| 00:06.02 | slav3_kitten | pounces Katty |
| 00:06.03 | *** join/#asterisk italorossi (~italoross@187.61.168.117) |
| 00:06.06 | slav3_kitten | huzzah |
| 00:06.19 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 00:06.19 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 00:06.28 | Katty | slav3_kitten: ohai ^_^ |
| 00:06.31 | Katty | slav3_kitten: how'rechu |
| 00:06.43 | Katty | slav3_kitten: also, my condolences about your father :< |
| 00:07.08 | slav3_kitten | it happens |
| 00:07.16 | slav3_kitten | i'm stuffed leopard |
| 00:07.29 | slav3_kitten | got ethopian |
| 00:11.06 | *** join/#asterisk JuStIcIa_ (~artur0@190.167.126.135) |
| 00:16.11 | Katty | slav3_kitten: how's life n such? |
| 00:16.20 | Katty | slav3_kitten: sorry about slow responses, lifting weights |
| 00:16.30 | *** join/#asterisk sputnik_too (~sputnik@funtoo/user/sputnik-too) |
| 00:16.39 | Free99 | you guys: a2billing is pissing me the hell off. Do you know any open-source alternatives? |
| 00:16.59 | slav3_kitten | it's cool, life sucks but i got some good food |
| 00:17.04 | carrar | PostgreSQL is opensource |
| 00:17.09 | carrar | write your own!! |
| 00:17.48 | Katty | slav3_kitten: good food is a good start! |
| 00:17.50 | Free99 | carrar: I could I guess, but I need this done by friday or my boss is going to buy some crap closed-source windows based telephony thing, and make me support it |
| 00:18.05 | Free99 | I like linux a helluva lot better |
| 00:18.22 | carrar | better get crackin |
| 00:18.24 | slav3_kitten | Katty, etheopian is <3 |
| 00:18.33 | Katty | slav3_kitten: not my thing, but hey...more for you! |
| 00:18.41 | Katty | slav3_kitten: back in a sec. got a set of squats to do |
| 00:19.15 | slav3_kitten | you ever had it? |
| 00:19.41 | Katty | yep. and african. and russian too |
| 00:21.00 | Katty | african wasn't my thing either, but russian is tasty! |
| 00:23.41 | Free99 | you guys: there is no other billing system I could use on top of asterisk? |
| 00:23.55 | Free99 | this is for practically a non-profit, so.. |
| 00:25.06 | Katty | so...? |
| 00:25.14 | Katty | please finish your statement (= |
| 00:25.58 | Free99 | iunno. I can't pay for it. |
| 00:26.19 | navaismo | http://www.voip-info.org/wiki/view/Asterisk+billing |
| 00:27.20 | *** join/#asterisk classix (~salven@silenceisdefeat.com) |
| 00:28.37 | Katty | navaismo: that was nife of you |
| 00:28.39 | Katty | nice |
| 00:28.43 | Katty | infobot: billing |
| 00:28.56 | Katty | infobot: billing is http://www.voip-info.org/wiki/view/Asterisk+billing |
| 00:28.56 | infobot | okay, Katty |
| 00:29.01 | Katty | infobot: <3 |
| 00:29.01 | infobot | rumour has it, <3 is love |
| 00:29.06 | Free99 | navaismo: thanks very much man, I owe you a beer b/c of how much a2b is stressing me the f**k out |
| 00:29.12 | Katty | infobot: good bot. |
| 00:29.12 | infobot | Katty: :) |
| 00:29.20 | Katty | infobot: botsnack |
| 00:29.20 | infobot | :), Katty |
| 00:30.19 | Katty | carrar: how's the wifie anyway? |
| 00:35.02 | *** join/#asterisk bchia (~Adium@75.76.51.204) |
| 00:42.38 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 00:42.38 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 00:55.15 | *** join/#asterisk krzdrn (~sb@static-71-185-213-22.phlapa.fios.verizon.net) |
| 00:58.18 | Free99 | peace guys |
| 01:02.26 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
| 01:18.35 | *** join/#asterisk mjordan (~mjordan@75.76.55.191) |
| 01:18.35 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 01:27.48 | *** join/#asterisk work (work@82.137.193.222) |
| 01:28.00 | *** join/#asterisk Spengler1 (~Spengler1@pool-98-117-215-165.bltmmd.fios.verizon.net) |
| 01:28.36 | Spengler1 | hey [TK]D-Fender : I finally got my ACD queue BLF to work |
| 01:29.06 | Spengler1 | i have a digium phone so I had to get DPMA working among other things which is why it took so long |
