| 00:02.20 | *** join/#asterisk b2 (~ion@pdpc/supporter/active/beckb) |
| 00:04.10 | *** join/#asterisk SunTsu (miyamoto@unaffiliated/suntsu) |
| 00:04.51 | *** join/#asterisk eerie (hoax@gateway/shell/bshellz.net/x-strkzfgmnrorpcdc) |
| 00:05.55 | *** join/#asterisk saxa (~sasa@host242-95-static.223-217-b.business.telecomitalia.it) |
| 00:07.04 | saxa | [Mar 11 01:14:53] WARNING[2276]: chan_iax2.c:4894 handle_call_token: Address xxx.xxx.xxx.xxx failed CallToken hash inspection |
| 00:07.09 | saxa | anybody know what this means ? |
| 00:07.38 | saxa | i just saw that in my console, * 1.8.2.3 |
| 00:08.44 | *** join/#asterisk CRCinAU (~CRCinAU@2002:cb38:f71b:1:badb:adc0:ffee:1) |
| 00:08.58 | *** part/#asterisk CRCinAU (~CRCinAU@2002:cb38:f71b:1:badb:adc0:ffee:1) |
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| 00:09.11 | CRCinAU_ | p3nguin: ping? |
| 00:21.52 | Carlos_PHX_ | Just when you think you've seen everything there is to be seen in the IP telephony world, you find that someone has driven sheet metal screws into a Polycom 501 because they were too stupid to line up the snap-on base. |
| 00:22.22 | CRCinAU_ | Mmmmm conductive |
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| 00:43.32 | jdoe | Carlos_PHX_: so YOU'RE where my "working pulls" come from... |
| 00:54.41 | Carlos_PHX_ | jdoe: LOL |
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| 01:09.37 | raden_work | anyone know how to fix music on hold issue in asterisk 1.8 ? |
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| 01:15.27 | quidpro | Any ODBC gurus here? |
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| 01:29.33 | CRCinAU_ | raden_work: care to be a bit more vague? ;) |
| 01:30.47 | raden_work | My music on hold does not work worked in 1.6 but wont work on 1.8 |
| 01:31.16 | raden_work | I have aastra 9133i and 480i phones |
| 01:31.39 | raden_work | my PAP2t when i flash will play music on hold |
| 01:32.04 | raden_work | i can call my music on hold extension and it works |
| 01:32.10 | raden_work | just not when i hit the hold button |
| 01:32.55 | CRCinAU_ | well, considering it works on one device, and not the others, it seems to be an issue with what the aastra phones are doing when you hit the hold button... |
| 01:33.15 | raden_work | this is just nuts |
| 01:33.22 | raden_work | i agree it the phones |
| 01:33.28 | raden_work | but why did asterisk 1.6 |
| 01:33.31 | raden_work | woirk |
| 01:34.05 | CRCinAU_ | maybe look at a sip debug? see what is being sent? |
| 01:34.15 | raden_work | hmmmm |
| 01:34.18 | CRCinAU_ | firmware update on the phone? |
| 01:34.31 | CRCinAU_ | maybe its an Aastra bug? |
| 01:34.44 | raden_work | just why would it be all phones :( |
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| 01:43.45 | *** part/#asterisk ManxPower (~ewieling@user-24-214-153-32.knology.net) |
| 01:47.18 | *** join/#asterisk LemensTS (~matthew@70.238.136.43) |
| 01:53.13 | paulc | Isn't there some RFC type message that indicates a hold condition? maybe handling of it changed |
| 01:58.26 | *** join/#asterisk fross (~fross@68-112-140-187.dhcp.stcd.mn.charter.com) |
| 02:00.27 | fross | its it possible to send out a fax on a sip trunk? |
| 02:01.28 | CRCinAU_ | what is this 'sip trunk' you speak of? ;) |
| 02:01.49 | CRCinAU_ | but yes, you can send faxes on a SIP peer using alaw/ulaw |
| 02:02.41 | LemensTS | If calling in on Dahdi to a sip phone, does jitter buffer do anything in sip.conf ? |
| 02:03.18 | fross | ok, cuase I have gotten to the point that i can call out on my trunk. Just need to figure out the fax. Ill look into alaw/ulaw |
| 02:04.03 | CRCinAU_ | fross: well it stands to reason you can't use g729 etc as they compress the audio. Fax data cant be compressed like voice |
| 02:04.33 | CRCinAU_ | so you can only use g711a/u (alaw or ulaw) to transfer the entire uncompressed bitstream |
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| 02:32.08 | p3nguin | ~siptrunk |
| 02:32.08 | infobot | from memory, siptrunk is something that doesn't exist -- there is no concept of a SIP trunk in Asterisk. You may be searching for iaxtrunk |
| 02:32.35 | CRCinAU_ | p3nguin: welcome! :) |
| 02:32.40 | CRCinAU_ | p3nguin: may I pm you for a moment? |
| 02:33.01 | p3nguin | Is it for something well-suited for the public channel? |
| 02:33.09 | CRCinAU_ | not really... |
| 02:33.25 | p3nguin | One moment, while I make sure you can get through. |
| 02:33.50 | p3nguin | Alright, give it a shot. |
| 02:36.13 | *** join/#asterisk karmst (~kevin@s1.acclaimpc.com) |
| 02:36.15 | karmst | Hello |
| 02:36.20 | karmst | anyone in here tonight? |
| 02:36.40 | *** join/#asterisk root52 (~root52@ip70-191-116-76.cl.ri.cox.net) |
| 02:36.57 | p3nguin | infobot: give karmst ask |
| 02:36.57 | infobot | ACTION gives ask to karmst |
| 02:37.05 | p3nguin | infobot: tell karmst about ask |
| 02:38.05 | CRCinAU_ | nothing? |
| 02:38.44 | karmst | When you build an asterisk box and you have only IP phones on the system, and you are only going to be using SIP trunking... Do you need an FXO card or anything? Or can you just use the Intel GB nics on the server? |
| 02:38.48 | p3nguin | I don't know, you tell me. I thought you had something you wanted to tell me that wasn't fit for public view. |
| 02:38.58 | root52 | Hey all. What is up with this. SIP.conf is set for two codecs ulaw first and g722 second. (In fact it was straight copied from a working server.) When I do sip show peer ... the codec says nothing. As i'm sure you know this is causing calls to fail with the error. "No compatible codecs" Asterisk 1.8.3. Any Thought. |
| 02:39.09 | CRCinAU_ | p3nguin: I sent you a pm... but got no reply - I assume it didn't work? :P |
| 02:39.10 | p3nguin | Ugh, again with the SIP trunking. |
| 02:39.26 | p3nguin | There is STILL no trunking support in SIP. Still. |
| 02:39.32 | p3nguin | ~siptrunk |
| 02:39.32 | infobot | from memory, siptrunk is something that doesn't exist -- there is no concept of a SIP trunk in Asterisk. You may be searching for iaxtrunk |
| 02:39.41 | p3nguin | karmst: ^^^^ |
| 02:39.41 | p3nguin | Still. |
| 02:39.46 | root52 | karmst: no need for an all IP system. |
