 | [TK]D-Fender | Boichev, I don't see you trying yet. |
| 15:14.40 | [TK]D-Fender | Boichev, Where you put things so far is wrong |
| 15:14.55 | kaldemar | Boichev: forget the AGI and do that in dialplan. |
| 15:15.13 | protocoldoug | Boichev: You might want to look up putting CDRs or CEL into a db, and pulling from that with your PHP scripts from the web |
| 15:15.53 | *** join/#asterisk sedeki (~textual@unaffiliated/sedeki) |
| 15:17.41 | *** join/#asterisk niluje (~niluje@bdv75-4-82-227-67-242.fbx.proxad.net) |
| 15:18.36 | niluje | I need to deploy the current stable version of asterisk (11) on debian. Do I mean to create the package myself or does it already exist somewhere? |
| 15:18.51 | jmetro | compile from sourcebro |
| 15:19.35 | [TK]D-Fender | niluje, Digium doesn't provide them for Debian, and Debian doesn't keep that current on their own. |
| 15:19.42 | [TK]D-Fender | niluje, So Source it is.... |
| 15:19.47 | niluje | :( |
| 15:20.13 | jmetro | its easy, read the book. |
| 15:20.34 | [TK]D-Fender | Or maybe just the tiny doc that comes with it... |
| 15:20.51 | niluje | jmetro: what do you mean? I know how to compile and run asterisk, that's not the problem |
| 15:21.21 | niluje | I need to deploy it on multiple servers on which there's no compiler, a debian package is exactly what I need :-) |
| 15:23.31 | Boichev | kaldemar, with exec? |
| 15:23.54 | kaldemar | Boichev: "exec"? |
| 15:24.19 | [TK]D-Fender | Boichev, CURL, System() SHELL(). Whever you want. |
| 15:24.37 | chris_n | does anyone have a fail2ban regex they would be willing to share built for this notice: http://pastebin.com/63UVuzVN |
| 15:25.00 | kaldemar | Boichev: Noop(${CURL(http://127.0.0.7/log_the_call.php,extension=...&clid=...)}) |
| 15:25.55 | igcewieling | chris_n: is 178.255.225.115 the IP of the server or the IP of the attacker? |
| 15:26.04 | chris_n | yes |
| 15:26.13 | *** join/#asterisk fishcooker (~chika@36.73.245.194) |
| 15:26.17 | igcewieling | you are attacking yourself? |
| 15:26.19 | Boichev | kaldemar, [TK]D-Fender aahhh let me try :) |
| 15:26.29 | fishcooker | hello |
| 15:26.37 | chris_n | igcewieling: nope |
| 15:26.47 | igcewieling | chris_n: please answer my question. |
| 15:27.02 | igcewieling | (11:25:55 AM) igcewieling: chris_n: is 178.255.225.115 the IP of the server or the IP of the attacker? |
| 15:27.06 | chris_n | argh... attacker |
| 15:27.18 | chris_n | checks to see if it is still monday |
| 15:27.58 | *** join/#asterisk ghost75 (~trechber@dslb-088-066-171-177.pools.arcor-ip.net) |
| 15:28.34 | igcewieling | chris_n: sorry, I only have a regex for failed registration |
| 15:29.49 | fishcooker | we want to develop video conferencing is there any service that we can use for building the client side.. is there any API support for our purpose? |
| 15:29.52 | chris_n | igcewieling: tnx; I'll just have to create one for it; I was hoping to save time |
| 15:30.10 | igcewieling | fishcooker: no. |
| 15:30.21 | fishcooker | Hmmmmm :-( |
| 15:30.32 | fishcooker | even asterisk? |
| 15:30.37 | fishcooker | igcewieling: |
| 15:31.05 | igcewieling | fishcooker: better information yields better answers. |
| 15:31.30 | fishcooker | we look for skype support |
| 15:31.36 | igcewieling | You can use ConfBridge and dialplan to do video conferencing, but Asterisk's support for video is rather primitive. |
| 15:31.38 | fishcooker | but no support for the api |
| 15:31.50 | fishcooker | is there any igcewieling |
| 15:32.00 | fishcooker | ? |
| 15:32.01 | igcewieling | For example, Asterisk does NOT transcode video and does NOT mix video. |
| 15:32.09 | igcewieling | fishcooker: is there any what? |
| 15:32.31 | fishcooker | any other alternative for skype? |
| 15:32.46 | igcewieling | fishcooker: I'm sure there is. Perhaps Google knows? |
| 15:33.01 | *** join/#asterisk mjordan (~mjordan@173.227.23.10) |
| 15:33.01 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 15:33.09 | igcewieling | Since you are on an ASTERISK channel, we don't know much about random video conferencing solutions. |
| 15:33.10 | fishcooker | hmmm im waiting for their answer |
| 15:33.30 | fishcooker | ok igcewieling thank for the info |
| 15:33.37 | igcewieling | Polycom has great video conferencing systems. |
