| 00:02.26 | *** join/#asterisk learath (47f6db23@gateway/web/freenode/ip.71.246.219.35) |
| 00:04.59 | Kobaz | so i have an asterisk behind a nat |
| 00:05.05 | Kobaz | externip is set, localnet is set |
| 00:05.19 | Kobaz | i can do like voicemailmain on a remote phone that's registered |
| 00:05.37 | Kobaz | and then after that i'll get: [2012-05-31 20:04:09] WARNING[576]: chan_sip.c:3641 retrans_pkt: Retransmission timeout reached on transmission a8ab3d3a-e7b644a3-4bdb5c94@10.0.0.117 for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions |
| 00:05.42 | Kobaz | and then the call is cut |
| 00:05.48 | Kobaz | [2012-05-31 20:04:09] WARNING[576]: chan_sip.c:3670 retrans_pkt: Hanging up call a8ab3d3a-e7b644a3-4bdb5c94@10.0.0.117 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). |
| 00:10.11 | *** join/#asterisk volga629 (~volga629@76-10-130-18.dsl.teksavvy.com) |
| 00:10.52 | volga629 | Good Day, I am getting this error when asterisk try modify ldap because Object class violation |
| 00:11.09 | volga629 | any help thank you |
| 00:11.19 | volga629 | asterisk 1.8.11 |
| 00:13.19 | *** join/#asterisk svm_invictvs (~patrick@unaffiliated/svminvictvs/x-938456) |
| 00:13.21 | svm_invictvs | Heya |
| 00:13.32 | svm_invictvs | My asterisk install isn't opening up a port for SIP to listen |
| 00:13.37 | svm_invictvs | I have this in my sip.conf |
| 00:13.50 | svm_invictvs | <PROTECTED> |
| 00:13.50 | svm_invictvs | <PROTECTED> |
| 00:13.51 | svm_invictvs | <PROTECTED> |
| 00:14.38 | WIMPy | And probably a fatal error somewhere else. Does chan_sip get loaded? |
| 00:15.54 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 00:16.53 | svm_invictvs | [May 31 17:09:31] NOTICE[28630] chan_sip.c: The 'username' field for sip peers has been deprecated in favor of the term 'defaultuser' |
| 00:17.01 | svm_invictvs | I'ms eeing that, but would it keep it from working all together? |
| 00:17.24 | WIMPy | no |
| 00:18.13 | svm_invictvs | I suspect that if chan_sip is throwing warnings it's loaded, but indeed the port isn't opening :( |
| 00:18.44 | WIMPy | 'module show like sip' |
| 00:19.58 | svm_invictvs | Module Description Use Count |
| 00:20.01 | svm_invictvs | chan_sip.so Session Initiation Protocol (SIP) 0 |
| 00:20.05 | svm_invictvs | app_adsiprog.so Asterisk ADSI Programming Application 0 |
| 00:20.10 | svm_invictvs | hm... |
| 00:20.19 | svm_invictvs | That file's gettng loaded |
| 00:20.53 | svm_invictvs | But no tused. |
| 00:22.21 | svm_invictvs | What else... |
| 00:22.59 | WIMPy | 'sip show settings' |
| 00:23.10 | *** join/#asterisk philipp64|laptop (~chatzilla@66-232-81-197.ctcweb.net) |
| 00:24.16 | philipp64|laptop | I'm running 1.8.6.0.... I'm talking to a switch that requires a Diversion header... Is there a way to generate this header through channel variables without having to call SIPAddHeader() explicitly? |
| 00:25.10 | WIMPy | To my knowledge that is not supported in any other way. |
| 00:27.20 | philipp64|laptop | yow... requiring users to know the format of SIP messages... that's painful. |
| 00:27.43 | WIMPy | You get used to it. |
| 00:28.40 | philipp64|laptop | and we're requiring them to memorize a now obsolete spec, on top of that. |
| 00:30.08 | svm_invictvs | ugh |
| 00:30.18 | svm_invictvs | It's enabled ccsp, but, not SIP |
| 00:30.50 | svm_invictvs | TCP bind address disabled |
| 00:30.51 | svm_invictvs | hm |
| 00:31.11 | philipp64|laptop | what's worse, our Taqua doesn't accept Remote-Party-Id: but requires P-Asserted-Identity: instead with a Diversion header. |
| 00:31.39 | svm_invictvs | er, sccp |
| 00:31.46 | WIMPy | Well, PAI seems to be the usual way. |
| 00:31.57 | philipp64|laptop | anyone notice if there's a way to detect that a call into the AT&T wireless network has gone to voice mail? |
| 00:32.26 | philipp64|laptop | does the CONNECTEDLINE state change or do we get a second "progressing" message or anything like that? |
| 00:32.31 | WIMPy | You might get a diversion information. |
| 00:32.36 | philipp64|laptop | this is over SIP, I mean. |
| 00:32.58 | philipp64|laptop | assuming the SS7/SIP gateway doesn't drop any messages... |
| 00:33.14 | WIMPy | Unlikely that the information survives translation to SIP. |
| 00:33.34 | WIMPy | That sounds like a very unlikely assumption. |
| 00:33.59 | philipp64|laptop | I thought SIP/SS7 interoperability was supposed to be pretty good... |
| 00:34.09 | philipp64|laptop | oh, well. |
| 00:34.41 | philipp64|laptop | during a find-me/follow-me i want to terminate the 2nd leg if I can detect that my call is going into AT&T voicemail. |
| 00:34.48 | WIMPy | Err. I don't even think it's possible, but I woudn't even expect the possible to have been done. |
| 00:34.56 | philipp64|laptop | that is, retain control and have it go to my own voicemail instead. |
| 00:35.33 | WIMPy | So you are forwarding to your mobile? |
| 00:36.17 | philipp64|laptop | I start by ringing on 3 phones.... then after 5 seconds add an additional outbound Dial() to my cell phone to the channel. |
| 00:36.21 | philipp64|laptop | err... to the call rather. |
| 00:36.38 | philipp64|laptop | but I don't want the cell's voice mail answering. |
| 00:36.41 | WIMPy | ON SS7 you could use CDO. |
| 00:36.58 | WIMPy | What about disabling the VM? |
| 00:37.55 | philipp64|laptop | CDO? |
| 00:38.07 | WIMPy | Call Diversion Override |
| 00:38.28 | philipp64|laptop | I want my voice mail to answer on direct calls to my cell #. |
| 00:38.31 | WIMPy | Assuming they do it via CF. |
| 00:38.41 | philipp64|laptop | what about Max-Hops? |
| 00:39.14 | WIMPy | Yes, that should be another option. |
| 00:39.26 | WIMPy | Didn;t think of that. I'm not that in to SS7. |
| 00:40.18 | philipp64|laptop | again, assuming that the Max-Hops interoperates into SS7 vocabulary. |
| 00:40.46 | WIMPy | No idea WHAT IT'S CALLD, BUT SOMETHIGN LIKE THAT EXISTS. |
| 00:40.47 | philipp64|laptop | errr... max-forwards rather. |
| 00:40.51 | WIMPy | oops |
| 00:41.01 | *** join/#asterisk sean-w (4118d64c@gateway/web/freenode/ip.65.24.214.76) |
| 00:42.22 | sean-w | I think RoadRunner is blocking incoming SIP. Has anyone else had a similar experience? |
| 00:51.45 | talntid | philipp64|laptop... |
| 00:51.52 | talntid | I did something like what you describe |
| 00:52.07 | philipp64|laptop | oh, yes? and how did it go? |
| 00:52.19 | talntid | basically... |
| 00:52.45 | talntid | instead of actually rining the phone inside the meetme, use another channel, AMD() and drop it if AMD = Machine, merge if HUMAN |
| 00:53.00 | talntid | I didn't finish it, as I didn't need to, but that's how I was making it work |
| 00:54.14 | talntid | basically, just open the additional outbound for the cell phone.. and if it goes HUMAN, join the meetme. |
| 00:54.19 | talntid | make sense? |
| 00:55.24 | talntid | I did it for an alarm system notification |
| 00:55.54 | WIMPy | But you have to pay for the VM you don't want. |
| 00:56.13 | talntid | nah, for the VM, just hang up. don't play anything |
| 00:56.38 | talntid | when it detects machine, just hang up |
| 00:56.46 | WIMPy | Yes, but you're already connected. |
| 00:56.56 | talntid | in the USA, that doesn't cost anything |
| 00:57.05 | talntid | if there is no VM message, no minutes are used |
| 00:57.13 | WIMPy | Free minutes to VM? |
| 00:57.17 | talntid | yeah |
| 00:57.23 | WIMPy | Nice |
| 00:57.26 | talntid | if you call your VM box, it costs your minutes.. |
| 00:57.35 | talntid | but it doesn't cost while someone is leaving you a VM |
| 00:57.36 | WIMPy | That's the big rip off here. |
| 00:57.49 | talntid | and if the system doesn't get a VM, you obviously won't call into VM |
| 00:58.32 | talntid | philipp64|laptop didn't respond. Maybe I misunderstood what he was wanting to do |
| 00:59.03 | WIMPy | doesn't think so |
| 00:59.36 | talntid | thinks philipp64|laptop lives near me |
| 01:01.37 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 01:02.18 | talntid | I'm working on a project.. |
| 01:02.25 | talntid | I spent 2 weeks writing it to spec |
| 01:02.32 | talntid | and it was perfect, exactly as spec.. |
| 01:02.43 | talntid | even spent some extra time to polish it a little bit.. |
| 01:03.11 | talntid | then, once customer saw it, they wanted me to re-do it in a different way, because once they saw how it worked, they realized the inherent issues with it |
| 01:03.19 | talntid | 14,000 lines of code.. trashed. lol :( |
| 01:05.38 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 01:09.47 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 01:09.47 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 01:12.58 | *** part/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 01:13.21 | philipp64|laptop | talntid: and that would be where? |
| 01:13.30 | talntid | huh? |