| 02:40.03 | root52 | but yeah what p3nguin said ;-) |
| 02:40.15 | karmst | well we have an all IP system |
| 02:40.39 | CRCinAU_ | same here. |
| 02:40.41 | karmst | I'm trying to find out if I need to purchase and special hardware, or if I can just use an ESX vm and run everything in it |
| 02:40.46 | karmst | any* |
| 02:41.01 | karmst | all the documentation talk about FXO cards |
| 02:41.10 | p3nguin | root52: In the peer definition, enter disallow=all and allow whichever codec you want to use, as long as the other side supports it. |
| 02:41.48 | p3nguin | Most telephony professionals will not recommend running VoIP systems from a virtual machine. |
| 02:42.32 | karmst | Ok... But it's ESX and with 4 SANs and a pool of 58 blades |
| 02:43.10 | p3nguin | It's a virtualization platform. If you run a virtual machine in it and put Asterisk on it, you fall into the category I just listed. |
| 02:43.41 | karmst | fine p3nguin but do you need special interface cards? |
| 02:43.45 | root52 | p3nguin: WOW me fail. When I said "straight copied" I meant from memory. Egg on my face.. I had the disallow after the codecs i want not before like it needs to be. Thanks for saying the work disallow that smaked me in the head |
| 02:43.45 | *** join/#asterisk micols (~mio@rlogin.dk) |
| 02:43.51 | karmst | or do you just use NICs? |
| 02:44.10 | p3nguin | If you are running IP only, you only need a NIC. |
| 02:45.40 | karmst | and you were saying SIP isn't supported? |
| 02:46.02 | karmst | our SIP provider gives detailed instructions on how to connect to them |
| 02:46.10 | paulc | Our IT manager really wanted to run Asterisk on a VM.. I'm trying to remember what for.. what the project was.. I wasn't happy - said that VoIP being time sensitive means it's better on real hardware.. that said, Mitel have been publishing stuff lately about running their UC solutions in VM.. so.. I dunno.. |
| 02:46.28 | karmst | oh yeah |
| 02:46.32 | paulc | karmst: He's saying that SIP is fine, you don't need analog hardware for timing etc - his real point was that there's no such thing as "a SIP trunk" |
| 02:46.35 | p3nguin | karmst: I said there is no TRUNKING in SIP. |
| 02:46.47 | paulc | or.. what he said ^^^ |
| 02:46.49 | karmst | VMs can run faster than physical hardware |
| 02:47.07 | *** join/#asterisk BeeBuu (3afc49a5@gateway/web/freenode/ip.58.252.73.165) |
| 02:47.07 | karmst | because you can pool resources together from multiple systems |
| 02:47.16 | paulc | "a SIP trunk" and "SIP trunking" is not the same thing (I knew it but kinda stand corrected - it's been a long day!) |
| 02:47.23 | p3nguin | There's no such thing as a SIP trunk. |
| 02:48.55 | karmst | i get it now |
| 02:49.02 | karmst | ;) |
| 02:50.36 | BeeBuu | i found a issue as follow: using a sip channel call a dahdi channel,when dahid channel hangup, the sip hangup after about 6 seconds. how can i change this? |
| 02:51.06 | paulc | BeeBuu: how do you want to change it? Do you want the 6 seconds to be less? |
| 02:51.25 | *** join/#asterisk csnook (~chris@c-76-19-64-161.hsd1.ma.comcast.net) |
| 02:51.28 | karmst | paulc that's old tech thinking |
| 02:51.54 | karmst | did you see where Exchange 2010 runs faster and more efficent being virtualized? |
| 02:52.09 | BeeBuu | paulc,i want the sip channel hangup immediately |
| 02:52.28 | BeeBuu | paulc: how can i do that? |
| 02:52.33 | paulc | karmst: no, I haven't. I'm not doubting what you're saying. I'm questioning it applies as well when latency is an issue |
| 02:52.59 | karmst | I guess that would depend on your server nics? |
| 02:53.17 | paulc | BeeBuu: What version of Asterisk are you running? And do you see the DAHDI channel hang up as soon as the remote end hangs up, or does it show on the console a few seconds after you hear the click? |
| 02:53.47 | BeeBuu | paulc: asterisk 1.6 |
| 02:56.21 | paulc | BeeBuu: Ok - and what about the DAHDI behaviour? |
| 02:56.37 | BeeBuu | actually,i want the "billsec" to be real as the service suppliers |
| 02:57.25 | paulc | Using AOC you mean? It's hard for outbound calls, because in the TDM world you can still hold a channel open until the timer expires. |
| 02:59.04 | BeeBuu | yes,paulc,that's what i want |
| 02:59.36 | BeeBuu | i'm using E1 |
| 03:00.05 | paulc | BeeBuu: Have you read https://wiki.asterisk.org/wiki/display/AST/Advice+of+Charge ? |
| 03:00.27 | paulc | I understand E1 (I'm from the UK originally) and understand the telephony concepts, but I've done little with it in Asterisk so far. |
| 03:02.00 | BeeBuu | paulc: thanks tons.Dial(DAHDI/g1/1112/A(s)) can do that? |
| 03:02.39 | paulc | I think you want A(e) though, for AOC at the END of the call? maybe? I'm not sure |
| 03:03.33 | BeeBuu | paulc: thanks,i will try it later soon. |
| 03:04.45 | paulc | Ok.. I'm off for steak'n'beers :-) |
| 03:06.46 | BeeBuu | paulc: add "aoc_enable=s,d,e" to where? |
| 03:07.18 | BeeBuu | beer? so good~~~ |
| 03:10.03 | CRCinAU_ | I did have issues when running asterisk in a Xen DomU |
| 03:10.12 | CRCinAU_ | it seemed to do with timing of various things |
| 03:10.37 | CRCinAU_ | ie voicemail would go "You have.... 4..... new messages...... press .... 1 .... for new messages" etc |
| 03:10.48 | CRCinAU_ | so now I run it on real hardware |
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| 04:09.45 | retentiveboy | Anybody notice the latest asterisk-core RPM in Digium's YUM repository now includes phoneprov configs? |
| 04:12.36 | *** join/#asterisk radic (~radic@dslb-094-216-251-226.pools.arcor-ip.net) |
| 04:13.03 | retentiveboy | I like to keep /etc/asterisk in an SVN repository so I don't install the asterisk-configs package but after a "yum update" I got additions/overwirites in /etc/asterisk/phoneprov. |
| 04:13.38 | retentiveboy | RPM doesn't seem to think a package owns stuff there. Weird. |
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| 05:23.48 | raden | Anyone know why aastra phones Hold button does not start music on hold in asterisk 1.8.2 and 1.8.3 but it works in 1.6 ? |
| 05:24.18 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 05:28.16 | kaldemar | raden: try setting pedantic=no in sip.conf under general. |