| 15:33.57 | igcewieling | You need the GDP of a small country to afford it, however. |
| 15:41.13 | AlexForster | enterprise-y videoconferencing is a market that's just *dying* for disruption |
| 15:41.22 | AlexForster | strictly because of the absurd price of it all |
| 15:41.27 | *** part/#asterisk domi (~Thunderbi@mail.tas.de) |
| 15:42.40 | fishcooker | that's one of the consideration |
| 15:53.23 | *** join/#asterisk mihamina (~mihamina@12.208.74.41-ip-dyn.orange.mg) |
| 15:53.25 | FLeiXiuS | Is there a difference between [genera] and [global] in sip.conf? |
| 15:54.29 | *** join/#asterisk SuperNull (~FreeManof@24-148-101-238.ip.mhcable.com) |
| 15:54.55 | *** join/#asterisk zemmali-voip (~zemmali@unaffiliated/zemmali-voip) |
| 15:55.38 | kaldemar | FLeiXiuS: general is valid and global is not. |
| 15:55.56 | [TK]D-Fender | FLeiXiuS, Yes. The former (when spelled correctly] is an actual specially reserved setion name. the latter isn't and would become the name a a user section like any othre peer, etc would have |
| 15:57.02 | FLeiXiuS | Thanks, I'm migrating some extremely old sip configurations |
| 16:00.49 | *** part/#asterisk fishcooker (~chika@36.73.245.194) |
| 16:01.57 | igcewieling | FLeiXiuS: [global] has never been a special section in sip.conf |
| 16:05.29 | *** join/#asterisk lorsungcu (~anonymous@75-146-164-137-Minnesota.hfc.comcastbusiness.net) |
| 16:15.01 | FLeiXiuS | igcewieling, Alright, just a confusing context name. |
| 16:24.52 | *** join/#asterisk StaRetji (~LittleAll@91.142.129.1) |
| 16:25.51 | StaRetji | Folks, I have extension to read response if busy |
| 16:25.53 | StaRetji | exten => _.,n,GotoIf($[ "${busy}" = "0" ]?a2billing,${EXTEN},1) |
| 16:25.54 | StaRetji | exten => _.,n,Hangup |
| 16:26.26 | StaRetji | else, to hangup, but it seems it doesn't recognize busy, so I am not sure if GotoIf($[ "${busy}" = "0" ] is correct |
| 16:26.37 | StaRetji | any thoughts? |
| 16:27.29 | igcewieling | StaRetji: do a Noop(busy is '${busy}') to verify the variable contains what you think it contains. |
| 16:28.09 | igcewieling | Also you do NOT want to use _. or you'll rerun your code when the caller hangs up. |
| 16:28.44 | StaRetji | _. should be what? and thx for reply ;) |
| 16:29.29 | *** join/#asterisk kresp0 (~screspo@246.116.216.87.dynamic.jazztel.es) |
| 16:29.34 | igcewieling | that depends on your dialplan |
| 16:31.26 | StaRetji | well, my dialplan is to call any destination trough a2billing |
| 16:31.54 | StaRetji | so, basicaly, if ratecard exist, can make a call, if no, can't |
| 16:31.58 | igcewieling | does any destination include the telephone number "123"? |
| 16:32.12 | igcewieling | or the telephone number "456723456909456789"? |
| 16:32.15 | StaRetji | yes, 123, 00123, 222 any |
| 16:32.16 | StaRetji | :) |
| 16:32.20 | StaRetji | yes :) |
| 16:32.21 | StaRetji | hehe |
| 16:32.24 | igcewieling | Then try _X. |
| 16:32.29 | StaRetji | ok, thx ;) |
| 16:33.37 | kresp0 | StaRetji: if you use _X. remember to filter $EXTEN to dial numbers only |
| 16:33.57 | StaRetji | ehm |
| 16:34.17 | StaRetji | thx kresp0, now I am confused, I can't just replace _. for _X. |
| 16:34.18 | StaRetji | ? |
| 16:34.25 | igcewieling | StaRetji: so when someone tries to dial "rm -rf /@your domain" it won't work |
| 16:34.27 | StaRetji | lol |
| 16:34.47 | igcewieling | StaRetji: his comment about filter would apply to either pattern |
| 16:34.55 | StaRetji | ok, thx ;) |
| 16:34.55 | jmetro | nice... |
| 16:34.58 | StaRetji | nice |
| 16:36.28 | StaRetji | regarding Noop(busy is '${busy}') |
| 16:36.46 | StaRetji | this show go before my part of the code, or this is just temporary |
| 16:36.46 | StaRetji | ? |
| 16:37.40 | [TK]D-Fender | ~book |
| 16:37.41 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 16:37.42 | [TK]D-Fender | ^^ |
| 16:37.55 | [TK]D-Fender | StaRetji, If you don't know why you're doing something.. you've got bigger problems. |
| 16:38.29 | StaRetji | Thx [TK]D-Fender for the tip |
| 16:39.03 | StaRetji | I thought it worth a try to ask here |
| 16:39.42 | [TK]D-Fender | StaRetji, It's a NoOp..... it only shove a line on CLI (unless you reference a function that takes an actual action .... so that one isjust displaying the contents of a variable. Where do you expect it to have a value for it to display? |