| 01:13.40 | talntid | what would be where? |
| 01:14.34 | philipp64|laptop | where do you live? I'm in Boise. |
| 01:14.51 | talntid | oh, Spokane :) |
| 01:15.18 | philipp64|laptop | closer as the crow flies than driving. :-) |
| 01:15.27 | talntid | yah, heh |
| 01:18.55 | WIMPy | LOL. Someone tries to sell phone books from 2005 on ebay. |
| 01:19.21 | carrar | cj |
| 01:19.25 | talntid | did my theory make sense, philipp64|laptop? |
| 01:19.30 | talntid | it should work for ya |
| 01:28.18 | philipp64|laptop | um, yeah... I just wish there were a cleaner way to do this that didn't rely on voice/machine detection working, etc. |
| 01:28.41 | philipp64|laptop | WIMPy: aren't those valuable archeological artifacts? |
| 01:29.17 | philipp64|laptop | one of these days I need to try to get SLA's working with SIP trunks and my Sipura 942's. sigh. |
| 01:31.34 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 01:34.04 | WIMPy | Pardon? |
| 01:34.24 | talntid | the phone bookls |
| 01:34.28 | WIMPy | Oh, the phone books. |
| 01:34.47 | WIMPy | I think they are a little too new to go as valuable. |
| 01:35.23 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 01:50.39 | *** join/#asterisk Beta2K (~Beta2K@d24-36-128-84.home1.cgocable.net) |
| 01:50.53 | Beta2K | Hello all |
| 01:51.06 | talntid | Hello |
| 01:53.09 | Beta2K | I'm having troubles with dundi |
| 01:53.50 | Beta2K | if I bring up dundi set debug on one side is receiving but the other isn't |
| 01:54.07 | Beta2K | This is across a vpn with no firewalling |
| 01:54.21 | Beta2K | both are transmitting packets |
| 01:54.38 | talntid | no idea, i'm a n00b |
| 01:55.50 | WIMPy | Are teh keys ok? |
| 01:56.08 | WIMPy | Try 'keys show'. |
| 01:57.49 | *** join/#asterisk serafie (~erin@75.76.38.159) |
| 01:58.45 | saliak | where can i find documentation for ReceiveFax() ? I'm working on my diaplan to receive faxes, but it seems to not execute all the commands after the fax reception, and i'm not sure why |
| 02:03.24 | Beta2K | Just did a tcpdump and looks like the packets _are_ getting clobbered somewhere... |
| 02:05.31 | WIMPy | Pretty interesting what some haxx0rs think you might use as a trunk prefix... http://wimpy.yeti.dk/pastebin |
| 02:16.29 | svm_invictvs | Sorry, Iwas asking earlier |
| 02:16.41 | svm_invictvs | But, I was curious, why would the logs tell me that there's no bind address for SIP |
| 02:16.49 | svm_invictvs | And secondly, how do I go about disabling Skinny/sscp? |
| 02:17.12 | WIMPy | Don't load it or don't even install it. |
| 02:17.57 | svm_invictvs | er, sccp |
| 02:18.07 | *** join/#asterisk HyperNerdV2 (~HyperNerd@cpe-98-149-122-251.socal.res.rr.com) |
| 02:18.13 | svm_invictvs | What configuration directive loads modules? |
| 02:18.13 | WIMPy | modules.conf |
| 02:18.50 | svm_invictvs | Ahhhh |
| 02:23.41 | svm_invictvs | Alright, added noload => chan_skinny.so, that worked |
| 02:23.46 | svm_invictvs | WIMPy: Thanks |
| 02:26.33 | svm_invictvs | Now, in my asterisk directory, is there an order in which Asterisk reads all the files? |
| 02:26.39 | svm_invictvs | Or does it just read through each *.conf file? |
| 02:27.41 | *** join/#asterisk ThinkGNU- (~ThinkGNU-@174-31-120-200.chyn.qwest.net) |
| 02:28.00 | WIMPy | The conf files are read by the various modules when they are loaded. |
| 02:28.04 | svm_invictvs | Heh, I wonder if I could write a module to integrate Asterisk with Ventrilo |
| 02:28.07 | WIMPy | except for asterisk.conf. |
| 02:28.24 | svm_invictvs | I see, so each module is hardcoed to look for a specific configuration file |
| 02:28.38 | WIMPy | yes |
| 02:28.48 | WIMPy | Or none or multiple. |
| 02:28.51 | svm_invictvs | Yeah |
| 02:28.58 | svm_invictvs | Specific file or files, rather |
| 02:31.14 | svm_invictvs | http://mysticpaste.com/private/VKo24s2y3J/ |
| 02:31.19 | svm_invictvs | Still not opening up a listen port |
| 02:31.21 | svm_invictvs | for SIP |
| 02:34.01 | *** join/#asterisk JuStIcIa_ (~JuStIcIa_@190.167.44.182) |
| 02:34.04 | WIMPy | Hmm. is bindaddr a valid option? |
| 02:34.48 | WIMPy | Yeah. Looks like I finally got an OpenHorst image. |
| 02:36.13 | svm_invictvs | WIMPy: According to voip-info.org and it's not complaining in the logs |
| 02:36.46 | WIMPy | cahn sip doesn't complain. It just silently fails. |
| 02:36.54 | svm_invictvs | heh |
| 02:37.01 | WIMPy | And better use wiki.asterisk.org. |
| 02:37.05 | svm_invictvs | WIMPy: I just realized I'm not configuring TCP on the server... |
| 02:37.15 | svm_invictvs | WIMPy: And I was telling the client ot use TCP, not UDP |
| 02:40.08 | svm_invictvs | ALright, it works. I can connect y SIP client... |
| 02:40.10 | svm_invictvs | HOWEVER |
| 02:40.16 | svm_invictvs | The client doesn't support voice... |
| 02:43.18 | svm_invictvs | Now, I wonder if I can configure sip logins using LDAP |
| 03:12.35 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-iroljbunzghyswwh) |
| 03:13.47 | svm_invictvs | Does Asterisk have a STUN server builtin? |
| 03:16.16 | jpsharp | No. |
| 03:17.24 | *** join/#asterisk d-_-b- (~d-_-b-@smtp.lowsellers.net) |
| 03:20.34 | svm_invictvs | WOuld SIP over TCP eliminate the need for STUN? |
| 03:36.19 | philipp64|laptop | ok, stupid question... |
| 03:37.30 | philipp64|laptop | I have an incoming call and I want to save some of the channel variables into the call... I tried doing a: Set(_SIPDOMAIN=${SIPDOMAIN}) in one context and then retrieving ${_SIPDOMAIN} in another context, but value I get is empty. |
| 03:37.35 | philipp64|laptop | what am I doing wrong? |
| 03:38.34 | philipp64|laptop | keeping in mind that most of my dialplan was written in 1.4.x and I'm surprised that any of it still works... |
| 03:41.29 | carrar | don't _ on retreiving |
| 03:41.54 | carrar | and 1 x _ is good only for one inhertience |
| 03:42.21 | carrar | __ is good till its over |
| 03:48.14 | jpsharp | svm_invictvs: No, because the RTP is still transported over UDP. |
| 03:49.34 | philipp64|laptop | carrar; ok, I'm confused... so do I: Set(__FOO=xyzzy) and then retrieve as ${_FOO} ? |
| 03:55.06 | svm_invictvs | If I sip show peers, and a peer is connected through STUN, does the STUN server's address sho, or the peer's? |
| 03:55.29 | carrar | no |
| 03:55.30 | carrar | Set(_FOO=xyzzy) retrieve as ${FOO} |
| 03:55.30 | carrar | Set(__FOO=xyzzy) retrieve as ${FOO} |
| 03:57.21 | jpsharp | svm_invictvs: You'll see the peer's. STUN doesn't act as a proxy. |
| 03:58.24 | svm_invictvs | oh |
| 03:59.46 | svm_invictvs | Ugh, does STUN need to be configured on both client/server? |
| 04:00.43 | jpsharp | No, just the client. |
| 04:01.17 | jpsharp | STUN just helps the client learn its globally routable IP. |
| 04:02.25 | svm_invictvs | oh |
| 04:03.43 | svm_invictvs | God damned wifi is so spotty |
| 04:06.33 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 04:24.05 | volga629 | might somebody know about this error ? res_config_ldap.c: Couldn't modify 'name'='', dn:uid=,ou=,dc=n,dc= because Object class violation |
| 04:31.12 | *** join/#asterisk saliak (~kailas@ip68-9-228-184.ri.ri.cox.net) |
| 04:31.20 | *** join/#asterisk bluregard (~matt@c-71-201-99-216.hsd1.il.comcast.net) |
| 04:31.27 | bluregard | good evening all |
| 04:32.34 | ChannelZ | people hi |
| 04:33.41 | jpsharp | yo |
| 04:39.18 | ChannelZ | (name that reference!) |
| 04:40.00 | slav3_kitten | ok, what's a good sub 150 (used) sip solution to wifi cordless phone? |
| 04:44.32 | carrar | Just buy a used SIP DECT Phone |
| 04:45.50 | slav3_kitten | < googling please wait |
| 04:46.20 | slav3_kitten | currently getting .1-100k/s down |
| 04:48.26 | slav3_kitten | oh nity |
| 04:48.47 | slav3_kitten | nifty* |
| 04:49.42 | drmessano | $15 DECT POTS phone + ATA works for me |
| 04:49.48 | drmessano | But then again, I am weird |
| 04:50.37 | volga629 | might somebody know about this error ? chan_sip.c: sip_xmit of 0x1a5049b0 (len 638) to ip:1053 returned -2: Success |
| 04:51.02 | slav3_kitten | drmessano, i'm mostly learning an playing around with stuff |
| 04:51.46 | drmessano | I dont condone enslaving kittens |
| 04:51.52 | drmessano | To each their own |
| 04:52.11 | slav3_kitten | eh |
| 04:56.03 | jpsharp | I've not run across a SIP wifi phone that was worth a crap. |
| 04:56.44 | slav3_kitten | sucks |
| 04:57.08 | slav3_kitten | i hear they are a pain in the ass to setup but seem to be widely used around doctors offices around here |
| 04:57.19 | jpsharp | And I tested half a dozen of them a while ago. They all had a combination of crap battery life, crap wifi coverage, and/or crap audio. |
| 04:57.27 | slav3_kitten | or so i'm told by a friend |
| 04:58.15 | jpsharp | Are they actual wifi phones or just regular cordless phones connected via a SIP adapter of somesort? |