| 05:28.32 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 05:29.29 | *** join/#asterisk citywok (~citywok@67-134-194-33.dia.static.qwest.net) |
| 05:29.29 | raden | kaldemar, what does that do ? |
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| 05:33.14 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 05:36.39 | kaldemar | raden: it enables strict checking of headers and tags, the default was changed from yes to no in 1.8.0. |
| 05:37.09 | raden | hmmmm |
| 05:37.28 | raden | so i should set it to yes ? |
| 05:38.29 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 05:39.41 | kaldemar | s/yes to no/no to yes/ |
| 05:41.40 | kaldemar | see sip.conf.sample and UPGRADE.txt in the source package. |
| 05:42.55 | raden | ill have to look that over again |
| 05:43.06 | raden | as usual everything at work being done 500x faster than practical |
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| 06:08.49 | raden | kaldemar, tried it both ways still not working |
| 06:08.54 | raden | nothing showing up on console either :( |
| 06:09.01 | raden | this is really starting to drive me nuts |
| 06:09.10 | raden | and i really dont want to revert back to 1.6 |
| 06:09.15 | CRCinAU_ | so downgrade? |
| 06:09.35 | raden | bla going back to 1.6 seems like a headache now that all is setup |
| 06:09.51 | raden | I have never seen anything soooo crazy |
| 06:09.55 | raden | MOH not working |
| 06:14.40 | CRCinAU_ | raden: I figure its something to do with your config.... as I figure people would notice that kind of bug... |
| 06:14.59 | CRCinAU_ | maybe its a context problem etc.... |
| 06:16.02 | raden | CRCinAU_, all other phones work |
| 06:16.11 | raden | PAP2T polycoms etcc |
| 06:16.16 | raden | just not the aastra |
| 06:16.24 | raden | and it was fine with version 1.6 |
| 06:16.30 | raden | and this is a issue im seeing more and more |
| 06:21.01 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 06:25.22 | kaldemar | raden: do you see the messages in sip debug? |
| 06:25.55 | raden | let me see if I can catch one |
| 06:27.25 | *** join/#asterisk zloidemon (~zloidemon@87.250.220.18) |
| 06:27.35 | zloidemon | hi everybody |
| 06:27.36 | raden | how can i just set debug on one peer |
| 06:30.09 | zloidemon | please see my log http://pastebin.com/3fzhqxwH :) I can't call with use the 73912069613, but I can call with use a 73912069611 and 73912069612. |
| 06:31.15 | *** join/#asterisk BeeBuu (78c5c19c@gateway/web/freenode/ip.120.197.193.156) |
| 06:31.39 | raden | kaldemar, WHEN I HIT HOLD http://pastebin.com/F09KSXwe |
| 06:31.47 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 06:35.52 | CRCinAU_ | raden: none of that looks to me like a hold event? unless I'm missing something.... |
| 06:36.17 | raden | CRCinAU_, I dont even know what to look for |
| 06:36.31 | raden | I know what hold looks like on the CLI normally but never seen a debug |
| 06:37.23 | CRCinAU_ | hmmm |
| 06:37.24 | CRCinAU_ | http://fonality.com/trixbox/forums/trixbox-forums/trixbox-endpoints/aastra-phones-troubles-music-hold |
| 06:37.34 | CRCinAU_ | maybe like I said originally, firmware update the aastras |
| 06:37.46 | raden | The problem can't be resolved. Looks like Aastra push us to by new equipment instead to upgrade firmware. And we find new equipment, Polycom phones. |
| 06:38.04 | CRCinAU_ | then try downgrading firmware. |
| 06:38.12 | CRCinAU_ | experiment :) |
| 06:38.14 | raden | lemee look |
| 06:38.43 | raden | I have the newest |
| 06:38.55 | raden | everything worked flawless in 1.6 |
| 06:39.03 | raden | so this is getting really annoying |
| 06:39.07 | CRCinAU_ | http://www.freepbx.org/forum/freepbx/users/aastra-6757i-ct-hold-issue |
| 06:41.11 | raden | yea its like the stupid phone handling hold internally |
| 06:41.25 | CRCinAU_ | maybe it is... |
| 06:43.13 | raden | why did it work on asterisk 1.6 ? |
| 06:43.16 | raden | and not 1.8 |
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| 06:48.19 | raden | *sigh* |
| 06:52.34 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 06:54.06 | raden | any other ideas |
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| 07:43.36 | *** join/#asterisk OldMonk (~raju@124.124.85.98) |
| 07:43.38 | OldMonk | hi |
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| 07:43.52 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 07:43.55 | OldMonk | what would be the best way to start recording a call AFTER the called party answers? |
| 07:44.15 | OldMonk | Dial() doesn't return until the call is over |
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| 07:50.27 | jkroon | has anybody managed to get SIP call transfers on Siemens Gigaset (A580IP) working by any chance? |
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| 08:28.43 | flashdeluxe | jkroon: Whats exactly the problem? |
| 08:29.23 | flashdeluxe | jkroon: I have several gigasets, eg C470IP and it works |
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| 08:42.13 | asterisk-learner | Hi, does anyone knows if i can "auto-answer" a polycom Soundpoint 330 thru the dialplan ? |
| 08:42.51 | asterisk-learner | SIPAddHeader(Alert-Info: Ring Answer) doesnt seem to be working |
| 08:43.29 | carrar | You need to add that into the polycom config files |
| 08:44.06 | carrar | tell it what Alert-Info tag to look for and what you want it to do once it see's it |
| 08:44.24 | carrar | might gander at the admin guide |
| 08:44.58 | asterisk-learner | carrar: mmmm ok thx |
| 08:45.48 | asterisk-learner | do u think it might be done from the web interface ? or u doubt |
| 08:45.54 | carrar | haha |
| 08:45.55 | carrar | no |
| 08:46.03 | carrar | web interface is a joke |
| 08:46.17 | asterisk-learner | :-( |
| 08:46.53 | *** join/#asterisk SiNGLer (~singler@78-60-139-121.static.zebra.lt) |
| 08:48.31 | asterisk-learner | too bad, on my linksys SPA 942 it worked like a charm |
| 08:48.59 | asterisk-learner | using this of course : exten => _XXXX,n,Set(__SIPADDHEADER=Call-Info:<sip:myip>\; answer-after=0) |
| 08:49.08 | asterisk-learner | exten => _XXXX,n,SIPAddHeader(Alert-Info: RA) |
| 08:50.05 | carrar | if you plan to use the polycom in any usfull capacity you are going to need to dig into the phone configs |