| 16:40.04 | [TK]D-Fender | StaRetji, Where you'd put this should be fairly evident based on when you expect to actually see something of value from it |
| 16:40.20 | StaRetji | thx, clearer now |
| 16:40.35 | StaRetji | but not crystal clear lol |
| 16:40.38 | StaRetji | hehe |
| 16:40.43 | [TK]D-Fender | StaRetji, We can't tell you why you are doing something. We can only help you with what you have CHOSEN to do |
| 16:41.06 | StaRetji | yes and I appreciate help, really |
| 16:42.30 | *** join/#asterisk navaismo (~navaismo@189.241.122.125) |
| 16:43.00 | *** join/#asterisk Ice_Strike (~Ice_Black@host213-120-117-212.in-addr.btopenworld.com) |
| 16:43.09 | Ice_Strike | Hi |
| 16:45.17 | kresp0 | welcome Ice_Strike |
| 16:51.17 | *** part/#asterisk StaRetji (~LittleAll@91.142.129.1) |
| 16:52.51 | Ice_Strike | Thanks |
| 16:55.46 | *** join/#asterisk threesome (~threesome@ip-94-113-12-74.net.upcbroadband.cz) |
| 16:58.33 | *** join/#asterisk gonewage (~gonewage@host-72-2-130-205.csinet.net) |
| 16:58.40 | *** part/#asterisk gonewage (~gonewage@host-72-2-130-205.csinet.net) |
| 16:59.57 | *** join/#asterisk gonewage (~gonewage@host-72-2-130-205.csinet.net) |
| 17:00.07 | *** part/#asterisk gonewage (~gonewage@host-72-2-130-205.csinet.net) |
| 17:03.05 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 17:12.15 | kresp0 | I'm having 401 Unauthorized when trying to make a call from a softphone. This softphone registers OK (it receive calls) so the password is correct. All other clients are ATAs and hardphones that works OK (IN/OUT). I'm almost sure that has to be something stupid, but I cannot find out after few minutes. What I'm doing wrong here? |
| 17:12.18 | kresp0 | sip.conf:http://pastebin.com/fV9L43Hk |
| 17:12.22 | kresp0 | extensions.conf:http://pastebin.com/B63fGtSm |
| 17:12.26 | kresp0 | sip CLI debug:http://pastebin.com/s2DdQzyz |
| 17:15.33 | *** join/#asterisk MrMeek (~meekhime@172-4-223-5.lightspeed.toldoh.sbcglobal.net) |
| 17:15.57 | navaismo | seems like your peer isn't reponding the challenge |
| 17:21.14 | igcewieling | kresp0: When a SIP client sends a call to a SIP server it does not provide auth password, the SIP server then responds with a 401 and the client retries with a password. This is normal. |
| 17:21.36 | *** join/#asterisk teff (~teff@client-80-1-161-61.bsh-bng-011.adsl.virginmedia.net) |
| 17:22.08 | *** join/#asterisk Ice_Strike (~Ice_Black@host213-120-117-212.in-addr.btopenworld.com) |
| 17:24.33 | kresp0 | thanks navaismo and igcewieling. But * respond "401 Unauthorized" 4 times, not just the first one. |
| 17:28.50 | AlexForster | kresp0: edit your logger.conf to include console => notice,warning,error and then issue a module reload |
| 17:31.09 | *** join/#asterisk malcolmd_ (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 17:31.09 | *** mode/#asterisk [+o malcolmd_] by ChanServ |
| 17:31.27 | kresp0 | ok AlexForster, let me try that |
| 17:32.29 | kresp0 | AlexForster, I already have that line on logger.conf |
| 17:33.13 | AlexForster | hm, then on the console do core set debug 9 / core set verbose 9 |
| 17:34.30 | igcewieling | kresp0: do you see your client responding to the 401s? |
| 17:37.52 | navaismo | i didint |
| 17:38.34 | FLeiXiuS | Is there a way in asterisk console to show srtp vs unsecure calls |
| 17:42.01 | kresp0 | igcewieling, here http://pastebin.com/s2DdQzyz on line 85 I can see the response |
| 17:42.20 | kresp0 | let me try that AlexForster |
| 17:43.04 | igcewieling | kresp0: Are you SURE the client is not sending a different password for registration .vs. calls? |
| 17:43.54 | kresp0 | no, not sure. But it asked once for passwd. I'll try with another softphone |
| 17:45.36 | *** join/#asterisk ccherret1 (~christoph@S01067cb21b3313fd.ed.shawcable.net) |
| 17:45.53 | kresp0 | AlexForster, I did core set debug 9 and core set verbose 9 but nothing apears on the CLI when I try to make the call if sip set debug off |
| 17:47.23 | ChannelZ | debug is almost never useful unless you're trying to track down a bug or very obscure problem. It says lots of alarming things that aren't really that alarming. |
| 17:47.41 | kresp0 | ok, the problem was on linphone. Now I'm using sflphone and now IT WORKS :D |