| 04:59.01 | slav3_kitten | supposedly actual wifi phones, cisco 7921g's iirc |
| 04:59.06 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 04:59.37 | slav3_kitten | lots of cisco stuff around here. which is why my home network is entirely cisco an i got a number of cisco voip phones that can accept a sip firmware |
| 04:59.53 | jpsharp | That's one I didn't test cause they were too spendy. |
| 05:00.07 | jpsharp | But I didn't hear good things about it. |
| 05:00.12 | jpsharp | May have gotten better, though. |
| 05:00.30 | saliak | how do I change the address that emails originating in asterisk come from and/or the mua used to send them? right now, it's asterisk@mysystem.com. i'd like to use an address that I host remotely. i have that address all configured in postfix, but not sure how to tell asterisk to use it |
| 05:00.44 | drmessano | All SIP Wifi phones have battery and call quality issues. Its inherent with using wifi. Definitely not worth the price. |
| 05:01.30 | slav3_kitten | drmessano, the dect phones seem to be pretty nice from what i'm reading |
| 05:01.37 | jpsharp | I just have a couple of 900Mhz cordless phones attached to a SIP ATA> |
| 05:01.57 | jpsharp | Haven't ventured into DECT yet. |
| 05:01.59 | drmessano | DECT is much better.. I use Uniden DECT phones from Walmart and Linksys ATAs. lol |
| 05:02.13 | drmessano | $14.99 for the phones, I believe |
| 05:03.26 | jpsharp | I'll have to look. I mainly just went for the 900Mhz phones so they didn't stomp all over the wifi. I know DECT won't do that either, since they're at 1.6Ghz-ish. |
| 05:04.02 | *** join/#asterisk Dovid (~Dovid@213.8.121.90) |
| 05:04.13 | slav3_kitten | hmm *nods* |
| 05:04.38 | slav3_kitten | i want to go all shiny voip phones, hopefully all poe powered |
| 05:04.47 | slav3_kitten | because i utterly hate power bricks all over |
| 05:05.08 | jpsharp | mmmm wall warts |
| 05:05.15 | slav3_kitten | not to mention when everything is in a network closet, it's easier to backup battery an maintain |
| 05:05.35 | slav3_kitten | hence mirkotik poe access points |
| 05:05.46 | slav3_kitten | btw i <3 mikrotik |
| 05:06.20 | philipp64|laptop | I have Sipura 942's and a Linksys SFE300 switch... |
| 05:06.59 | philipp64|laptop | works well enough. plus it powers 2 MPEG4 cameras, and an HAI HTX IR flasher for the home theatre... |
| 05:07.38 | philipp64|laptop | I even have a remote Netgear 8-port GigE switch that's powered by POE. |
| 05:07.40 | slav3_kitten | i have a cisco 2621xm with nm-2v and a vic-2fxo voice card, various 7900 cisco phones, 2 mirktik access points, cisco 2900xl switch, cisco 3524 poe switch |
| 05:07.50 | slav3_kitten | the 3524 powers the phones |
| 05:08.09 | philipp64|laptop | you clearly have more budget than I do. :-) |
| 05:08.15 | slav3_kitten | because cisco is too good to make 803.2af poe phones in a price range i can afford :| |
| 05:08.26 | slav3_kitten | philipp64|laptop, all acquired over 5 or 6 years |
| 05:08.34 | slav3_kitten | and acquired dirt cheaply |
| 05:08.43 | philipp64|laptop | off the back of a truck? |
| 05:09.01 | jpsharp | off lease and "gone out of business" surplus. |
| 05:09.04 | philipp64|laptop | I've had most of my gear about the same amount of time, except the HTX and the Trendnet IP cameras. |
| 05:09.12 | slav3_kitten | ebay, and going out of business surplus |
| 05:09.44 | slav3_kitten | i really want to get some of those new ubiquiti ip cameras |
| 05:09.50 | slav3_kitten | but they are way out of my price range |
| 05:10.06 | slav3_kitten | also want to get my hands on the ubiquiti unifi gear one day |
| 05:10.40 | *** join/#asterisk HyperNerdV2 (~HyperNerd@cpe-98-149-122-251.socal.res.rr.com) |
| 05:11.09 | slav3_kitten | so yea, voip pbx is my new project/learning experience |
| 05:11.45 | jpsharp | Asterisk is pretty awesome. I've been using it for most than 10 years now. |
| 05:12.08 | slav3_kitten | that's fairly impressive jpsharp |
| 05:12.58 | slav3_kitten | i'm making my first headway into voip, and voip to pots. kinda want to see if i can get a decently priced UK did number so i can talk with friends there inexpensively an without skyping |
| 05:13.06 | slav3_kitten | which i'm told i can do |
| 05:13.20 | jpsharp | Yep, you can. |
| 05:13.48 | slav3_kitten | awesome |
| 05:13.53 | slav3_kitten | i loathe skype |
| 05:13.58 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 05:14.41 | jpsharp | I used to have a UK DID that I got from sipgate.co.uk |
| 05:15.09 | slav3_kitten | what was it roughly per minute and monthly base cost? |
| 05:16.42 | jpsharp | The number was free, incoming was free. |
| 05:16.47 | jpsharp | I never used the outgoing. |
| 05:16.57 | slav3_kitten | nice |
| 05:17.16 | slav3_kitten | i'm going to look it up when then storm lets up an my bandwidth returns |
| 05:17.28 | jpsharp | But it looks like the majority of their calls into the UK are 1.19p/min |
| 05:17.46 | jpsharp | Or 9.9p/min for calling mobile numbers. |
| 05:17.46 | slav3_kitten | so like 2 cents usd |
| 05:20.33 | jpsharp | Ohwait. The trick to getting the free number is that I had to sign up using an IP address from a UK ISP. |
| 05:21.11 | jpsharp | Er, free incoming number. Otherwise they give you a national rate number that costs a few p/min to call. |
| 05:21.15 | slav3_kitten | lol |
| 05:22.14 | jpsharp | But fortunately, I had access to a UK IP block over satellite. |
| 05:22.46 | slav3_kitten | that sounds troublesome |
| 05:22.55 | slav3_kitten | and expensive |
| 05:24.12 | jpsharp | Not especially, cause I was the head network engineer at said satellite provider. |
| 05:25.43 | slav3_kitten | awesome in that case |
| 05:26.00 | slav3_kitten | i'm currently attempting to get an education in some tech desk job after a work accident made it so i need crutches for the rest of my life |
| 05:26.14 | slav3_kitten | up side, if you ever get hurt i can recommend some awesome crutches |
| 05:27.25 | jpsharp | I understand that completely. I had a car accident a few years ago that resulted in a stroke and permanent disability. |
| 05:28.09 | slav3_kitten | oh man, i'm so sorry to hear that |
| 05:28.18 | slav3_kitten | you need crutches to get around? |
| 05:29.51 | jpsharp | No, I'm fully mobile. Just substantial vision loss from the stroke. |
| 05:30.28 | slav3_kitten | man i don't know how i'd live like that honestly |
| 05:30.37 | slav3_kitten | any chance it will heal over time? |
| 05:31.37 | jpsharp | Not any more. Regained a bit up till about 3 months after, but its the way it is for the rest of my life. |
| 05:31.54 | slav3_kitten | sorry to hear that :( |
| 05:32.13 | slav3_kitten | i just broke my spine a little on rigid conduit with the help of gravity |
| 05:32.57 | jpsharp | Really, its not that bad. Bad enough that I can't work, but not so bad that I'm a helpless invalid. |
| 05:33.13 | slav3_kitten | so i have nerve damage and a hyperactive nerve disorder which makes my right leg useless an gave me a total disability rating. had to hire help to run all my network cable because i can't get under houses anymore |
| 05:33.35 | slav3_kitten | but i manage to do what i can |
| 05:33.49 | slav3_kitten | pbx seems to promise to absorb a bunch of my time |
| 05:33.51 | slav3_kitten | which is good |
| 05:34.40 | jpsharp | yes. I still pick up side jobs here and there doing Asterisk. |
| 05:34.59 | slav3_kitten | you can actually get side jobs doing it? |
| 05:36.17 | jpsharp | They're hard to come by cause you're competing against "offshore" guys doing it for $20 a day versus $50/hour, but they're there. |
| 05:36.36 | slav3_kitten | interesting... |
| 05:37.04 | slav3_kitten | well i want to learn the ins an outs especially going from voip to pots using the voice fxo module in my 2621 |
| 05:37.17 | jpsharp | The last couple of jobs I got were "I hired this offshore guy, paid him a bunch of money, and I haven't got shit to show for it...HELP!" |
| 05:37.53 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 05:38.54 | jpsharp | Its all about configuing the dial-peers on the Cisco box. |
| 05:40.15 | slav3_kitten | yea i know nothing about that at this moment. figured i'd read the definitive guide to asterisk and then make a asterisk box (i hear there is an all inclusive distro for pbx using asterisk) and start fooling around |
| 05:43.21 | philipp64|laptop | God, I hate my ISP's Taqua switch... |
| 05:44.15 | *** join/#asterisk TriJetScud (~TriJetScu@testarossa.smb.curriegrad2004.ca) |
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| 05:49.57 | slav3_kitten | philipp64|laptop, i'll trade you |
| 05:50.05 | slav3_kitten | i'm on a quasi reliable wisp, and natted 3 times. if i want a public IP i have to pay them 15 usd a month |
| 05:50.27 | slav3_kitten | in addition to the 60 i pay for 3 down 1.5 meg up |