| 01:29.11 | Spengler1 | but it works nicely now |
| 01:29.40 | *** join/#asterisk sputnik_too (~sputnik@funtoo/user/sputnik-too) |
| 01:31.35 | work | hello - please in case PRI-CPE is signaling contain both call party called and calling ? or one part and switch number |
| 01:35.20 | *** join/#asterisk droemel (~droemel@p4FCAD9D1.dip.t-dialin.net) |
| 01:38.31 | *** join/#asterisk Sorcier_FXK (~nssystem@unaffiliated/sorcierfxk) |
| 01:40.12 | *** join/#asterisk Bradada (~Bradada@114-32-7-232.HINET-IP.hinet.net) |
| 01:40.39 | work | any tool to view all pri signalling with dahdi ? |
| 01:40.57 | work | in debug mode |
| 01:42.25 | *** join/#asterisk blizzow (~jburns@75-171-154-172.hlrn.qwest.net) |
| 01:42.44 | *** join/#asterisk TSM (~the_softw@fw-lon1.wenn.com) |
| 01:42.58 | sruffell | work: pri intense debug span <x> doesn't give you what you need? |
| 01:43.56 | work | how to do it ? |
| 01:44.57 | sruffell | from the asterisk console enter that command |
| 01:45.02 | sruffell | asterisk CLI rather |
| 01:45.32 | work | what the output should be ? |
| 01:48.24 | *** join/#asterisk Bradada (~Bradada@114-32-7-232.hinet-ip.hinet.net) |
| 01:48.34 | sruffell | it'll take me a few minute to grab some output for you. BUt if you have a system setup…you should be ok to turn it on and look. |
| 01:49.13 | *** join/#asterisk andresmujica (~andresmuj@ubuntu/member/andresmujica) |
| 01:49.59 | work | actually at current time E1 not avalube I move server to mentenance in my office |
| 01:52.37 | work | and compiling new asterisk it need afew minutes |
| 01:58.23 | *** join/#asterisk Bradada (~Bradada@114-32-7-232.hinet-ip.hinet.net) |
| 02:03.13 | *** join/#asterisk zemmali-voip (~zemmali@unaffiliated/zemmali-voip) |
| 02:07.02 | work | Sruffell sorry what command exaxtry should excute in asterisk console ? |
| 02:07.31 | work | it is ready now |
| 02:25.30 | *** join/#asterisk fling (~fling@fsf/member/fling) |
| 03:02.47 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 03:11.35 | *** join/#asterisk ghost75 (~trechber@dslb-188-105-019-055.pools.arcor-ip.net) |
| 03:30.12 | din3sh | mrning all |
| 03:39.59 | *** join/#asterisk zemmali-voip (~zemmali@unaffiliated/zemmali-voip) |
| 03:52.00 | *** join/#asterisk mintos (~mvaliyav@112.79.45.208) |
| 03:53.34 | *** join/#asterisk FireAndIce (~FireAndIc@123.201.161.27) |
| 03:54.37 | *** join/#asterisk mjordan (~mjordan@75.76.55.191) |
| 03:54.37 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 03:58.00 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 03:58.00 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 04:02.15 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 04:15.15 | *** join/#asterisk Bradada (~Bradada@114-32-7-232.hinet-ip.hinet.net) |
| 04:24.15 | sruffell | work: Still here? I had to step away. |
| 04:37.48 | sruffell | well…if you come out, here is an example: https://gist.github.com/sruffell/5260669#file-gistfile1-txt |
| 04:37.51 | sruffell | come back I mean |
| 04:45.19 | *** join/#asterisk radic (~radic@dslb-178-010-252-092.pools.arcor-ip.net) |
| 04:46.26 | *** join/#asterisk felimwhiteley_ (~quassel@89.101.203.26) |
| 04:54.57 | *** join/#asterisk nightrid3r (~kvirc@62.205.67.197) |
| 04:56.53 | *** join/#asterisk felimwhiteley_ (~quassel@89.101.203.26) |
| 05:02.12 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 05:07.30 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 05:10.12 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.71) |
| 05:11.32 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 05:22.46 | *** join/#asterisk felimwhiteley (~quassel@89.101.203.26) |
| 05:35.43 | *** join/#asterisk felimwhiteley (~quassel@89.101.203.26) |
| 06:18.48 | *** join/#asterisk felimwhiteley_ (~quassel@89.101.203.26) |