| 17:47.47 | ChannelZ | And depending on how far you're getting you might not see anything with verbose. |
| 17:48.52 | kresp0 | thank you AlexForster, igcewieling, navaismo and ChannelZ |
| 17:50.47 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 17:51.49 | igcewieling | My brain must not be working today. Does anyone see an error in this? ExecIf($["${SM_DNIS_SWITCH_SID}" != "" && ${SM_DNIS_SWITCH_SID} > 7],Hangup,34) |
| 17:52.31 | Greenlight | You missing the "'s around the 2nd one |
| 17:52.56 | Greenlight | Ah wait your evaluating as numeric |
| 17:53.02 | igcewieling | Greenlight: exactly |
| 17:53.22 | igcewieling | the first condition is because sometimes SM_DNIS_SWITCH_SID will be empty |
| 17:53.53 | Greenlight | I usually seperate each evaluation into separate $[ ] secions and nest them, not totally sure if it's required though |
| 17:53.58 | Ice_Strike | Hey Greenlight |
| 17:54.04 | Greenlight | Hiya Ice_Strike hows it going |
| 17:54.05 | Ice_Strike | Greenlight Did you get my PM? |
| 17:54.43 | igcewieling | Greenlight: I'll give that a try |
| 17:54.50 | Greenlight | ExecIf(${ $["${SM_DNIS_SWITCH_SID}" != ""] && $[${SM_DNIS_SWITCH_SID} > 7],Hangup,34) |
| 17:54.52 | Greenlight | Like that |
| 17:54.58 | Greenlight | Um |
| 17:55.02 | Greenlight | ExecIf($[ $["${SM_DNIS_SWITCH_SID}" != ""] && $[${SM_DNIS_SWITCH_SID} > 7],Hangup,34) |
| 17:55.03 | Greenlight | That even |
| 17:55.17 | [TK]D-Fender | no |
| 17:55.30 | ChannelZ | it's ExecIf(statement?doiftrue:doiffalse) |
| 17:55.41 | ChannelZ | you don't seem to be using the whole thing correctly |
| 17:55.52 | Greenlight | oooh yea, the "?" |
| 17:56.29 | [TK]D-Fender | And mismatched and unnecessary extra expressions |
| 17:57.23 | Greenlight | I didn't think it was neccissary but I've always seprated them out like that, somehow seems to look neater to me |
| 17:57.38 | igcewieling | [TK]D-Fender: , is correct for asterisk 1.4 |
| 17:58.04 | igcewieling | actually let me check that, 1.4 is inconsistent between ExecIf and GotoIg |
| 17:58.30 | igcewieling | nope, I had it correct Usage: ExecIF (<expr>|<app>|<data>) |
| 17:59.08 | [TK]D-Fender | igcewieling, I wasn't really commenting on the ? |
| 17:59.23 | ChannelZ | You can yell at me. I didn't know you were in the past. |
| 18:03.42 | ChannelZ | if SM_DNIS_SWITCH_SID sometimes will be empty, that shouldn't be greater than 7, so I don't think you need to test for its emptiness |
| 18:06.08 | igcewieling | ChannelZ: When SM_DNIS_SWITCH_SID is empty the expression becomes $[ > 7] which is Not Good |
| 18:06.25 | igcewieling | This is why you test for emptyness before using < or > |
| 18:08.47 | *** join/#asterisk Chotaire (chotaire@chotaire-home.vipri.net) |
| 18:09.41 | ChannelZ | is nothing greater than 7? |
| 18:10.55 | ChannelZ | or does it throw an error? |
| 18:13.54 | *** join/#asterisk anonymouz666 (~anonymouz@189-25-77-58.user.veloxzone.com.br) |
| 18:14.28 | ChannelZ | nm tried, looks like it confuses the parser |
| 18:16.58 | *** join/#asterisk [TK]D-Fender (~Joe@216-191-106-165.dedicated.allstream.net) |
| 18:17.33 | igcewieling | yup! They should change the error "ast_yyerror(): syntax error: syntax error, unexpected '>', expecting $end;" to read "YOU FORGOT YOUR QUOTES IDIOT!" |
| 18:18.51 | ChannelZ | although if empty won't the second half still fail? Or is ExecIf smart enough to stop the expression when it sees the && and that the first expression already failed? |
| 18:20.11 | igcewieling | parsers are generally smart enough to not eval the 2nd expression if the first expression is false (when using &&, not ||) |
| 18:20.26 | ChannelZ | Yes but we are talking about Asterisk |
| 18:20.58 | igcewieling | Asterisk's parser is smart enough |
| 18:21.07 | ChannelZ | Noop($[${EXISTS(${DDFJK})}>0 && ${DDFJK}>7]) fails over here when it gets to the second eval. |
| 18:21.21 | sruffell | heh…The C preprocessor doesn't use lazy evaulation. I got hit by that not too long ago. |
| 18:21.35 | igcewieling | Noop($[${EXISTS(${DDFJK})}>0] && $[${DDFJK}>7]) |
| 18:21.42 | igcewieling | I don't know why that is needed |
| 18:22.32 | ChannelZ | not sure that's even valid, && doesn't mean anything outside an expression |
| 18:22.51 | ChannelZ | anyway, back to work for me.. |
| 18:24.01 | igcewieling | maybe $[$[${EXISTS(${DDFJK})}>0] && $[${DDFJK}>7]] 8-) |