| 08:52.08 | *** join/#asterisk manji (~manjiki@94-193-165-46.zone7.bethere.co.uk) |
| 08:52.38 | kaldemar | asterisk-learner: put voIpProt.SIP.alertInfo.1.value="auto-answer" in <alertInfo/> in the phone config file and then SIPAddHeader(Alert-Info: auto-answer) in dialplan. it won't work from dialplan alone, you must change the phone configuration. |
| 08:53.20 | carrar | heh |
| 08:53.29 | carrar | assuming he is even using config files :) |
| 08:53.42 | carrar | of which I doubt |
| 08:53.44 | asterisk-learner | carrar: Bingo, i was not :-$ |
| 08:54.11 | carrar | setup a FTP server to config your polycoms |
| 08:54.14 | asterisk-learner | can i access polycom files thru filezilla or the like |
| 08:54.42 | asterisk-learner | if yes, arent the credentials the same as the web interface ? |
| 08:55.02 | carrar | You need to start with downloading the admin guide |
| 08:55.24 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 08:55.39 | asterisk-learner | ok |
| 08:55.41 | kaldemar | carrar: if he wants auto answer, he will use config files. :) |
| 08:56.30 | carrar | yup |
| 09:04.23 | *** join/#asterisk awclin (~alinford@80.169.133.251) |
| 09:09.10 | asterisk-learner | I downloaded the admin guide, found the default user name and password and still cant access it thru ftp |
| 09:09.28 | asterisk-learner | is there a way i can "factory" reset the phone manually ? |
| 09:09.46 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-ghdeyprgtuqrvggw) |
| 09:11.02 | kaldemar | asterisk-learner: the phone does not have an ftp server, you need to setup one on your server and tell the phone where it is. the phones try to download configuration and firmware upon boot automatically. |
| 09:12.34 | asterisk-learner | kaldemar: so i cant "read" the already existing config files on it ? |
| 09:12.56 | kaldemar | you don't need a factory reset, but it is described in the admin manual. |
| 09:13.55 | kaldemar | asterisk-learner: i don't know if there's a way to read the current configs. |
| 09:14.51 | asterisk-learner | kaldemar: ok thx |
| 09:15.14 | *** join/#asterisk `paul (~kutimoy@119.93.15.82) |
| 09:15.38 | `paul | how do you implement random() in ael? (want to record calls 10% of the time) |
| 09:16.21 | kaldemar | `paul: which version of asterisk are you using? |
| 09:16.25 | `paul | 1.4 |
| 09:17.32 | `paul | or should i just use ${RAND(0,100) < 10 ? |
| 09:17.41 | kaldemar | yes, use func RAND. |
| 09:22.34 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 09:24.35 | *** join/#asterisk Denial (Denial@drgi.co.uk) |
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| 09:34.48 | *** join/#asterisk Cadey (~Cadey@62.84.178.106) |
| 09:36.13 | Cadey | Hi peeps, I have put a + in front of my caller ID but when i call my mobile that + gets turned into a 0 and so i cant call it back, am I doing anything wrong here? So for example my caller ID is going out as +448444128704 but showing up on peoples caller ID's as 0448444128704 which in an invalid number |
| 09:36.55 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:b156:dcdc:4540:8d30) |
| 09:37.35 | WIMPy | So someone inbetween seems to replace it. Are you calling out via an ITSP? |
| 09:37.52 | Cadey | im calling out over a SIP trunk? |
| 09:38.14 | WIMPy | Then ask your ITSP what's an acceptable format for them. |
| 09:39.39 | Cadey | oh right ok so its at the carrier level - nothing stupid im doing? |
| 09:40.54 | WIMPy | No, it's in fact the best way to do it, you did, but if you're pedantic, a + isn't allowed. |
| 09:41.14 | WIMPy | There is no other format that makes sense worldwide. |
| 09:42.03 | WIMPy | So it might be a good idea to convince them that it's a good idea. |
| 09:42.30 | wdoekes2 | + is perfectly valid |
| 09:42.41 | wdoekes2 | if pedantic breaks it, then pedantic is broken |
| 09:43.02 | WIMPy | Not the pedantic in the Asterisk configuration. |
| 09:43.08 | wdoekes2 | check the sip debug.. if you send a literal +, it's good |
| 09:43.51 | Cadey | sorry whats pedantic? |
| 09:44.07 | wdoekes2 | (if you send %2B you're looking at a bug that is fixed in some versions only) |
| 09:45.48 | wdoekes2 | Cadey: you can try double-0 instead |
| 09:45.57 | Cadey | true |
| 09:46.16 | WIMPy | Yes, but 00 isn't valid everywhere. |
| 09:46.19 | Cadey | but it was my understanding + was supposed to be sent so local coms system insert the correct + |
| 09:47.04 | wdoekes2 | you're right WIMPy, but you're only sending it to your itsp |
| 09:47.09 | wdoekes2 | so it either works or it doesn't |
| 09:47.22 | WIMPy | From what I've been told there is some sort of standard for transmitting numbers that forbids the +, but everone sane enough, uses it anyway. |
| 09:47.41 | Cadey | oh right :) |
| 09:47.57 | Cadey | nothing like a multi-national-non-standard to mess things up :D |
| 09:47.57 | WIMPy | Sure, but it would be nice if you can use the same caller ID if you try to do direct IP calls. |
| 09:49.14 | WIMPy | One of the many points where you can see very clearly that SIP wasn't made for VOIP. |
| 09:49.28 | wdoekes2 | yeah.. but often you need to do specific translations for specific itsp's anyway.. I rewrite stuff at the last moment, keeping it canonical until it goes out |
| 09:50.00 | WIMPy | Yes, unfortunately :-( I have to do the same. |
| 09:52.31 | Cadey | arrr my carrier ask for nothing to be sent before the number |
| 09:52.34 | Cadey | and they work it out |
| 09:52.58 | Cadey | so just 448444128704 with no + or double 00 |
| 09:53.48 | WIMPy | Ja, that's the officially correct version, or so I've been told. |
| 09:54.14 | WIMPy | But it's impossible form that to see it it's really in international format. |
| 09:58.14 | *** join/#asterisk jblack (~jblack@pool-71-181-219-109.sctnpa.east.verizon.net) |
| 10:00.33 | wdoekes2 | A tel URL usually consists of the number in E.164 format preceded by a plus sign, e.g.,: tel:+12025332600. This format is so useful that it has been incorporated into the baseline SIP specification; the user portion of a SIP URI can contain a tel URL (without the scheme string, like sip:+12025332600@carrier.com;user=phone). |