| 06:28.02 | *** join/#asterisk moy (~moy@unvlon55-1176057127.sdsl.bell.ca) |
| 06:35.28 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 06:41.18 | *** join/#asterisk din3sh (2988ff6e@gateway/web/freenode/ip.41.136.255.110) |
| 06:44.30 | *** join/#asterisk felimwhiteley (~quassel@89.101.203.26) |
| 06:45.45 | *** join/#asterisk Alagar (~helpdesk@vsusm15.vernalissystems.com) |
| 06:50.31 | *** join/#asterisk mihamina (~mihamina@66.209.74.41-ip-dyn.orange.mg) |
| 07:04.11 | *** join/#asterisk mirela666 (~Thunderbi@212.200.146.253) |
| 07:07.31 | *** join/#asterisk felimwhiteley (~quassel@89.101.203.26) |
| 07:09.18 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 07:13.26 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 07:20.55 | *** join/#asterisk santa0536 (~santa@cn-bgp-nat.portaone.com) |
| 07:29.57 | *** join/#asterisk felimwhiteley (~quassel@89.101.203.26) |
| 07:38.06 | *** join/#asterisk HmdP_Mobile (~HmdP_Mobi@D9799130.cm-3-2c.dynamic.ziggo.nl) |
| 07:39.43 | *** join/#asterisk linocisco (~linocisco@193.134.242.12) |
| 07:41.23 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.41) |
| 07:41.35 | linocisco | hi all, I have found cisco SPA502G phone is available in local market. I would like to know if this phone is ready to be used with asterisk or we need to convert to sip first. |
| 07:43.46 | *** join/#asterisk iprouteth0 (~james@unaffiliated/iprouteth0) |
| 07:47.40 | ChannelZ | It should already be SIP |
| 07:49.32 | linocisco | ChannelZ, are u sure? |
| 07:49.57 | linocisco | ChannelZ, I have Cisc 7942G which is with SCCP by default |
| 07:50.08 | linocisco | ChannelZ, I have Cisco 7942G which is with SCCP by default. |
| 07:50.26 | ChannelZ | Well I think it can do SIP and SPCP. I have a 504G but it's not plugged in at this second, I can't remember if it's a toggle or you have to flash its brain with different firmware |
| 07:50.44 | din3sh | hello |
| 07:51.03 | ChannelZ | Those phones are the "higher end" ones which yes by default want to talk to their Communications Manager or whatever the hell its called |
| 07:51.05 | Linkforsoad | @Linocisco Change to SIP-firmware is the best option |
| 07:51.06 | din3sh | I have tried both SIP and SCCP on asterisk (cisco 79xx and 6945) |
| 07:51.40 | ChannelZ | But the 50x's generally ship with SIP, or should anyway |
| 07:51.44 | linocisco | din3sh, namastay |
| 07:51.52 | din3sh | namaste too |
| 07:51.53 | din3sh | :D |
| 07:54.10 | linocisco | din3sh, could you share me how to make Cisco 7942G work with its default SCCP and asterisk. I heard SCCP chan driver but never been ok to install on my QNAP TS-259 Pro NAS |
| 07:55.58 | din3sh | installed chan_sccp few days back |
| 07:56.09 | din3sh | i see cisco 7911 working like a charm |
| 07:57.28 | din3sh | why i tried chan_sccp? |
| 07:57.51 | din3sh | I am runnign hundreds of 79xx series since 2-3years with SIP with no problem really |
| 07:58.13 | din3sh | but if you use sip firmware 9.3.x (new one) |
| 07:58.32 | din3sh | features like cfwding, no longer work |
| 07:59.25 | din3sh | with asterisk 1.8 i mean |
| 07:59.49 | din3sh | also other glitches i discoverd like call drop on directed pickup |
| 08:02.04 | *** join/#asterisk felimwhiteley (~quassel@89.101.203.26) |
| 08:08.25 | *** join/#asterisk beardy (~beardy@unaffiliated/beardy) |
| 08:10.17 | linocisco | din3sh, mine is with asterisk 1.4 on QNAP TS-259 Pro + |
| 08:20.14 | linocisco | Linkforsoad, what I am always having as headache is converting to sip and vice versa. I crashed one NOrtel Avaya IP 1140E phone to SIP and now it is just 30 days trial version of sip . I can't revert to Unitsim anymore. I was blamed for that alot |
| 08:21.30 | iprouteth0 | the old 79xx series is a big pain to work with |
| 08:21.42 | iprouteth0 | Not sure why one would want to use SCCP or MGCP |