| 18:29.27 | igcewieling | aha ExecIf($[0${SM_DNIS_SWITCH_SID} > 7],Hangup,34) |
| 18:30.52 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 18:32.00 | igcewieling | I can't believe I didn't think of that before. |
| 18:36.56 | *** join/#asterisk Sorcier_FXK (~nssystem@unaffiliated/sorcierfxk) |
| 18:37.20 | ccherret1 | will drivers and freepbx work with the latest asterisk 11.2.1? or has there been any api changes that would not allow other tools to work with the nexer asterisk? |
| 18:37.39 | ccherrett | I guess what I am asking is should I go with 1.8 or 11? |
| 18:37.47 | jmetro | 11 |
| 18:38.00 | igcewieling | ccherrett: start by reading the UPGRADE-*.txt files in the asterisk 11 source directory |
| 18:44.38 | *** join/#asterisk ke-esc (~ke-esc@155.229.78.130) |
| 18:45.02 | ke-esc | Hello all- has anybody had success with TLS signalling on Asterisk 11 and Polycom phones? |
| 18:45.27 | BludSuckingFiend | man... I think I am going to take [TK]D-Fender's advice and just ditch users.conf. Breaks all kinds of stuff |
| 18:45.38 | BludSuckingFiend | not worth it. |
| 18:45.40 | ke-esc | BludSuckingFiend, good call |
| 18:46.22 | igcewieling | users.conf is mostly there to help GUIs, not people. |
| 18:47.01 | [TK]D-Fender | Only asterisk-gui really ... and that has been dead for a long time now... |
| 18:54.55 | leifmadsen | and it doesn't even do that very well |
| 18:56.43 | *** join/#asterisk glaz (strke@hiro.glaciuz.com) |
| 19:00.33 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 19:01.25 | igcewieling | A blue bells feature |
| 19:02.54 | ke-esc | So nobodies dealt with TLS? :( |
| 19:03.52 | glaz | I have a problem with a queue, even if I set ringinuse = no, it still shows agent as "Not in use" in queue show foo |
| 19:04.39 | glaz | when on a call, obviously |
| 19:06.53 | drmessano | ke-esc: I have TLS set up, but not with Polycoms |
| 19:07.39 | *** join/#asterisk malcolmd_ (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 19:07.40 | *** mode/#asterisk [+o malcolmd_] by ChanServ |
| 19:08.38 | ke-esc | drmessano, I've gotten it working with a soft phone, but it seems there are some quirks with the poly's that are causing trouble |
| 19:10.13 | drmessano | Ah |
| 19:10.14 | glaz | Anyone? |
| 19:12.01 | navaismo | glaz, which asterisk version, how you add the member, can you show us the queue show when agent in call? |
| 19:20.15 | *** join/#asterisk mattsl (~user@68.169.165.190) |
| 19:27.30 | glaz | asterisk 11.2.1 , I add the member with the QueueAddMember() function |
| 19:27.52 | glaz | this guy is on a call now, SIP/111 with penalty 1 (ringinuse disabled) (dynamic) (Not in use) has taken 3 calls (last was 14 secs ago) |
| 19:29.30 | glaz | oh god, he left. |
| 19:30.09 | jmetro | what happens if he is on a call and another comes in, he ring? |
| 19:30.18 | jmetro | or is it just his availability not updating |
| 19:33.37 | glaz | yes |
| 19:33.47 | glaz | I think the phone is not telling asterisk that its in use |
| 19:33.50 | glaz | or something like that |
| 19:34.00 | *** join/#asterisk Sorcier_FXK (~nssystem@unaffiliated/sorcierfxk) |
| 19:34.24 | *** join/#asterisk Devon_ (~Devon_@63.214.236.169) |
| 19:34.26 | glaz | the availability is not updating |
| 19:34.49 | glaz | the call center guy picks up the phone and it stays as "Not in use" |
| 19:35.06 | jmetro | if the queue gets another call, does it ring him? |
| 19:35.14 | glaz | yes |
| 19:35.19 | jmetro | aka is the phone system actually ringing an in use phone |
| 19:35.33 | glaz | yes, but the system doesn't seem to know it's in use |
| 19:35.45 | glaz | it still says not in use, and it should be in use |
| 19:36.27 | jmetro | remove him , de-register, reboot phone, register, re-add |
| 19:36.37 | glaz | I tried |
| 19:36.53 | jmetro | remove and add with queue remove tabtabtab..etc |
| 19:37.50 | glaz | done, let me test |
| 19:39.53 | glaz | same thing |
| 19:40.36 | jmetro | i remember having thi issue once and it wound up being a syntax error with how I added him |
| 19:41.23 | glaz | me too I had this issue before |
| 19:41.42 | glaz | I know call-limit=1 would fix that, but that's a patch not a fix |
| 19:41.50 | glaz | they can't put customer on hold and make another call |