| 10:00.43 | wdoekes2 | (from random rfc 3824) |
| 10:01.12 | WIMPy | Too many SIP RFCs :-) |
| 10:01.20 | wdoekes2 | hehe, indeed |
| 10:01.54 | WIMPy | I guess that should have been ":-(" |
| 10:02.51 | WIMPy | But yes, tel:-URIs are clear on that. |
| 10:03.20 | WIMPy | Good to see it found it's way to SIP. |
| 10:03.27 | WIMPy | its |
| 10:04.00 | kaldemar | http://tools.ietf.org/html/rfc5411 |
| 10:09.46 | WIMPy | Damn, that's a real shitload of stuff :-( |
| 10:10.44 | kaldemar | i pity the fool who wants to "know SIP". :) |
| 10:15.35 | Cadey | I dont know, ie read rfc's which are much longer than 5411 :) |
| 10:16.31 | dimm | please say some words related to silence when i try to dialing via asterisk via dahdi |
| 10:16.56 | Cadey | ? |
| 10:16.58 | dimm | i'm not hearing opponent, but i hearing tone signals when my opponent hang down |
| 10:19.31 | kaldemar | Cadey: well, i meant knowing all rfc's that are referred to in 5411 inside out. :P |
| 10:19.59 | Cadey | haha yeah good luck with that :) |
| 10:20.16 | Cadey | I think that knowledge is reserved for the CCIE's of this world :) |
| 10:20.59 | Cadey | im fairly certain any CCIE would be able to bore you to death with millions of rfc definitions |
| 10:21.15 | WIMPy | And then some people call H323 complicated. |
| 10:22.32 | kaldemar | the "H.323 is complicated"-mantra was born when SIP was still pretty simple, and it stuck. |
| 10:22.49 | WIMPy | ja |
| 10:23.15 | WIMPy | simple and useless |
| 10:23.17 | Cadey | ok time for a simple question :) |
| 10:24.22 | Cadey | I have a queue which i send the r so it rings but it uses the Yank ringing tone. Can I pass anything to the queue app or set somthing in the queues context to define another ring type? |
| 10:24.39 | coppice | SIP is a perfect example of "confidence is a feeling you have before you fully understand the problem" |
| 10:24.49 | Cadey | lol |
| 10:25.32 | WIMPy | :) |
| 10:26.56 | *** join/#asterisk codefreeze-lap (~Steve_Mur@mail.parsetree.com) |
| 10:28.23 | coppice | it took a monumental level of arrogance by the SIP guys to believe that 100 years of make phone calls work had taught the ITU guys nothing when they put H.323 together |
| 10:29.19 | Cadey | yeah sorry did anyone see my simple queue question because google is comming up short :) |
| 10:29.26 | *** join/#asterisk codefreeze-lap (~Steve_Mur@mail.parsetree.com) |
| 10:30.32 | WIMPy | is always cutting in line |
| 10:31.48 | Cadey | im starting to think the only way is to set the MOH class to play the country of choices ringing tone...?? |
| 10:31.52 | *** join/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 10:34.07 | dimm | is this line saying that 'from-internal' application is running? |
| 10:34.09 | dimm | -- Executing [84952230100@from-internal:1] Macro("SIP/117-0000009c", "user-callerid,SKIPTTL,") in new stack |
| 10:35.03 | WIMPy | No, it says that Macro is running for a call in the context from-internal. |
| 10:35.05 | kaldemar | dimm: it says that channel SIP/117-0000009c is in context from-internal, extension 84952230100 and priority 1. |
| 10:35.18 | WIMPy | So I smell freepbx there? |
| 10:35.36 | skrusty | morning all |
| 10:40.58 | Cadey | Arg why there is no option to set the ringer type for the queue :( |
| 10:41.19 | Cadey | can it be done with a channel varible? |
| 10:41.39 | Cadey | Set(CHANNEL(ringingtone)=UK) for example |
| 10:42.13 | WIMPy | IIRC the language changes tones as well. |
| 10:42.16 | dimm | guyz! |
| 10:42.44 | dimm | when i dialing to dahdi, i not hear sound. But Mixmonitor record sound fine. What problem? |
| 10:43.58 | Cadey | WIMPy : like this ? Set(CHANNEL(LANGUAGE)=UK) ?? |
| 10:44.37 | WIMPy | Cadey: Yes |
| 10:45.02 | WIMPy | *IF* I remember correctly that it has an influence. |
| 10:45.21 | Cadey | im not sure i need the CHANNEL() part?> |
| 10:45.31 | Cadey | as LANGUAGE() is a channel variable anyway right ? |
| 10:46.58 | WIMPy | LANGUAGE() would be a function, but that does not exist. |
| 10:47.15 | Cadey | oh ok |
| 10:47.20 | Cadey | is new to dialplan syntax :) |
| 10:48.12 | Cadey | Set(CHANNEL(LANGUAGE)=uk) <-- still rings yank style :( |
| 10:51.40 | WIMPy | CHANNEL(tonezode) |
| 10:51.47 | Cadey | =uk? |
| 10:51.49 | WIMPy | CHANNEL(tonezone) |
| 10:52.12 | Cadey | trying... |
| 10:52.13 | WIMPy | yes |
| 10:52.23 | Cadey | result! |
| 10:52.37 | Cadey | thanks WIMPy :) |
| 10:54.18 | saxa | hi ppl, anybody can help me with understand why when I call from one box to another box connected via iax the called box answers with congestion ? |
| 10:54.24 | saxa | http://pastebin.com/bS84BiYp |
| 10:55.28 | kaldemar | saxa: what makes you think the other box answers? set iax debug on to see if you even get an answer. |
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| 10:58.37 | saxa | kaldemar: hmm, i tought it answers with congestion, but you are probably right |
| 10:58.46 | saxa | let me try to set debug on |
| 10:59.10 | kaldemar | saxa: nothing in your paste says that the other end answers in any way. |
| 11:12.31 | saxa | kaldemar: http://pastebin.com/WJF0sPv8 |
| 11:12.43 | saxa | thats what i get with debug on |
| 11:14.43 | kaldemar | saxa: the originating side doesn't know how to authenticate. you dial with IAX2/xxx.xxx.95.245/900, which would require a peer by the name of xxx.xxx.95.245 in iax.conf with the correct credentials. |
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| 11:17.25 | saxa | ok , but I have type=friend in iax.conf |
| 11:17.53 | saxa | kaldemar: thx, this means something is wrong with iax.conf , will recheck that |
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| 11:30.58 | dimm | please answer to me what is it? |
| 11:31.00 | dimm | exten => _849[59]X.,n,Macro(dialout-trunk,2,9${EXTEN},) |
| 11:31.24 | dimm | am i right that 2,9 it is a options for Macro ? |
| 11:33.39 | mzb | dimm: your question doesn't make sense |
| 11:34.01 | dimm | mzb, can you recognize my line? |
| 11:34.28 | mzb | I can recognise it ... I have no idea what your question is |
| 11:35.17 | *** join/#asterisk Pranav_rcmas (~prp_rcmas@117.193.203.253) |
| 11:38.37 | dimm | i try to say another words |