| 08:22.38 | iprouteth0 | I've used GNS3 to run an emulated cisco router with call manager express installed for 79xx phones in the past for testing |
| 08:31.43 | *** join/#asterisk madhatt3r (~madhatt3r@62.117.203.84.dyn.user.ono.com) |
| 08:31.58 | linocisco | iprouteth0, but i m now in testing environment like GNS3. My office has no budget for Cisco Call Manager |
| 08:32.12 | linocisco | iprouteth0, but i m not in testing environment like GNS3. My office has no budget for Cisco Call Manager |
| 08:32.16 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.41) |
| 08:33.05 | iprouteth0 | is there a reason you're wanting to use SCCP rather than sip? |
| 08:33.43 | din3sh | chan_sccp works wittyh asterisk 1.4 LSO |
| 08:33.47 | din3sh | also* |
| 08:34.02 | *** join/#asterisk Bradada (~Bradada@114-32-7-232.hinet-ip.hinet.net) |
| 08:34.16 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 08:34.48 | linocisco | iprouteth0, yes. because we will use * with cisco 7942G which has SCCP by default |
| 08:36.37 | iprouteth0 | not sure that answered my question :) It's so easy to get SIP firmware on 79xx |
| 08:37.07 | linocisco | iprouteth0, I dare not because |
| 08:38.23 | linocisco | iprouteth0, I dare not because reverting to SCCP may not be successful as I really scare due to past expereince with avaya IP 1140E |
| 08:39.29 | din3sh | linocisco: you can change firmware at will |
| 08:39.30 | *** join/#asterisk ujjain (ujjain@unaffiliated/ujjain) |
| 08:39.31 | iprouteth0 | I've never had any trouble with it provided you have the correct files |
| 08:39.46 | din3sh | i mean you can swap between sccp - sip |
| 08:40.02 | iprouteth0 | if you purchased the phones used it's a bit trickier to find the needed firmware files |
| 08:40.29 | iprouteth0 | there is a new implementation of SCCP as I understand as well |
| 08:41.05 | linocisco | actually. we have 4 pilot cisco 7942G to our offices by HQ. Our HQ plan is to deploy CUCM in the future. but |
| 08:41.11 | din3sh | yep, you need to have a Cisco login to download the firmware |
| 08:42.13 | *** join/#asterisk bulkorok (~chatzilla@85.183.36.36) |
| 08:42.29 | linocisco | but that is planned for big offices. for small offices , they don't have budget for CUCM. so I will go with asterisk , but the phone I have is only SCCP default 7942G. If I destroyed them, I won't revert them to SCCP and can't be usable with CUCM |
| 08:42.53 | iprouteth0 | I've always managed to find my firmwares without having the cisco support, but that can get hairy if you're not very careful |
| 08:43.05 | iprouteth0 | I was working with 7940s which are dirt cheap nowadays |
| 08:43.19 | iprouteth0 | didnt worry much about if they got bricked |
| 08:43.58 | linocisco | here 200 USD without adapters |
| 08:44.03 | *** join/#asterisk threesome (~threesome@customer-79-127-150-148.net.angelnet.cz) |
| 08:44.27 | iprouteth0 | CUCM can use 79xx phones loaded with SIP |
| 08:44.37 | din3sh | iprouteth0: have you tried cisco 6945? |
| 08:44.40 | iprouteth0 | some features might work differently or not be available |
| 08:44.44 | din3sh | am going nuts with this phone |
| 08:45.40 | iprouteth0 | nope, never tried the 6945. Not even sure I've seen those at work. |
| 08:46.07 | iprouteth0 | we generally deploy polycom endpoints unless the customer was sold a cisco UC500 system |
| 08:46.16 | iprouteth0 | then I think they've been the 50X series |
| 08:46.57 | din3sh | 6945 is entry level with gigabit ports |
| 08:47.12 | iprouteth0 | I'm generally not that impressed with a good handful of the cisco VoIP products. CUCM is too damn expensive |
| 08:48.52 | iprouteth0 | CUCM does some cool stuff granted, but I'd rather use asterisk or sipXecs |
| 08:49.14 | iprouteth0 | lately I've been trying to catch up on Metaswitch stuff |
| 08:49.25 | linocisco | all cisco phones are java based. if someone is java expert, you can hack |