| 05:55.36 | slav3_kitten | i sort of have the feeling when they made this wisp it was all "so if we get 10 ip addresses, a rack of load balancers and nat the shit out of things.... |
| 06:05.07 | philipp64|laptop | sounds about right. |
| 06:05.45 | philipp64|laptop | well, I'm on G.PON with 3 static routable addresses, 20M/5M, for about $90/mo. |
| 06:06.19 | slav3_kitten | looks envious |
| 06:06.45 | slav3_kitten | funny thing, we bought this house with comcast telling us business class cable internet no problem |
| 06:07.14 | slav3_kitten | when we actually took possession comcast was like "well it's at the main road but we can't run a tap to your house, sorry!" |
| 06:07.25 | slav3_kitten | therein lies the wisp solution |
| 06:13.26 | *** part/#asterisk kaldemar (~kaldemar@unaffiliated/kaldemar) |
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| 06:41.13 | dandre | <PROTECTED> |
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| 07:08.33 | bulkorok | hi |
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| 07:56.59 | schmidts | good morning |
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| 08:17.03 | bulkorok | hi... where can I find the table schemes for queue and queuemember!? |
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| 08:21.52 | bulkorok | ok... found sth in {src}/contrib/realtime |
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| 08:31.54 | jacc0 | hi all! |
| 08:38.58 | jacc0 | almost weekend! |
| 08:39.08 | jacc0 | And I just took a new job!! |
| 08:39.37 | jacc0 | as: information security consultant |
| 08:42.29 | bulkorok | grat :-) |
| 08:42.39 | jacc0 | thanks |
| 08:43.05 | bulkorok | sounds like an "I hate this security guy" job ;-) |
| 08:46.00 | *** join/#asterisk mchou (~quassel@unaffiliated/mchou) |
| 08:56.15 | jacc0 | :P |
| 08:56.44 | jacc0 | I'm now maken security software; then I'll be breaking it |
| 08:56.55 | jacc0 | *making |
| 08:57.46 | jacc0 | and I won't be in this channel that much anymore |
| 08:58.11 | jacc0 | starting at @new job the first of septembre |
| 08:58.50 | jacc0 | *september |
| 08:58.53 | *** join/#asterisk freeedrich| (friedrich@perplexa.be) |
| 09:00.59 | jacc0 | my first assigment will be pentesting one of the bigger social network sites :) |
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| 09:28.39 | schmidts | has anyone have some knowledge about hylafax? |
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| 09:35.54 | din3sh | schmidts: installation problem? |
| 09:46.11 | *** join/#asterisk Scapal (~Scapal@2001:6f8:14f8:0:b0b7:c527:cc3f:9c37) |
| 09:49.58 | Scapal | hi. I'm trying to find an elegant way for shared meeting room phone. My way to do this would be to trigger a app giving you a code on your private phone and then to use a dialplan app on the meeting room phone for it to act as a surrogate, taking all the calls your private phone would normally receive and placing all the calls on the outgoing line your private phone would normaly use. |
| 09:52.21 | Scapal | I'm using Cisco spa5xxg phone. I use Asterisk app.SendText() to send the code on the private phone, my only question would be: how to hangup without having the phone giving a busy tone on the speaker. I tried different hangup cause codes without any luck. |
| 09:54.02 | Scapal | I also tried using Cisco XML Services tied to a line key but then I cannot use the BLF to light up the led in red when the feature is active. |
| 09:54.13 | Scapal | any idea? |
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| 10:15.09 | schmidts | din3sh no its running fine but i have a problem with hylafax running on a virtual (openvz) server cause i cant open the ftp port i need |
| 10:15.29 | schmidts | there is a kernel modul which should do this but i dont have access to this kernel modul from inside the virtual machine |
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| 11:19.43 | markfoodyburton | I have calls being cleared from a SIP server after the server's registration expiry period (30 min) (Apparently they are using the cirpack server, if that makes a difference). On the Asterisk (client) side all I see is the 'BYE'… Am I missing an obvious setting somewhere in sip.conf? |
| 11:23.14 | bulkorok | hi... how do I have to configure queuemembers with variable registrar? sth like SIP/1234@${regserv1} to queue test? |
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| 11:32.18 | teloniusz | hi. I have a Sangoma ISDN card. I've defined "context = " in the DAHDI users.conf trunk definitions I use for calling outside, but all the incoming connections are landing in a context defined in chan_dahdi.conf |
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| 11:32.54 | teloniusz | can I configure it to use users.conf contexts so that I can know through which channel the connection came? |
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| 11:40.26 | *** join/#asterisk sawgood (~sawgood@173-13-158-29-sfba.hfc.comcastbusiness.net) |
| 11:42.54 | sawgood | Hi: From the extensions.conf file on one Asterisk box (1.8.x) can I put a exten => entry which would be something like this: |
| 11:43.17 | sawgood | exten => 4155551212,1,Play(SIP/2ndAsteriskbox/filename.wav) |
| 11:43.30 | sawgood | basically, can I use the Play command IF the .wav file is on another box? |
| 11:43.45 | sawgood | there is a SIP trunk working between the two Asterisk boxes |
| 11:46.02 | *** join/#asterisk bobman (~bobman@2001:470:7:3b::2) |
| 11:46.13 | bulkorok | sawgood: I dont think so... |
| 11:46.51 | sawgood | No worries ... I'll just send the call to the right box then ... |
| 11:46.59 | bulkorok | but you can Dial the other box and let there play the wav |
| 11:47.00 | bulkorok | ^^ |
| 11:48.44 | sawgood | ty |
| 11:53.21 | *** join/#asterisk matteludde (3e5f11d2@gateway/web/freenode/ip.62.95.17.210) |
| 11:54.01 | matteludde | hello, any asterisk experts active? |
| 11:54.27 | bulkorok | ~ask |
| 11:54.27 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 11:55.12 | markfoodyburton | ok… how about: |
| 11:55.12 | markfoodyburton | I have calls being cleared from a SIP server after the server's registration expiry period (30 min) (Apparently they are using the cirpack server, if that makes a difference). On the Asterisk (client) side all I see is the 'BYE'… Am I missing an obvious setting somewhere in sip.conf? |
| 11:57.56 | *** join/#asterisk Bullmoose (~Bullmoose@71-33-0-219.bois.qwest.net) |
| 11:57.59 | matteludde | I am investigating the possibility to replace our old PBX with asterisk but I need a feature where users can set "away codes" (like lunch, vacation etc) and time/date when they will return. When someone calls them they should be informed of this by a pre-recorded voice |
| 12:00.02 | bulkorok | matteludde: in fact asterisk can handle this scenario. but it depends on your prerequisites how it has to do this |
| 12:02.17 | matteludde | bulkorok: can you please point me in the right direction on how to configure this? i have googled myself senseless but not found where to do it yet.. :/ |
| 12:05.09 | WIMPy | matteludde: The way you want it to work. There are lots of ways to do it. |
| 12:06.10 | bulkorok | so a user should type somehwere "out to lunch" and set his status to "away" and asterisk should not send the call to the users phnoe, but instead to VM? |
| 12:07.19 | bulkorok | this can be done with xmpp, text2speech and dialplan/database handling |
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| 12:14.14 | matteludde | well, our current PBX allows the user to select a few different hard coded away reasons and enter time/date when user returns. but i guess typing reason manually should also work |
| 12:14.41 | Beta2K | I'm having troubles with dundi and it seems to be a network issue on one of the pbx's. I see packets being transmitted by pbx A to pbx B on the line via tcpdump, I see pbx B receive them and ACK back on the asterisk console but that's it, tcpdump on pbx B doesn't show it transmitting back to pbx A |
| 12:14.51 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 12:15.00 | matteludde | i found some references to xmpp while googleing, but isn't that for combining asterisk with a im/chat software? |
| 12:15.48 | matteludde | do you know of any good guides or examples on how to use the xmpp in the way i want it? |
| 12:20.15 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 12:22.36 | slav3_kitten | off to car repair, lets how this does not end terribly |
| 12:23.51 | bulkorok | no... but xmpp is just a proposal... you can handle this with database and some php/html/javascript/perl/whateverest language and AMI too |
| 12:28.43 | matteludde | do you think it would be possible to purchase this setup as a service from an external company (and if so which one)? i do not have the time to sit down and learn how to code programs for asterisk |