| 11:39.10 | dimm | is 2,9 it is a parameteres for macros dialout-trunk ? |
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| 12:27.47 | mzb | err ... perhaps you would like the answer: "2" and "9${EXTEN}" are params for the dialout-trunk macro |
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| 12:59.09 | asterisk-learner | if asterisk (1.4.36) is already running and i type "asterisk start" instead of "asterisk restart" I get "Asterisk died with code 1 ..." |
| 12:59.17 | asterisk-learner | is this an expected behiavor ? |
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| 13:13.42 | Tozz_ | yes |
| 13:13.48 | Tozz_ | because you're starting it twice |
| 13:13.51 | Tozz_ | and that wont work |
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| 13:21.11 | Dovid | hello all |
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| 13:45.33 | zamba | what do you guys think about microsoft's lync? |
| 13:46.23 | zamba | i know, not the best way of starting a conversation.. by swearing :) |
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| 13:55.31 | leifmadsen | zamba: never heard of it, what does it do? |
| 13:55.49 | leifmadsen | ah small communications server it seems... |
| 13:56.34 | leifmadsen | I doubt many people here have tried it |
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| 14:03.25 | anonymouz666 | nat/digium |
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| 14:12.10 | atan2 | Is there a setting for voicemail.conf that will send the person directly to the next message after they press '7' to delete? |
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| 14:20.05 | asterisk-learner | Tozz_: ok thx |
| 14:25.41 | raden | leifmadsen, you have any idea whats the issue with Music on hold and asterisk phones in 1.8 ? when I put someone on hold there is no music PAP2T and polycoms work |
| 14:26.32 | raden | leifmadsen, sip debug of when i put someone on hold http://pastebin.com/F09KSXwe |
| 14:27.10 | raden | This seems to be somewhat of a known issue |
| 14:34.40 | raden | this makes me thing real hard about ever buying a aastra phone again :( |
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| 14:45.46 | SuPrSluG | atan2: there is a setting for that. |
| 14:47.03 | jaytee | raden, is there a silence suppression option in the Aastra config? I've run into problems with MoH not working on another brand of phone due to that. |
| 14:47.23 | raden | ill try that |
| 14:47.42 | jaytee | I've also seen it referred to as "comfort noise" |
| 14:48.14 | raden | lemee try that jaytee |
| 14:48.20 | jaytee | but I don't think the two are actually the same thing. |
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| 14:49.37 | raden | never know |
| 14:49.42 | raden | this is soooo stupid |
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| 14:49.53 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:50.29 | raden | nope |
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| 15:41.38 | jaytee | I've started getting echo on some external calls lately. All my external inbound/outbound calls are via SIP using Flowroute as my ITSP. I don't have any TDM cards in my * server. Not sure what to look at to troubleshoot this. Can anyone advise me on this? |
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| 15:49.55 | GTXComm | Quesion in SIPURI, if I allowguest=yes , does this make me vonerable ... I have several contexts for inbound routes. includes like from-providera and from-provider-b, so I thought about including context = from-sipuri, and having ,,in AEL,, john => jump 102@context; }; |
| 15:50.20 | GTXComm | ? |
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| 15:54.49 | GTXComm | Or, I can allowguests on each sip account in the sip.conf? |
| 15:55.13 | GTXComm | I am not understanding the security risks for allowing guests. |
| 15:58.13 | skrusty | afternoon |
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| 16:00.16 | jkroon | is there anybody here that understands how "flash" events functions over SIP? |
| 16:01.15 | jkroon | the gigaset A580IP seems to generate some kind of flash event resulting in * warning about chan_sip.c: Don't know how to indicate condition 9. |
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| 16:01.20 | *** join/#asterisk eugeneoden (~goden@conference/pycon/x-snlspwodfndtkpyf) |
| 16:01.52 | jkroon | this happens when pressing the R button, which is supposed to initiate a transfer or something, in a similar way the hookflash on an analog system would from what I understand. |
| 16:03.38 | *** join/#asterisk p3nguin (~xwQ5kwYl6@24-171-79-245.dhcp.mtvr.il.charter.com) |
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| 16:10.11 | GTXComm | I will just implement it afterhours, and test. LOL thanks. |
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| 16:18.47 | skrusty | :] |
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| 16:27.33 | shamelessn00b | ~itsp-us |
| 16:27.52 | shamelessn00b | ~itsplist-us |
| 16:27.52 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
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| 16:45.30 | p3nguin | I don't remember who it was that I was talking to the other day about the Polycoms, but that ebay listing we found had the price increased $15. I hope you bought all of yours before the price increased. |
| 17:03.23 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
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| 17:07.27 | raden | p3nguin, you have any idea why music on hold would not work with aastra phones on 1.8.2 and 1.8.3 all newest firmware |
| 17:09.39 | raden | polycom and Pap2t work |
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| 17:25.38 | *** join/#asterisk KC9NVQ (~andrewk@funtoo/staff/trelane) |
| 17:25.56 | KC9NVQ | I need a sip device with a loud klaxon ringer and strobe or flashing light (for use in a very loud datacenteR) |
| 17:25.59 | KC9NVQ | suggestions? |
| 17:30.13 | n3hxs | Triggered by ringing? |
| 17:30.20 | n3hxs | KC9NVQ, ? |
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| 17:31.09 | KC9NVQ | n3: I was going to put it in a dial() statement with the actual phone at the DC |