| 08:50.50 | iprouteth0 | http://www.grandstream.com/index.php/products/ip-voice-telephony/enterprise-ip-phones/gxp2200 |
| 08:50.58 | iprouteth0 | way cooler than cisco 79xx phones |
| 08:51.25 | iprouteth0 | and pretty fair price considering what one could do with it |
| 08:51.46 | linocisco | i like grandstream but some geeks said they are not good for enterprise. |
| 08:52.11 | linocisco | I have handytone |
| 08:52.21 | linocisco | HT702 and GXW4104 |
| 08:53.36 | *** join/#asterisk Bradada (~Bradada@114-32-7-232.HINET-IP.hinet.net) |
| 08:53.48 | iprouteth0 | what grandstream in general? Don't see why they "arent good for enterprise" |
| 08:54.04 | iprouteth0 | I could see why someone would say that analogue telephone adapters are not good for enterprise |
| 08:54.11 | din3sh | am deploying hundreds of grandstream gxp 1450 |
| 08:54.23 | din3sh | pretty satisfied with their product/pricing |
| 08:54.42 | din3sh | also their support was very prompt when i had issues |
| 08:54.44 | iprouteth0 | SNOM is the one I'd really like to try |
| 08:54.51 | linocisco | the thing not interesting is their blue color display |
| 08:54.52 | iprouteth0 | they've got a few real cool models |
| 08:55.01 | din3sh | i have hundred of snoms deployed |
| 08:55.11 | linocisco | Is SNOM cheaper like USD50 for unit price? |
| 08:55.12 | din3sh | not really ideal for asterisk i'd say |
| 08:55.15 | iprouteth0 | would love to play around with the openvpn snom |
| 08:56.12 | iprouteth0 | I've the most experience with the older cisco 79xx in both asterisk and call manager express... It's been 5 years since I messed with the full call manager |
| 08:56.40 | iprouteth0 | and next most experience is with the polycom soundpoint series which is pretty excellent |
| 08:56.52 | iprouteth0 | probably spent about equal time with both |
| 08:56.58 | din3sh | also deployed polycoms, not bad |
| 08:57.11 | din3sh | managed to get the ldap directory working on them |
| 08:58.27 | iprouteth0 | BLF works very nicely on them, especially with call park and directed pickup |
| 08:58.42 | linocisco | what is BLF? |
| 08:58.49 | iprouteth0 | Got things very very close to a key system functionality |
| 08:59.10 | din3sh | for BLF to work with new frimware 9.x and asterisk 1.8.x |
| 08:59.13 | iprouteth0 | Busy Lamp Field |
| 08:59.17 | din3sh | you're in for a nice ride |
| 08:59.18 | din3sh | :D |
| 08:59.37 | linocisco | i dont know never heard BLF |
| 08:59.39 | iprouteth0 | firmware 9x on the ciscos? Never tried BLF on those |
| 09:00.10 | iprouteth0 | Many key systems do BLF. I trust you know how a key system operates from an end user perspective? |
| 09:00.55 | linocisco | yes. |
| 09:01.04 | din3sh | there's a patched version of asterisk by gareth |
| 09:01.23 | din3sh | which enables you to do blf /prsence with cisco phones on asteriak |
| 09:01.24 | iprouteth0 | ok. BLF is what shows the status. So if someone is using line 1, line 1 lights up as in use on everyone's stations |
| 09:01.26 | din3sh | asterisk * |
| 09:01.32 | iprouteth0 | thats interesting |
| 09:01.41 | iprouteth0 | never had new enough cisco phones to try BLF with |
| 09:02.00 | din3sh | https://issues.asterisk.org/jira/browse/ASTERISK-13145 |
| 09:02.08 | din3sh | i tried the patch, works ok |
| 09:02.25 | iprouteth0 | the digium phone is pretty fun to mess around with, but not sure they're established enough yet for my taste device wise |
| 09:04.23 | linocisco | i feel like using proprietary with digium phone and digium asterisk |
| 09:05.02 | din3sh | i have a d40 lying here |
| 09:05.10 | iprouteth0 | unfortunately I haven't done many deployments lately. Stopped working with my former business partner on voip since he always seemed to sell the system to business owners with weird disabilities |