| 12:29.50 | WIMPy | You might have luck in here, but I've been told, the asterisk-biz mailing list is the right place to ask. |
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| 12:32.15 | matteludde | i am running trixbox, perhaps fonality would be a good place to start? |
| 12:34.02 | *** join/#asterisk eduzimrs (~eduzimrs@201.22.86.124.static.gvt.net.br) |
| 12:34.19 | [TK]D-Fender | nope |
| 12:34.28 | [TK]D-Fender | Trixbox is DEAD, first off |
| 12:34.37 | [TK]D-Fender | and they run forked junk. |
| 12:34.56 | [TK]D-Fender | matteludde, I'd check the WIKI(s), BIZ lists, etc |
| 12:35.40 | matteludde | ok.. at least now i need where to start |
| 12:36.01 | matteludde | thanks, i'll investigate further on my own :) |
| 12:37.27 | matteludde | b.t.w. what version of asterisk do you recommend? I like trixbox since it's one prepackaged installation |
| 12:38.01 | [TK]D-Fender | Once you're taling "custom", nonoe of that matters |
| 12:38.09 | [TK]D-Fender | talking* |
| 12:38.10 | [TK]D-Fender | gah |
| 12:38.39 | [TK]D-Fender | If youwant an easier distro just to do the * part regardless of any GUI, I'd say AsteriskNOW |
| 12:39.07 | [TK]D-Fender | It also offers FreePBX as a base for the more boring SMB bits |
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| 12:41.09 | matteludde | ok.. thanks heaps everyone, i'll be back another day! :) |
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| 13:09.48 | sawgood | In Asterisk 1.8, did the file location for recorded files 'change' from /var/lib/asterisk/sounds/custom to something different? |
| 13:10.14 | WIMPy | Whatever you configure. |
| 13:10.17 | sawgood | not recording calls ... .wav or .gsm files ... for IVR messages or Playback |
| 13:10.23 | sawgood | oh ... cool |
| 13:10.45 | leifmadsen | "custom" was never something Asterisk used |
| 13:10.50 | leifmadsen | unless you created it and were using it |
| 13:11.59 | sawgood | would the directory for IVR files be defined in /etc/asterisk/asterisk.conf? |
| 13:12.26 | WIMPy | What are "IVR files"? |
| 13:12.41 | *** join/#asterisk [sr] (~kvirc@pal-213-228-181-48.netvisao.pt) |
| 13:12.43 | [sr] | howdy |
| 13:12.48 | [sr] | hi WIMPy |
| 13:13.03 | WIMPy | But, yes, your directories are defined there. |
| 13:13.06 | WIMPy | Hi [sr] |
| 13:13.13 | [sr] | whats up |
| 13:13.29 | sawgood | actually, scratch IVR files ... when using the Playback application, is the directory the application uses to look for files (is that inside of /etc/asterisk/asterisk.conf? |
| 13:13.43 | leifmadsen | asterisk.conf, yes |
| 13:13.54 | sawgood | Thank you, Leif! |
| 13:14.57 | sawgood | You would 'think' the directory would be /var/lib/asterisk/sounds |
| 13:15.03 | sawgood | not /var/lib/asterisk |
| 13:19.26 | *** join/#asterisk _stressed_ (2987afbf@gateway/web/freenode/ip.41.135.175.191) |
| 13:19.29 | leifmadsen | sawgood: well the astvarlib directory is /var/lib/asterisk, which has things other than "sounds" in it |
| 13:20.00 | leifmadsen | agi-bin, astdb, moh, keys, etc. |
| 13:20.42 | _stressed_ | Elo |
| 13:21.32 | sawgood | For ease of use, do you think Asterisk prefers files is .gsm or .wav format? |
| 13:21.57 | WIMPy | The format it has to send out. |
| 13:21.57 | sawgood | I'd like to come up with a known specific file type to save .wav files as (which would work on any box) |
| 13:22.22 | WIMPy | Or one that is easy to transcode, preferrably wav. |
| 13:22.51 | _stressed_ | Does anybody know if asterisk still support 2-way radio via USB as I've been doing a bit of research on it and all I found was articles pre 2010 and links that don't work anymore. |
| 13:28.28 | *** part/#asterisk festr (~festr@37.157.192.45) |
| 13:33.29 | *** join/#asterisk bitwize (~bitwize@h87-96-213-2.dynamic.se.alltele.net) |
| 13:36.33 | _stressed_ | as everything point so app_rpt but I don't find documentation |
| 13:36.55 | _stressed_ | so just wondering if its still around or am I looking in the wrong place or what |
| 13:37.30 | leifmadsen | _stressed_: that app_rpt stuff is basically being handled out of tree now |
| 13:38.08 | sawgood | strange, I created two .wav files (both saved as PCM signed, 16 bit, mono) (using GoldWave), one plays fine when using Playback, and the other does not |
| 13:38.27 | leifmadsen | _stressed_: although doing some googling, I have no idea where they keep that code now |
| 13:38.49 | leifmadsen | it's not in Asterisk proper though as they have not ported past Asterisk 1.4 and are no longer submitting changes upstream |
| 13:38.57 | sawgood | thing is ... the filenames have only one letter difference in them (sound-file and sound-file1) |
| 13:39.07 | _stressed_ | Yea it was odd for me as well |
| 13:39.15 | leifmadsen | sawgood: it should be 8-bit mono |
| 13:39.38 | sawgood | Is a .wav file saved as PCM 8 bit, mono (ok)? |
| 13:39.42 | _stressed_ | because I still found reference to it in the sample config files but thats about it |
| 13:40.37 | [TK]D-Fender | 16 bit mono, 8khz |
| 13:40.43 | leifmadsen | sawgood: yes, because that's what asterisk will try to play it at |
| 13:40.51 | *** join/#asterisk acidfoo (~nib@modemcable094.94-70-69.static.videotron.ca) |
| 13:40.54 | _stressed_ | but thanks for the help, I was scared I'm doing something stupid or looking in the wrong place or the name has changed |
| 13:40.58 | leifmadsen | _stressed_: ya it hasn't been removed yet (or maybe was removed recently) |
| 13:41.09 | leifmadsen | it would be removed in Asterisk 11 I think |
| 13:41.19 | _stressed_ | oh ok |
| 13:41.22 | _stressed_ | :( |
| 13:41.24 | leifmadsen | the most up to date version of it is not in Asterisk though (and it doesn't even compile I'm pretty sure) |
| 13:41.39 | leifmadsen | you need to find the people who are supporting it outside of the Asterisk tree |
| 13:42.08 | _stressed_ | ok I'll try and get hold of them |
| 13:44.25 | *** join/#asterisk gbit (~no@unaffiliated/gbit) |
| 13:46.08 | gbit | Hello, I'm tring to connect 2 asterisks on a vpn, but I got an error when I try to call. Sending fake auth rejection for device "asterisk" <sip:asterisk@192.168.0.5:5060> . But this IP is not the asterisk IP. Can I allow this IP and done? |
| 13:48.48 | [TK]D-Fender | make a trunk for it |
| 13:49.52 | gbit | I did on sip.conf |
| 13:50.57 | gbit | is that right? |
| 13:51.43 | *** join/#asterisk sp00kz (~ilubj00@unaffiliated/sp00kz) |
| 13:51.49 | [TK]D-Fender | yes |
| 13:52.01 | [TK]D-Fender | and that should have been "peer".... |
| 13:52.09 | [TK]D-Fender | missed what channel i was in for a moment there. |
| 13:52.30 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-ltranlkxhgozrsse) |
| 13:52.44 | gbit | I did it peer too, I read the definitife guide. |
| 13:53.11 | gbit | and it works when bouth servers was on the same subnet |
| 13:53.24 | gbit | now they are on different subnet and I got that error. |
| 13:53.58 | [TK]D-Fender | ~pb |
| 13:53.59 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 13:54.00 | [TK]D-Fender | ^^^ |
| 13:54.14 | gbit | moment please |
| 13:54.14 | [TK]D-Fender | Show us the call with SIP DEBUG enabled, along with your configs masking only the secret |
| 13:56.47 | *** join/#asterisk serafie (~erin@nat/digium/x-xxtmvfashztnbzhr) |
| 13:57.07 | *** join/#asterisk Gaiax (~Gaiax@unaffiliated/gaiax) |
| 13:59.09 | *** join/#asterisk nighty- (~nighty@static-68-179-124-161.ptr.terago.net) |
| 14:00.29 | gbit | [TK]D-Fender: Here it is: http://pastebin.com/sbnJ1JSS |
| 14:00.41 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 14:00.42 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 14:01.29 | *** join/#asterisk RubyRails (~justin@209.33.214.243) |
| 14:01.49 | *** join/#asterisk RubyRails (~justin@209.33.214.243) |
| 14:03.05 | [TK]D-Fender | gbit, I asked for the complete call SIP DEBUG from * CLI... |
| 14:04.17 | gbit | sorry, moment please |
| 14:05.59 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 14:06.35 | gbit | [TK]D-Fender: http://pastebin.com/0FxVre7P |
| 14:07.03 | [TK]D-Fender | "sip set debug on" <-------------------------------- |
| 14:07.14 | [TK]D-Fender | both sides |
| 14:07.22 | gbit | my bad again |
| 14:08.08 | *** join/#asterisk din3sh (din3sh@41.136.83.24) |
| 14:15.51 | *** join/#asterisk zerohalo (~zerohalo@74-94-141-204-ma.hfc.comcastbusiness.net) |
| 14:19.20 | gbit | [TK]D-Fender: I hope is ok now, http://pastebin.com/bw76UAT2 |
| 14:20.42 | [TK]D-Fender | <--- SIP read from UDP:192.168.0.5:5060 ---> <------------- |
| 14:20.58 | [TK]D-Fender | whatever you're using as a gatway appears to be rewriting the packets. |
| 14:21.06 | [TK]D-Fender | Stop that |
| 14:21.14 | gbit | is an iptables firewall |
| 14:21.20 | gbit | any tips? |