| 19:42.11 | igcewieling | try call-limit=99 |
| 19:42.48 | glaz | ok |
| 19:42.49 | igcewieling | or even a more reasonable number like 6 or 8 or whatever is more than the maximum number of calls your endpoints can handle, just as sort of a failsafe |
| 19:44.06 | glaz | I put call-limit=3 |
| 19:44.08 | glaz | let's see |
| 19:45.11 | glaz | seems like it fixed it |
| 19:45.16 | glaz | <PROTECTED> |
| 19:45.22 | glaz | im testing with a second call |
| 19:46.21 | glaz | god, it works. |
| 19:46.26 | glaz | thanks igcewieling |
| 19:47.13 | igcewieling | glaz: no problem, though more google searching may have turned up an answer. Your issue is not all that uncommon |
| 19:47.30 | glaz | you found an answer to this problem on google? |
| 19:47.34 | glaz | i googled like crazy |
| 19:47.57 | igcewieling | glaz: no. I've been using Asterisk for more than 10 years -- you pick up a few things along the way. |
| 19:48.23 | glaz | heh, I configure it once a while, even if I'm a dCAP I'm not very up to date |
| 19:48.52 | glaz | got my dCAP 6-7 years ago, didn't play with asterisk much the last 3-4 years |
| 19:49.20 | igcewieling | I manage a not small number of Asterisk servers |
| 19:49.36 | igcewieling | I think around 60 the last time I counted |
| 19:49.59 | glaz | jesus! |
| 19:50.08 | glaz | I manage Metaswitches |
| 19:50.26 | glaz | around 8000 concurent calls, but no asterisk in the way :( |
| 19:51.19 | glaz | thanks again igcewieling, I gotta run |
| 19:51.26 | igcewieling | We usually have around 400 calls transiting our core during the busy parts of the day |
| 19:51.35 | igcewieling | glaz: np |
| 19:51.41 | glaz | heh, we have schoolboards as customers |
| 19:51.45 | glaz | they push a lot of calls |
| 19:51.56 | igcewieling | heh, we have a school bus company as a customer |
| 19:52.06 | glaz | you sell hosted services? |
| 19:52.52 | glaz | oh well, I really gotta run, later! |
| 19:53.00 | igcewieling | glaz: only when forced to. 8-| We usually provide the internet service so we can QoS the circuit and put media GWs in a the customer to handoff to their PBX (at the non-asterisk sites) |
| 19:53.52 | ke-esc | 0320154100|sip |4|03|[TLS] Server Certificate Common Name 'asterisk-na.blah.net' doesn't match any of the following: |
| 19:53.53 | ke-esc | 0320154100|sip |4|03|[TLS] Hostname: asterisk-na.blah.net |
| 19:53.58 | ke-esc | wtf!?! blargh |
| 19:55.43 | igcewieling | ke-esc: what version of Asterisk? |
| 19:56.01 | ke-esc | 11.2.0 |
| 19:56.26 | igcewieling | I think I saw an e-mail about SRTP being broke in 11.3-rc but I don't know for sure. |
| 19:56.50 | ke-esc | igcewieling, I haven't even gotten to srtp yet |
| 19:57.20 | igcewieling | I suspect all the crypto uses the same code |
| 19:58.07 | AkkerKid | anyone know how to reset a locked cisco iad2400 series device to factory defaults without the password? |
| 19:58.12 | ke-esc | igcewieling, true..i'll have to dig for that a bit |
| 19:58.13 | igcewieling | nothing in the changelog for 11.2.0 -> 11.2.1 |
| 19:58.33 | igcewieling | also try adding a . at the end of the full host name |
| 19:59.01 | igcewieling | I doubt that is the issue, but worth a try. |
| 19:59.02 | drmessano | SRTP seems to be wonky period |
| 19:59.14 | drmessano | But I think the fix is in libsrtp |
| 19:59.41 | drmessano | As in, the issue is libsrtp, and I believe there are patches on the SF page between the tickets and forum |
| 19:59.45 | jmetro | akkerkid: flash it with new firmware? |
| 19:59.48 | ke-esc | in otherwords its still probably best to stick with a sip server in front of asterisk? |
| 20:00.04 | igcewieling | or stop being so paranoid 8-) |
| 20:00.17 | ke-esc | igcewieling, NEVER! :) |
| 20:00.30 | drmessano | ke-esc: For SIP TLS and SRTP, more and more I am finding out that indeed that may be necessary.. There's LOTS of SMALL issues, and you'll end up chasing them until your hair falls out |
| 20:00.45 | igcewieling | AkkerKid: Cisco would have documentation on how to do that |
| 20:01.00 | jmetro | we had another guy trying to do SRTP and i think he wound up spontaneously combusting |
| 20:01.20 | ke-esc | drmessano, I did start playing around with kamailio a few days ago- but its got a pretty huge learning curve.. |
| 20:01.28 | ke-esc | and the docs are pretty shoddy |