| 17:31.18 | KC9NVQ | we can't hear it over the chiller fans |
| 17:32.06 | n3hxs | There are lots of them which have loud bells and strobe triggered by ringing. |
| 17:33.52 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 17:34.48 | KC9NVQ | n3hxs: any suggestion as to which? |
| 17:35.40 | n3hxs | No, but I see them from $25 up on Google. |
| 17:36.17 | n3hxs | As I recall, those are all cheaper than the one Radio Shack used to carry. |
| 17:43.09 | p3nguin | Radio Shack sold SIP phones with strobes? I have my doubts about that... I used to work for Radio Shack. |
| 17:43.44 | p3nguin | Now if you said they sell batteries... |
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| 17:47.48 | KC9NVQ | p3nguin: they sure don't sell radios |
| 17:48.00 | KC9NVQ | n3hxs I'm looking for sip specifically |
| 17:48.32 | p3nguin | We used to sell radios of all sorts. We had FM radios, 2-way radios, etc. |
| 17:50.03 | p3nguin | I worked for a Radio Shack dealer, so we had Radio Shack products plus other brands that you don't find in the company stores. |
| 17:50.24 | n3hxs | doesn't know about SIP Strobe Lights. :) |
| 17:51.28 | n3hxs | worked for RS before too, until they started playing with the bonus for managers... Exit, Stage Left. |
| 17:52.17 | p3nguin | That's something else that we never had to deal with in a non-company store. |
| 17:53.17 | p3nguin | No corporate bullshit at all. |
| 17:54.42 | n3hxs | I enjoyed winning the cruse to the Bahamas. |
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| 18:39.30 | thews | Does anyone know if a modern polycom config generator exists? |
| 18:39.55 | thews | I've been trying to do it manually, and keep running into problems |
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| 19:02.57 | *** join/#asterisk ssureshot (~digitolx@12.168.79.178) |
| 19:05.34 | ssureshot | creating a new astrisk server with an astribank usb card,, I'm getting the following error when I restart dahdi "No hardware timing source found in /proc/dahdi, loading dahdi_dummy" |
| 19:06.06 | ssureshot | all the packages are the latest and compiled from source |
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| 19:11.09 | p3nguin | That doesn't look like an error to me. |
| 19:11.47 | p3nguin | It looks more like a notice. It says it couldn't find a hardware timing source, so it is loading dahdi_dummy. Problem solved. |
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| 19:30.30 | ssureshot | p3nguin: dahdi isn't seeing any spans on my t1 |
| 19:31.15 | ssureshot | t1 is up and functioning,, just trying to configure a new box |
| 19:31.22 | carrar | h4X0r |
| 19:31.56 | ssureshot | they are in bad need of upgrades |
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| 19:46.20 | nny | what's the preferred method for sending CDR data to two databases locally? |
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| 20:03.04 | nny | nm found it http://www.mail-archive.com/asterisk-users@lists.digium.com/msg210511.html |
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| 20:26.14 | *** join/#asterisk moos3 (~rgenthner@cpe-72-224-105-166.maine.res.rr.com) |
| 20:27.07 | moos3 | if i have exten => horizons,n,Read(optcode,,1) followed by exten => horizons,n,GotoIf($["optcode" == "1"]?vmb) should only go to vm if the user presses 1 right ? |
| 20:28.14 | Qwell | moos3: No, it would only go to vm if the string "optcode" is equal to the string "1" |
| 20:28.19 | Qwell | (hint: it never will be) |
| 20:28.38 | moos3 | aww |
| 20:28.42 | Qwell | (${} is your friend) |
| 20:29.02 | moos3 | ${optcode} makes a strin g? |
| 20:29.11 | Qwell | ${optcode} references the variable optcode |
| 20:29.15 | moos3 | aww |
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| 20:30.01 | moos3 | Qwell: so like this horizons,n,GotoIf($[${optcode} == 1]?vmb |
| 20:30.18 | Qwell | that will fail if the user pressed nothing |
| 20:30.47 | moos3 | so as long as they press 1 it will take them to vmb which is all i'm after |
| 20:30.50 | moos3 | so cool thanks |
| 20:30.58 | Qwell | no, it will fail to execute |
| 20:31.11 | Qwell | wrapping it in quotes will work fine |
| 20:31.43 | moos3 | which part should wrap in quotes |
| 20:31.49 | Qwell | just as you had it |
| 20:32.07 | Qwell | "${optcode}" == "1" |
| 20:32.12 | moos3 | k |
| 20:32.20 | moos3 | thats what i thought you ment |
| 20:32.37 | Qwell | or the old sh script trick: x${optcode} == x1 |
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| 20:39.14 | *** join/#asterisk Jouva (Jouva@fluffy.moufette.com) |
| 20:40.11 | Jouva | So I'm trying to get video calling to work through asterisk. Audio works just fine. But the video doesn't, and it seems like it's a nat issue as an error pops up in the console about being unable to transmit to a 192.168.x.x IP |
| 20:40.21 | Kobaz | can you do xml applications on the cisco spa phones without needing a cisco server? |
| 20:40.53 | Jouva | But I'm confused as to why audio would work two way just fine but video wouldn't |
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| 20:43.28 | _Corey_ | Kobaz: Yeah, there are third-party apps out there |
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| 20:45.37 | Kobaz | k |
| 20:45.58 | Kobaz | it was a little disheartening to see in the specs 'xml applications when using cisco blah blah with the blah blah cisco system' |
| 20:46.26 | _Corey_ | Kobaz: http://www.voip-info.org/wiki/view/Asterisk+Cisco+79XX+XML+Services |
| 20:46.41 | Kobaz | not the 7900 but that's good to know |
| 20:46.47 | Kobaz | i was talking about ther spa5xx |
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| 20:47.06 | _Corey_ | ahh... not sure about those :( |
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| 21:06.05 | KNERD | What is the libraries missing for this compile error? http://pastebin.com/RxZaS7vn |
| 21:06.54 | pabelanger | KNERD: what OS? |
| 21:07.34 | KNERD | CentOS |
| 21:07.49 | KNERD | maybe libdbi? |
| 21:08.06 | KNERD | no |
| 21:10.17 | *** join/#asterisk LemensTS (~matthew@70.238.136.43) |
| 21:10.49 | LemensTS | http://pastebin.com/hj9BE9F2 <-- how do I delete an item from an array? I am css dumb |