| 14:22.08 | [TK]D-Fender | get rid of any SIP PROXY, ALG, or any other module like it in the way |
| 14:22.40 | gbit | ok I will do that, thanks |
| 14:23.01 | nickfennell | nf_nat_sip |
| 14:23.06 | nickfennell | of nf_conntrack_sip |
| 14:23.09 | nickfennell | nat_sip |
| 14:23.18 | gbit | yes I have nf_nat_sip enabled |
| 14:23.23 | gbit | turning off now and testing |
| 14:24.49 | nickfennell | that will prevent contrack from inspecting the sip headers and rewriting packets |
| 14:24.54 | nickfennell | quite useful sometimes but not in others |
| 14:25.56 | *** join/#asterisk edge (~edge@97-64-216-2.client.mchsi.com) |
| 14:26.08 | gbit | funny is that I enabled that helping me out |
| 14:26.37 | *** join/#asterisk Gaiax (~Gaiax@unaffiliated/gaiax) |
| 14:27.04 | edge | Should i be using some kind of buffer between my DADHI chan and my voicemail? I find that some of the audio playback is gapped and poor |
| 14:27.26 | gbit | only conntrack need to be disabled too? |
| 14:28.01 | [TK]D-Fender | Everything related to SIP |
| 14:28.07 | gbit | got it |
| 14:28.10 | [TK]D-Fender | Do not mess with any of it |
| 14:28.29 | gbit | I will test it now, thanks in advance |
| 14:31.53 | *** join/#asterisk mbrit (~mbrit@186.120.97.195) |
| 14:34.03 | edge | When i make changes to chan_dahdi, all i should have to do to make those changes take effect is 'dahdi restart' ? |
| 14:35.17 | *** join/#asterisk GeoGeek (~steve-o_@12.71.122.227) |
| 14:36.33 | gbit | Now the source IP is OK, but still the same error: http://pastebin.com/8Z4BRiM9 |
| 14:41.06 | [TK]D-Fender | <--- SIP read from UDP:192.168.2.1:5060 ---> |
| 14:41.11 | [TK]D-Fender | 2.1 is not 2.247 |
| 14:41.19 | *** join/#asterisk learath (47f6db23@gateway/web/freenode/ip.71.246.219.35) |
| 14:41.27 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 14:41.28 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 14:41.43 | gbit | this 2.1 is a simple firewall if no sip modules, only nat masqueraded |
| 14:42.20 | [TK]D-Fender | Your peer that is receiving that call is expecting ANOTHER IP |
| 14:42.25 | [TK]D-Fender | You are screwing with it |
| 14:43.07 | [TK]D-Fender | host=192.168.2.247 <--- if it doesn't come from here = GTFO |
| 14:44.29 | [TK]D-Fender | NAT is another can of worms, especially if you have it withing 2 LAN segments and ever have a need to connect to resources on the internet |
| 14:46.01 | gbit | OK, I'm just trying to figure out how to make the same on both sides. |
| 14:50.49 | [TK]D-Fender | Stop NATing |
| 14:51.16 | gbit | How? I can't get real IP on ethernet. |
| 14:53.01 | [TK]D-Fender | ""real IP"? Pardon? |
| 14:53.17 | [TK]D-Fender | You're VPNing. there is no reason you should be NATing one subnet to another like that |
| 14:53.38 | gbit | you mean only use port forwarding? |
| 14:59.15 | gbit | When I enable allowguest=yes is working :/ |
| 15:00.33 | [TK]D-Fender | You should not HAVE to forward. Packets should jsut flow sourced as noraml. |
| 15:01.07 | gbit | [TK]D-Fender: Ok I will try that |
| 15:01.12 | gbit | Thanks alot |
| 15:04.13 | gbit | [TK]D-Fender, When I disable ip_forward on the linux gateway, my connection to the other network stops. |
| 15:04.41 | [TK]D-Fender | facepalms |
| 15:04.58 | [TK]D-Fender | <gbit> this 2.1 is a simple firewall if no sip modules, only nat masqueraded <-- NO MASQUERADE |
| 15:05.10 | [TK]D-Fender | only FORWARD |
| 15:05.36 | gbit | yes, I turnded of nat. But it only works when I change to allowguests=yes on sip.conf |
| 15:06.10 | gbit | off* |
| 15:07.09 | [TK]D-Fender | I'm not seeing your call.. |
| 15:08.01 | gbit | sorry.. |
| 15:08.49 | *** join/#asterisk hariom (~hariom@117.195.188.190) |
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| 15:29.14 | rgsteele | So, I've been trying to track down why my disposition is always "NOT ANSWERED" for inbound calls, even when I'm answering them myself. This morning, I set 'unanswered = no' into cdr.conf, and it started filling in some of the missing CDR variables. It seems to me that it's trying to detect whether it's answered too soon. Is there any way to delay that check? Is ForkCDR relevant here? |
| 15:29.28 | rgsteele | (Again, somehow the disposition logged to Master.csv and postgres via cdr_pgsql reports "ANSWERED" as the disposition, it's only 'core show channel <chan>' that reports "NO ANSWER" via the Asterisk CLI) |
| 15:39.09 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
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| 16:09.24 | Kalidarn | does anyone know how i can get asterisk to compile on freebsd with symbols |
| 16:09.34 | Qwell | with symbols? |
| 16:09.40 | Kalidarn | for gdb |
| 16:09.45 | Kalidarn | im getting a segfault |
| 16:09.46 | Qwell | enable DONT_OPTIMIZE in menuselect |
| 16:09.51 | Kalidarn | oh |
| 16:12.07 | Kalidarn | i didn't see that in make config Qwell |
| 16:12.29 | Qwell | it's there |
| 16:12.35 | *** join/#asterisk Ice_Strike (~Ice_Black@host213-120-117-212.in-addr.btopenworld.com) |
| 16:12.38 | Kalidarn | i gried doing WITH_DEBUG=yes STRIP= install |
| 16:12.38 | Ice_Strike | Hello |
| 16:12.52 | Qwell | why? |
| 16:15.59 | Kalidarn | Qwell: these are the only options i see http://pastebin.com/pKEUGKGQ |
| 16:16.09 | Kalidarn | they seem all driver relate |
| 16:16.13 | Qwell | wtf is that? |
| 16:16.21 | Qwell | I said make menuselect |
| 16:16.46 | Kalidarn | oh i thought you said make config |
| 16:17.19 | carrar | heh |
| 16:17.31 | Kalidarn | hmm i wonder if i can access it through freebsd's port though |
| 16:17.35 | carrar | Hello Ice_Strike |
| 16:17.39 | Qwell | doubt it. Don't use ports. |
| 16:18.05 | carrar | Never use port 5060 |
| 16:18.07 | Kalidarn | yeah but then isn't it going to get all the paths wrong and install stuff all over the place |
| 16:18.32 | Kalidarn | or does asterisk have provisions for freebsd |
| 16:18.43 | Qwell | ./configure --help |
| 16:18.53 | carrar | Just have to do it different;y? |
| 16:24.18 | Kalidarn | oh Qwell it looks like it configured it for freebsd 9, the make file obviously has knowledge of that |
| 16:24.33 | Kalidarn | afterall i guess asterisk does officially support freebsd |
| 16:24.39 | Ice_Strike | I have thousands wav files in the asterisk monitor folder.. how to convert them all to MP3 into mp3 folder |
| 16:24.44 | Ice_Strike | and then delete all the wav files |
| 16:24.54 | Qwell | Ice_Strike: Don't do that |
| 16:25.11 | Ice_Strike | @Qwell Oh? |
| 16:25.24 | Qwell | mp3 is evil |
| 16:25.24 | carrar | heh |
| 16:25.33 | *** join/#asterisk rneese (~Richard@70.127.108.139) |
| 16:25.38 | Ice_Strike | Qwell When what is a solution? |
| 16:25.44 | carrar | What is the problem? |
| 16:25.49 | rneese | how did asterisk jump from 1.8 to 10.x.x |
| 16:25.54 | Qwell | ~asterisk 10 |
| 16:25.57 | Qwell | ~asterisk10 |
| 16:25.58 | infobot | Asterisk 10 -- http://blogs.digium.com/2011/07/21/the-evolution-of-asterisk-or-how-we-arrived-at-asterisk-10/, or a Standard Release. It was released on 2011-12-15, with maintenance until 2012-12-15. Asterisk 10 will be end of life on 2013-12-15. |
| 16:25.58 | mjordan | Kalidarn: I would not say that Asterisk "officially" supports FreeBSD. We do not run our continuous integration tests on FreeBSD, and at this time, we have no plans to do so |
| 16:26.00 | rneese | why did yo unot go 2.0 |
| 16:26.14 | mjordan | rneese: read the article. |
| 16:26.29 | Kalidarn | mjordan: i guess, not commercially supported but still supported. |
| 16:26.31 | Ice_Strike | @Qwell Each file is about 30 seconds long - so quality does not matter reallly asong voice recording is understanable |
| 16:26.40 | Kalidarn | probably the only commercially supported things are rhel |
| 16:26.42 | rneese | there are asterisk ports on bsd |
| 16:26.50 | rneese | and they are tested and maintained |
| 16:26.59 | [TK]D-Fender | Ice_Strike, what is the point of converting them? |
| 16:26.59 | Kalidarn | yeah except i can't' get it to compile with symbols for gdb |
| 16:27.12 | Kalidarn | thats the only reason i took Qwell's advice and manually compiled it |
| 16:27.27 | Ice_Strike | [TK]D-Fender For the monitoring purposes.. |
| 16:28.02 | [TK]D-Fender | Ice_Strike, "man lame" |
| 16:28.10 | Qwell | [TK]D-Fender: lame, man |
| 16:28.15 | rneese | The Server could not get you the page you are looking for. The content might have been moved or deleted. |
| 16:28.16 | rneese | Don't lose your hope just yet. You might try searching through the Archives. |
| 16:28.23 | rneese | the url is broken |
| 16:28.32 | Qwell | rneese: No, it most definitely is not. |
| 16:28.42 | rneese | i clicked on it |
| 16:28.48 | Qwell | Get a better IRC client then. |
| 16:29.19 | Ice_Strike | so use lame? |
| 16:29.26 | rneese | no rm the , after the rnd / |
| 16:29.35 | carrar | Ice_Strike, what are you trying to do? |