| 20:01.58 | drmessano | ke-esc: I have been looking for a simple something to put in front of Asterisk for TLS and SRTP, but yeah.. |
| 20:02.03 | igcewieling | jmetro: I heard SRTP caused the TUNGUSKA event of 1908 |
| 20:02.26 | ke-esc | igcewieling, I thought that was zrtp? |
| 20:02.48 | igcewieling | Perhaps it was. |
| 20:03.27 | drmessano | ke-esc: If you check the SRTP SF page, at the very least there is a patch in a CLOSED ticket that hasn't been committed that mentions in the description "This patch resolves a crash in Asterisk that we could easily reproduce", paraphrasing |
| 20:03.59 | *** join/#asterisk kontinui_ (~kontinuit@122.178.229.113) |
| 20:04.10 | ke-esc | hmm..i'll take a look at that |
| 20:04.40 | AkkerKid | Regarding Cisco: all I have is a serial connection to the box so far... |
| 20:04.45 | drmessano | ke-esc: SO there's SRTP issues that break Asterisk, Asterisk issues that break SRTP, and that doesn't count the few TLS issues sitting in JIRA with patches that have yet to be commited |
| 20:04.52 | AkkerKid | or a "console" cable as they call it |
| 20:05.23 | drmessano | ke-esc: I have a handful of patches I have gathered, and built SRTP from source to get to where I am now.. which is hardly anywhere |
| 20:05.58 | ke-esc | drmessano, does freeswitch handle it any better you think? |
| 20:06.17 | igcewieling | AkkerKid: every cisco I've had to bypass the password on I did with only a console cable |
| 20:06.27 | drmessano | ke-esc: I don't know.. I had considered that as well.. |
| 20:06.38 | *** part/#asterisk kontinui_ (~kontinuit@122.178.229.113) |
| 20:07.02 | AkkerKid | igcewieling: well this is going to be the first Cisco device I've ever bothered trying to use so it's new to me |
| 20:07.19 | igcewieling | AkkerKid: Cisco should have the procedure documented on their site. |
| 20:07.23 | jmetro | AkkerKid: Like most macs, the most efficient use is boat-anchor. |
| 20:07.34 | file | the only thing that we (as Asterisk) do in regards to SRTP is negotiate it - otherwise we just call into libsrtp to encrypt/decrypt, and to set the keying details and such |
| 20:07.38 | drmessano | ke-esc: Of course, all this will be irrelevant when Ast 12 comes out, as chan_sip will be completely new.. and it may just work |
| 20:07.44 | AkkerKid | that's the first thing I tried. but it seems they assume there's already no password protection o nthe device |
| 20:08.10 | igcewieling | drmessano: the first version of Asterisk to use a totally new sip stack? I am not optimistic. |
| 20:08.23 | igcewieling | AkkerKid: incorrect. |
| 20:08.36 | file | I think actually the new SIP work will do very very nicely... |
| 20:08.39 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
| 20:09.08 | igcewieling | When the router start booting you send a break to it to get into rommon, then set a config register to tell the router to NOT read the config file (and password) and boot the router. The exact method varies depending on the router. |
| 20:09.10 | AkkerKid | it says "method 1 uses the config-register 0x2102 command in global configuration mode." what the heck does that mean? |
| 20:09.22 | AkkerKid | hmmm. |
| 20:09.29 | ke-esc | intruiging... of course I was hoping to get started with the LTS release :( |
| 20:09.31 | igcewieling | file: I'm sure it will eventually, but I'm not optimistic about the first release. |
| 20:09.37 | AkkerKid | i think i have a break key on my keyboard... |
| 20:10.04 | file | igcewieling, it won't have feature parity but it will have the major stuff - and so far even under the load testing I've done it's been fine but only the real world can truly tell |
| 20:10.38 | drmessano | ke-esc: I can PM you a few of the specifics I have here regarding patches and workarounds.. It may not be as bloody as it looks. Some of these I have found in just the last few days |
| 20:12.38 | ke-esc | drmessano, thanks, but I think I'm going to go back to messing with sip servers in front of asterisk.. This is going to be for a production system so I really don't want to start messing around with patches |
| 20:14.57 | AkkerKid | igcewieling: ok, i've gotten a "rommon 1 >" prompt. how do I turn that into a "router#" prompt? |
| 20:15.25 | igcewieling | Addisk: I am not going to search the Cisco web site and read the docs for you. |