| 21:10.55 | pabelanger | KNERD: are you building res_ael_share.c |
| 21:11.30 | LemensTS | or filter getElementsByTagName to not grab input.hidden |
| 21:11.32 | KNERD | pabelanger: I really do not know |
| 21:12.33 | KNERD | LemensTS: you do not "delete" from an array. you either change the value to NULL or remove a reference from it |
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| 21:15.10 | KNERD | pabelanger: looking now |
| 21:18.42 | Katty | leif |
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| 21:26.50 | KNERD | pabelanger: i just don;t know..i do not see any reference to it |
| 21:27.24 | pabelanger | KNERD: what does menuselect display? |
| 21:27.51 | KNERD | pabelanger: i have been looking at it, but I cannot locate it so far |
| 21:28.43 | KNERD | got it...1 sec |
| 21:29.15 | KNERD | pabelanger: yes, found it..it is checked |
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| 21:46.41 | leifmadsen | Katty |
| 21:47.14 | Qwell | leifmadsen |
| 21:47.20 | leifmadsen | Qwell |
| 21:47.23 | Qwell | hi |
| 21:47.26 | leifmadsen | o/ |
| 21:48.11 | thehar | Katty |
| 21:48.32 | Qwell | Harley. |
| 21:48.40 | thehar | ... |
| 21:48.41 | thehar | hides |
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| 21:49.29 | jaytee | Elvis |
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| 21:58.44 | Katty | hugs Qwell |
| 21:58.47 | Katty | hugs leifmadsen |
| 21:58.49 | Katty | hugs jaytee |
| 21:58.51 | Katty | hugs thehar |
| 21:58.57 | Qwell | yay first! |
| 21:59.00 | Katty | afternoon gentlemen |
| 21:59.01 | Qwell | take THAT |
| 21:59.05 | Katty | Qwell: you owe me an update -> |
| 21:59.17 | Qwell | new car! |
| 21:59.23 | Katty | no i heard about the car, dear |
| 21:59.27 | Qwell | I know :p |
| 21:59.27 | thehar | hugs Katty |
| 21:59.29 | Katty | i meant about your health |
| 22:00.35 | jaytee | I didn't hear about the car! what kind? |
| 22:01.07 | Katty | white nissian i believe? |
| 22:01.20 | Qwell | black, altima |
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| 22:01.29 | Katty | ah, black. |
| 22:01.33 | Katty | i wonder where i got white from ^_- |
| 22:01.47 | Qwell | it's all shiny and new |
| 22:01.58 | Katty | you should post a photo of it |
| 22:01.59 | jaytee | :-) |
| 22:02.00 | ClintGoudie-Nice | Heya all, does anyone know why a TLS sip trunk that my asterisk is connecting to would say: == Problem setting up ssl connection: error:14094413:SSL routines:SSL3_READ_BYTES:sslv3 alert unsupported certificate |
| 22:02.24 | ClintGoudie-Nice | how can I figure out what about the certificate is unsupported? |
| 22:02.43 | jaytee | I bought one of those Trek symbols that look like the Christian fish symbol for the back of my new-to-me used car. |
| 22:03.32 | jaytee | that and the ThinkGeek.com license plate frame that says "Will work for bandwidth" really makes the buggy look fly. |
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| 22:05.55 | leifmadsen | pretty fly for a white guy |
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| 22:12.50 | jaytee | :-) |
| 22:13.25 | KNERD | who knows what libraries I am missing for this compile error? http://pastebin.com/vSUNsN18 |
| 22:18.11 | Qwell | KNERD: what version of Asterisk? |
| 22:18.25 | KNERD | 1.8.3 on CentOS 5.5 |
| 22:18.39 | Qwell | probably res_crypto |
| 22:20.08 | Qwell | jaytee: damn you. now I have to buy thing or two from thinkgeek |
| 22:20.21 | KNERD | Qwell: res_crypto is available |
| 22:28.54 | *** part/#asterisk LemensTS (~matthew@70.238.136.43) |
| 22:32.32 | pabelanger | KNERD: libssl-dev? |
| 22:34.18 | KNERD | pabelanger: no such libraries that I can find |
| 22:34.36 | pabelanger | openssl development |
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| 22:35.56 | KNERD | pabelanger: already installed |
| 22:36.14 | pabelanger | pb your config.log |
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| 22:43.52 | KNERD | pabelanger: it's a bit big...maybe I should delete it and try combiling again |
| 22:49.26 | KNERD | pabelanger: make a backup of the config to log file and make a new one with..it;s also over 1MB in size |
| 22:49.32 | KNERD | how much you want of it? |
| 22:57.13 | *** part/#asterisk ClintGoudie-Nice (~clint@smtp.callware.com) |
| 22:57.41 | jmordica | tried to use remote backup options in freepbx 2.8 and it won't restore from the other freepbx server.. i've copied ssh key |
| 22:57.51 | jmordica | I know it's working somewhat because the primary server cpu usage goes up to like 50% for a couple of seconds when I hit submit but no data transfers.. |
| 22:59.10 | jmordica | I've asked this question in freepbx but noone responds.. i have no idea. Obviously the second box is hitting up the first box for it;s files, the first box just doesn't want to hand them over |
| 22:59.23 | Jouva | So I'm trying to get video calling to work through asterisk. Audio works just fine both ways with clients behind NAT and server w/ a static IP. But the video doesn't, and it seems like it's a NAT issue as an error pops up in the console about being unable to transmit to a 192.168.x.x IP |
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| 23:31.31 | *** part/#asterisk nny (~Scott_2@174.107.201.103) |
| 23:44.23 | vfabi | hi all |
| 23:44.27 | vfabi | I have found problem using lua + luasql.postgres. Simple db request script (http://pastebin.com/eNzCS3gA) get an error "ERROR[12282]: pbx_lua.c:1090 lua_get_state: Error loading extensions.lua: error loading module 'luasql.postgres' from file '/usr/local/lib/lua/5.1/luasql/postgres.so': dynamic libraries not enabled; check your Lua installation" |
| 23:44.49 | vfabi | how to fix it ? |
| 23:45.42 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 23:46.36 | vfabi | if it run from script it's ok, if it's run from extensions.lua - error |
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| 23:48.56 | LemensTS | http://www.voip-info.org/wiki/view/Asterisk+cmd+Milliwatt Anyone use the PlayTones(1004/1000) to tune tx/rx gain on dahdi, or do you call out to a test line? |
| 23:49.29 | LemensTS | I assume you call into asterisk from an outside phone into that |
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