| 16:29.35 | rneese | rnd/end |
| 16:29.47 | rneese | or move it over 1 space |
| 16:29.48 | carrar | Why convert them to mp3? |
| 16:30.10 | Ice_Strike | carrar because each file is about 500KB for 30 seconds |
| 16:30.15 | carrar | compress them |
| 16:30.24 | Ice_Strike | carrar how? |
| 16:30.28 | Ice_Strike | compress wav? |
| 16:30.35 | carrar | bzip2 |
| 16:30.38 | Ice_Strike | Oh that |
| 16:30.53 | carrar | Hard drive space is cheap |
| 16:31.31 | Ice_Strike | It is not about hard drive space.... its that agents from abroad will be listening the calls and they may have slow internet connnection |
| 16:32.08 | [TK]D-Fender | Ice_Strike, listening how? |
| 16:32.26 | Ice_Strike | [TK]D-Fender I have made a website they can listen the call via web. |
| 16:32.43 | coppice | Ice_Strike: if you want small files for distribution why use MP3? any of several VoIP codecs would be better |
| 16:33.59 | Ice_Strike | [TK]D-Fender They listen the call on the website... on the browswer it will show player plugin |
| 16:34.10 | Ice_Strike | coppice I am using ulaw. |
| 16:34.13 | [TK]D-Fender | Ice_Strike, Well you've got your answer. lame is it. |
| 16:34.30 | *** join/#asterisk saliak (~kailas@ip68-9-228-184.ri.ri.cox.net) |
| 16:34.59 | coppice | Ice_Strike: ulaw is only 240k for 30s |
| 16:36.01 | saliak | I'm trying to test outgoing faxes from the console. However, when it dials out in my dial plan, the Dial command turns into: Dial("Console/dsp", "SIP/ … which seems to generate the following warning chan_oss.c:485 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory. why dos that happen (the /dsp thing), and is there a way to suppress it? |
| 16:36.27 | [TK]D-Fender | saliak, how is CONSOLE supposed to send a fax? |
| 16:36.40 | *** part/#asterisk rneese (~Richard@70.127.108.139) |
| 16:37.06 | pabelanger | Fax over Speakers |
| 16:37.09 | Ice_Strike | coppice Well every files is 32 seconds - between 500-530KB |
| 16:37.11 | saliak | [TK]D-Fender: well, it doesn't really. i'm using the SendFax function with a previously generated tiff. this is mostly to debug. |
| 16:37.23 | saliak | pabelanger: i'm really good at whistling |
| 16:37.31 | coppice | Ice_Strike: then they are not ulaw |
| 16:38.49 | coppice | unless they are stereo files, keeping the two direction distinct |
| 16:42.11 | Ice_Strike | coppice I am using Mixmonitor() |
| 16:42.25 | Ice_Strike | coppice They are deffo ulaw i am sure it is |
| 16:43.15 | saliak | so how does it work exactly? my plan was to have a shell script get the pdf from an incoming email, convert it to a tiff , and issue asterisk cli commands to do a console dial of the outgoing fax extension (that dials the remote number and calls SendFax. what i'm doing now (just issuing the dial command at the CLI) doesn't seem to be working, and i'm not sure if its because of the /dev/dsp error). my fax connects and sits there for a while |
| 16:43.16 | saliak | but nothing comes through |
| 16:47.47 | *** join/#asterisk ChkDigit (~u388mw@74.3.144.66) |
| 16:50.27 | [TK]D-Fender | Ice_Strike, and what format are you saving them in? Choose another then |
| 16:52.13 | Ice_Strike | [TK]D-Fender same => n,MixMonitor(${calr_id}.wav,a) |
| 16:52.28 | Ice_Strike | I only need to record audio going in |
| 16:52.31 | Ice_Strike | not OUT |
| 16:53.56 | carrar | save them as gsm files! |
| 16:54.59 | Ice_Strike | hmm gsm |
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| 16:56.30 | Ice_Strike | Hmm would Windows Media Player on the browser support gsm |
| 16:56.43 | Ice_Strike | it will be need to be compatible |
| 16:57.43 | [TK]D-Fender | Ice_Strike, so don't use wav and don't use WMP |
| 16:58.33 | WIMPy | Can you use wav49 with mixmonitor? |
| 17:03.26 | Ice_Strike | WIMPy i will check :) |
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| 17:27.20 | Kalidarn | Qwell: can use freebsd ports, what i had to do was go /usr/ports/net/asterisk16 |
| 17:27.23 | Kalidarn | then do make extract |
| 17:27.37 | Kalidarn | then in /usr/ports/net/asterisk16/work/asterisk-source-blah |
| 17:27.45 | Kalidarn | i could do ./configure and then make menuselect |
| 17:28.16 | Kalidarn | and then the make install in /usr/ports/net/asterisk16 |
| 17:28.59 | Kalidarn | that way i could use bsd ports but enable the DEBUG_THREADS and DONT_OPTIMIZE |
| 17:29.23 | blizzow | Do different audio codecs have an effect on duplexing? When I make a call and speak, I can't hear what's going on on the other side. If I get any background noise while someone is talking to me, it seems like their audio is cutting out. |
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| 17:30.45 | blizzow | I currently have ulaw, alaw, and gsm enabled on the asterisk side, but am interested to know what the best choice might be for wireless voip using a softphone. |
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| 17:54.05 | rgsteele | Hm... with a call, there are two CDR records - one for the SIP channel associated with the VoIP provider, and a 2nd for the agent that answered the call. For calls where the disposition is correctly set to ANSWERED on *both* channels (after the agent picks up), the Application is set to "Bridged Call", but on the calls where disposition is incorrectly left as "NOT ANSWERED", the Application... |
| 17:54.08 | rgsteele | ...is set to AppQueue. |
| 17:54.38 | *** join/#asterisk dorphalsig (be55157a@gateway/web/freenode/ip.190.85.21.122) |
| 17:54.42 | *** part/#asterisk mpe (~mpe@109.70.54.56) |
| 17:55.28 | rgsteele | However, the call is definitely answered (I'm answering it). I've seen this with three different 1.6 versions (including the latest), and the same config worked with 1.4.17, so I don't think it's a config issue (I don't really muck w/ the CDR other than to set the userfield). |
| 17:56.56 | Qwell | rgsteele: What happened with 1.8/10? |
| 17:57.43 | dorphalsig | Hello, I am trying to setup a fax to email gateway on my * box, but I have the strangest error of them all. When a local fax (local line, outside of *) sends a multi page fax, it goes through with not even a hiccup. However if an international fax arrives, the fax fails to recieve. |
| 17:58.45 | rgsteele | Qwell: I had 1.8 set up, but there's an internal app that doesn't work right w/ it yet (code needs refactoring to work w/ the latest version of Asterisk and AMI). I supposed I could install it again for giggles just to see if anything changes. |
| 17:58.57 | Qwell | dorphalsig: That's one of the least strange fax issues I've seen. |
| 17:58.58 | rgsteele | It may just be a weird DAHDI thing too - the old system used Zaptel. |
| 17:59.07 | Qwell | International calls have terrible audio, therefore, faxes will fail frequently. |
| 17:59.21 | Qwell | rgsteele: well.. |
| 17:59.27 | Qwell | ~upgrade asterisk |
| 17:59.28 | Qwell | ~upgradeasterisk |
| 17:59.29 | infobot | Before requesting assistance, you should be running the latest version of a supported release branch. See the channel topic for the latest versions available in currently supported branches. |
| 17:59.42 | rgsteele | 1.10 isn't an option, but I can do 1.8 |
| 17:59.48 | Qwell | 10* |
| 18:00.11 | rgsteele | Er, yeah, 10. |
| 18:00.28 | Kalidarn | oh and nevermind Qwell that didn't actually work, doing that causes it to segfault. :( |
| 18:01.07 | dorphalsig | Qwell: What could be wrong? |
| 18:01.19 | Qwell | dorphalsig: I just said what was wrong |
| 18:01.39 | dorphalsig | ah, yeah, but constantly? |
| 18:02.18 | Qwell | if the audio is consistently bad, then yes, faxes will consistently fail |
| 18:02.34 | dorphalsig | What can be done then? |
| 18:02.57 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 18:02.57 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 18:03.13 | Qwell | get a local box in that country, or an ITSP there that does T.38 |
| 18:03.39 | *** join/#asterisk Korolev (~Korolev@204.88.28.115) |
| 18:04.02 | dorphalsig | Qwell: Strange thing is, when I hook up a normal fax machine it works. |
| 18:04.19 | Qwell | what are you using for fax? |
| 18:04.48 | dorphalsig | Recvfax command |
| 18:06.17 | Qwell | dorphalsig: spandsp? |
| 18:06.27 | Qwell | maybe coppice can chime in |
| 18:07.26 | dorphalsig | yeah |
| 18:09.14 | dorphalsig | Qwell: Its been some time since a general optimization is done to my * system, how much would you charge to come in, take a look do a diagnosis and tell us whats wrong and how much would it cost to fix? |
| 18:09.34 | Qwell | dorphalsig: I don't do consulting. |
| 18:10.07 | dorphalsig | Qwell: Do you know anybody who does and is damn good at it? |
| 18:10.24 | *** join/#asterisk serafie (~erin@75.76.38.159) |
| 18:12.12 | rgsteele | Qwell: No change w/ 1.8. Sure a bug (if it is indeed a bug) this glaring wouldn't survive through two major releases? |