| 20:15.44 | igcewieling | If you need additional help I suggest you try #cisco |
| 20:16.04 | AkkerKid | k. |
| 20:16.08 | drmessano | If you can get someones attention that maintains libsrtp, there's a patch for a memory leak and then the one that allegedly prevents the asterisk "crash" that should be merged, and they should stamp out a release for the first time in 6 years |
| 20:16.34 | drmessano | Because the diff between trunk and 1.4.4 isn't huge, but there's some work that has been done |
| 20:17.23 | drmessano | Then you won't be using a "patched libsrtp" :) |
| 20:18.21 | ke-esc | true true |
| 20:18.47 | drmessano | Which distro are you running? |
| 20:19.06 | ke-esc | i'm on centos 6 |
| 20:19.53 | drmessano | Ah.. because I was going to reach out to the maintainer of the debian packages and see what they thought about getting these patches in |
| 20:24.47 | *** join/#asterisk bpgoldsb (~bpgoldsb@li134-54.members.linode.com) |
| 20:25.30 | bpgoldsb | Using users.conf, is it possible to have an the dialplan for a extension put someplace besides the context 'default'? |
| 20:30.01 | *** join/#asterisk imox (~imox@91-66-32-57-dynip.superkabel.de) |
| 20:35.49 | *** join/#asterisk FLeiXiuS (~FLeiXiuS@unaffiliated/fleixius) |
| 20:36.27 | AlexForster | [1234] / username = 1234 / secret = $ecret / context = someplace-besides-default / ... |
| 20:37.32 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 20:52.55 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:06.30 | *** join/#asterisk kresp0 (~kresp0@81.61.25.130.dyn.user.ono.com) |
| 21:07.47 | bpgoldsb | AlexForster: That handles their subscribe context, but not where their stdexten dialplan entry is made (sorry if my Asterisk vocab is a bit weak) |
| 21:08.25 | *** part/#asterisk keycruncher (~Adium@c-174-55-112-94.hsd1.pa.comcast.net) |
| 21:09.49 | *** join/#asterisk Ice_Strike (Ice_Strike@87.115.83.87) |
| 21:11.16 | Ice_Strike | I am having performance problem - it is really slow copying file from my drive to network |
| 21:11.38 | Ice_Strike | I am only getting between 7MB/sec to 30MB/sec |
| 21:11.40 | jmetro | performance problems |
| 21:11.52 | AlexForster | hm, it looks like sip.conf is the place to define where the originating context is |
| 21:12.15 | Ice_Strike | Ooops |
| 21:12.18 | Ice_Strike | wrong channel lol |
| 21:14.15 | bpgoldsb | AlexForster: I tried looking through there, but admittedly, I got lost a few times. Any specific ideas? |
| 21:16.32 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 21:17.34 | AlexForster | the stock users.conf header basically says "ignore me and use sip.conf" |
| 21:18.18 | AlexForster | it looks like you want to stick to sip.conf |
| 21:18.32 | *** join/#asterisk Sorcier_FXK (~nssystem@unaffiliated/sorcierfxk) |
| 21:24.18 | *** join/#asterisk elico (~Thunderbi@bzq-79-176-184-165.red.bezeqint.net) |
| 21:46.18 | *** join/#asterisk mchou (~quassel@unaffiliated/mchou) |
| 21:49.58 | *** join/#asterisk sp00kz (~ilubj00@unaffiliated/sp00kz) |
| 21:58.56 | *** join/#asterisk benasse (~Thunderbi@176.222.51.3) |
| 22:11.26 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 22:32.42 | *** join/#asterisk Linkforsoad (~Linkforso@2001:1af8:fec1:0:d4ad:de96:c365:4dd3) |
| 22:48.24 | *** join/#asterisk sedeki (~textual@unaffiliated/sedeki) |
| 22:51.19 | *** join/#asterisk tamiel (~tamiel@197.158.64.3) |
| 22:53.14 | *** join/#asterisk navaismo (~navaismo@189.241.122.125) |
| 22:54.29 | *** part/#asterisk gladier (~gladier@pedobear.molested.me) |
| 22:58.56 | ccherrett | anyone know if the sangoma driver installs to the newest kernel |
| 23:06.16 | *** join/#asterisk digilink (~digilink@unaffiliated/digilink) |
| 23:07.53 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-toznkfmuuuxmpcyz) |
| 23:07.53 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 23:19.50 | *** join/#asterisk Sorcier_FXK (~nssystem@unaffiliated/sorcierfxk) |
| 23:28.01 | *** join/#asterisk Neptu (~Hej@c213-89-2-159.bredband.comhem.se) |
| 23:30.17 | *** join/#asterisk zerohalo (~zerohalo@74.61.196.236) |
| 23:36.37 | *** join/#asterisk Petchaw (~Pascal@198.105.46.46) |
| 23:39.40 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 23:39.40 | *** mode/#asterisk [+o pabelanger] by ChanServ |