| 18:12.49 | rgsteele | Surely, even. |
| 18:14.08 | rgsteele | The behavior is exactly the same - Application's value is "AppQueue" instead of "Bridged Call", Data's value is "(Outgoing Line)", disposition's value is "NO ANSWER" |
| 18:14.55 | rgsteele | Despite the fact that both "Connected Line ID" and "Connected Line ID Name", and other indicators of an active call, are populated. |
| 18:15.45 | rgsteele | If it's not a bug, there's been some kind of config change that changes the CDR's disposition when the inbound caller dials an extension at a prompt. |
| 18:16.41 | rgsteele | The only diff is that in the old system, Extension (i.e., the dialed extension after the inbound caller is connected) is not populated with a value, whereas in the new one, it lists the extension dialed by the caller. |
| 18:17.09 | rgsteele | I'm guessing it's got something to do with not being able to track the call thru the transfer |
| 18:17.52 | rgsteele | But the "why" escapes me - I'm not even sure where to look that I already haven't. |
| 18:18.52 | rgsteele | I'm wondering if a well placed ResetCDR or ForkCDR could be a workaround, though it feels hacky |
| 18:20.06 | rgsteele | In fact, I'd be happy to know if there's a better way to determine how disposition is set as the call makes its way through the dialplan. |
| 18:39.16 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v000-095.mobile.uci.edu) |
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| 18:44.37 | *** join/#asterisk drhonk (~drhonk@static.57.112.4.46.clients.your-server.de) |
| 18:44.48 | drhonk | good afternoon |
| 18:45.28 | drhonk | anyone here running using SIP client in TCP mode? |
| 18:46.25 | drhonk | I'm having issue with smart phone SIP client unable to go to background mode when using TCP connection |
| 18:47.35 | Qwell | sounds like a broken app |
| 18:47.37 | drhonk | or disconnect the session when losing connection ... asterisk some how still see the session as active |
| 18:47.47 | drhonk | but works with UDP .. just not TCP |
| 18:48.01 | drhonk | are session-timers expected to work with TCP? |
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| 19:01.12 | anonymouz666 | session timer is nice but...some clients does not support it, then for me... it's useless :( |
| 19:02.31 | drhonk | hmm.. |
| 19:02.44 | *** join/#asterisk Jacke (jacke@valhalla.bofh.org) |
| 19:02.49 | drhonk | yeah for some reason asterisk doesn't even send reinvite if sip client connects using TCP |
| 19:05.07 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 19:18.54 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 19:32.57 | Ice_Strike | Is there a command to check codec is channel currently using |
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| 19:35.16 | [TK]D-Fender | sip show channel [x] |
| 19:35.53 | carrar | No such SIP Call ID starting with '[x]' |
| 19:36.07 | anonymouz666 | Ice_Strike: sip show channels |
| 19:36.21 | Ice_Strike | yea ulaw |
| 19:36.46 | anonymouz666 | sip show channelstats is also a cool |
| 19:37.05 | anonymouz666 | command |
| 19:37.22 | Ice_Strike | Thats cool! |
| 19:37.35 | Ice_Strike | core show channels is also cool |
| 19:38.05 | anonymouz666 | it is not. |
| 19:38.21 | Ice_Strike | How do you guys listen recorded audio on the web? |
| 19:38.47 | Ice_Strike | What file format do you us |
| 19:38.49 | Ice_Strike | use |
| 19:41.39 | Ice_Strike | 23 active channels :) |
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| 19:51.04 | vader-- | have any of you guys used talkswitch phones with an asterisk system? |
| 19:53.27 | sawgood | vader--: I definitely have |
| 19:53.50 | sawgood | its no fun ... let them go and use Aastra phones |
| 19:58.34 | vader-- | im trying to figure out if i can get their old TS-200 analog phones to work |
| 20:02.00 | sawgood | oh ... in that case ... definitely with an ATA device you are set |
| 20:02.11 | sawgood | I remember that phone |
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| 20:09.25 | vader-- | it's a digital phone |
| 20:09.32 | vader-- | not sure if it will work :-/ |
| 20:12.19 | [TK]D-Fender | vader--, Dead end. Proprietary digital phones that weren't worth anyone making a convertor for. |
| 20:12.26 | [TK]D-Fender | vader--, Sell them off and be rid of them |
| 20:15.35 | l0st-soul | sawgood: what do you think about snoms? we have like 80 of them and i'm quite happy with them |
| 20:15.55 | l0st-soul | i'm interrested why aastra are better ? |
| 20:16.00 | l0st-soul | if they are |
| 20:18.12 | l0st-soul | (we've got snom series 300/320/360/370) |
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| 20:48.06 | *** join/#asterisk ruied (~ruied@po-217-129-155-146.netvisao.pt) |
| 20:55.34 | ruied | hello, I need to schedule the following at every 17:00pm "exten => D_N,1,Set(GLOBAL(Day_Night)=${Night_Mode})" but no idea the way to do it... something that could be done from cron, but do not know if it is possible.... |
| 20:56.24 | [TK]D-Fender | ruied: Call-files, CLI/AMI Originate. Take your pick |
| 21:01.02 | ruied | [TK]D-Fender, thanks, going to check that... |
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| 21:07.48 | leifmadsen | ruied: I'd just do something like, "asterisk -rx 'dialplan set global Day_Night 1" |
| 21:08.42 | [TK]D-Fender | ruied: You should use AstDB as that has better persistence and could also set that off -rx as well |
| 21:08.58 | pabelanger | why do just use gotoiftime |
| 21:09.10 | leifmadsen | pabelanger: that's what I've done as well :) |
| 21:09.30 | leifmadsen | pabelanger: the nice thing with not using that method is that you can set the day_night mode whenever you want |
| 21:09.47 | leifmadsen | although I usually combine them, and use GotoIfTime() and a "check day mode" or something |
| 21:09.54 | leifmadsen | so you can set day/night mode early if you want |
| 21:10.05 | leifmadsen | like on a vacation day or something |
| 21:11.28 | leifmadsen | enables weekend mode |
| 21:13.00 | ruied | nice ideas :) |
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| 21:24.39 | slav3_kitten | <PROTECTED> |
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| 21:45.57 | Qwell | slav3_kitten: as long as your client isn't stupid, sure |
| 21:46.18 | slav3_kitten | it's x-chat thus tends to be semi-smart |
| 21:46.39 | slav3_kitten | when my friend finishes rebuilding his server raid array i'll idle my remote client irssi if it's cool |
| 21:47.02 | Qwell | I can assure you, the 182 people in here aren't all watching their screens. :p |
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| 21:47.30 | slav3_kitten | i'd run irssi in screen on my server in the rack... if i bothered to pay the 15/m the wisp wants to get me a public IP |
| 21:48.14 | slav3_kitten | which i'll need to do if i want to get a UK DID so i can talk with my friends over there regularly |
| 21:48.32 | Qwell | people actually use phones to talk to friends still? |
| 21:49.01 | slav3_kitten | busy people leading busy lives do |
| 21:50.27 | slav3_kitten | thou everyone in this house is convinced i've lost my mind with my current drive to learn ip-pbx using sip phones to pstn an what not |
| 21:50.52 | slav3_kitten | likely because i'm going to run 6 phones, 1 per room and 1 garage phone |
| 21:51.08 | Qwell | I think we've all had the losing our mind conversation at home. |
| 21:51.52 | Qwell | "So wait, why do you need 20 different models of phones again?" |
| 21:52.12 | slav3_kitten | lol |
| 21:52.41 | *** part/#asterisk GeoGeek (~steve-o_@12.71.122.227) |
| 21:54.51 | slav3_kitten | also i wanted to cry when i was watching person of interest the other night. it's the episode where they introduce the "root" character, and she all sent him an IRC PM... and her IP started 455.x.x.x |
| 21:55.02 | slav3_kitten | all i could think was "screw this, i'm out!" |
| 21:57.48 | ruied | can't set AstDB from dialplan cmd... changed global variable to AstDB, it's working, but can't change it from asterisk cli... |
| 21:59.02 | slav3_kitten | off to the hackerspace i go |
| 21:59.07 | slav3_kitten | peace out |
| 21:59.28 | ruied | from cli I can see the 'dialplan set global' but no 'dialplan set db' |
| 21:59.40 | ruied | am I missing something? |
| 22:01.45 | l0st-soul | ruied: if you want to change a db value from cli: database put |
| 22:04.56 | ruied | l0st-soul, thanks :) its working... |
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| 22:11.54 | ruied | l0st-soul, do you know how to set 'Set(DEVICE_STATE(Custom:lamp1)=INUSE)' from cli? |
| 22:16.38 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 22:21.36 | ruied | found it! devstate change Custom:lamp1 NOT_INUSE |
| 22:23.21 | l0st-soul | great |
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