| 09:05.11 | din3sh | not tested it yet though |
| 09:05.31 | iprouteth0 | had one guy that was almost blind, making training and deployment both very difficult |
| 09:05.42 | din3sh | o.O |
| 09:08.12 | din3sh | you ever tried atcom phones? |
| 09:10.18 | iprouteth0 | another guy he sold one to had some sort of mental disability and could not for the life of him figure out how to use the 7940s we sold hi |
| 09:11.02 | iprouteth0 | kills me how much difficulty people have when learning those things.... You explain it, and they say it makes sense, but they all seemed to panic when actually trying to use it |
| 09:11.33 | iprouteth0 | Thats when we tried the polycoms with BLF and call parking and directed pickup so it worked pretty well exactly like a key system |
| 09:12.07 | iprouteth0 | and he had trouble with that one, which was odd since we were switching from his old key system that was starting to fail |
| 09:12.44 | iprouteth0 | Felt pretty bad for the guy cause he was really trying hard though. Ended up getting him on an old avaya key system |
| 09:13.23 | iprouteth0 | He was amazing at his trade though which was appliance repair |
| 09:14.31 | iprouteth0 | linocisco: the digium and certified asterisk work very nicely for autoprovisioning |
| 09:15.35 | linocisco | iprouteth0, i see. but I dont prefer to use certified asterisk. and digium asterisk + digium phone way. I love open choice. I konw it is smooth as silk to use the same brand |
| 09:16.15 | iprouteth0 | Yeah, I've got mixed feelings on the digium stuff since all the cool features require certified asterisk and those are closed source binaries |
| 09:16.31 | iprouteth0 | for the DPMA |
| 09:21.49 | *** join/#asterisk HmdP_Mobile (~HmdP_Mobi@D9799130.cm-3-2c.dynamic.ziggo.nl) |
| 09:22.32 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 09:22.35 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 09:25.27 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 09:32.43 | *** join/#asterisk ghost75 (~trechber@dslb-188-105-019-055.pools.arcor-ip.net) |
| 09:40.13 | *** join/#asterisk HmdP_Mobile (~HmdP_Mobi@D9799130.cm-3-2c.dynamic.ziggo.nl) |
| 09:58.26 | skrusty | morning |
| 09:59.45 | *** join/#asterisk srp_ (~sandeep@115.119.115.26) |
| 10:00.48 | linocisco | skrusty, morning |
| 10:05.28 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 10:16.35 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 10:29.43 | linocisco | hi all |
| 10:30.29 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:6467:5070:e370:4c73) |
| 10:35.37 | *** join/#asterisk vedic (~V@183.82.27.174) |
| 10:38.15 | *** part/#asterisk mirela666 (~Thunderbi@212.200.146.253) |
| 10:38.16 | vedic | Is AGI record command and dial plan record commands different? I see that for Asterisk 1.8 there are options defined for dialplan record command but in AGI there is no mention of options. (eg: a: to append to existing recording, q: quiet (do not play a beep tone)) etc. |
| 10:38.26 | *** join/#asterisk mirela666 (~Thunderbi@212.200.146.253) |
| 10:40.50 | *** join/#asterisk tonyclewis (uid6025@gateway/web/irccloud.com/x-gskhqqhyxhhbwynq) |
| 10:44.23 | *** join/#asterisk Natureshadow (nik@shore.naturalnet.de) |
| 10:48.21 | *** join/#asterisk n3hxs (~ed@pool-108-16-94-10.phlapa.fios.verizon.net) |
| 10:50.35 | *** join/#asterisk StaRetji (~LittleAll@91.142.129.1) |
| 10:50.42 | StaRetji | Howdy folks |
| 10:51.28 | StaRetji | just quick question. If I have number 1020 in blacklist, will asterisk match only 1020 or all ids starting with 1020 |
| 10:51.30 | StaRetji | :) |
| 10:53.04 | kaldemar | depends on your blacklist. |
| 10:53.10 | StaRetji | database show blacklist gives /blacklist/1020 :jackblack so I wonder how it matches it. |
| 10:53.16 | StaRetji | thx kaldemar |
| 10:53.24 | kaldemar | what is your blacklist? |
| 10:53.31 | StaRetji | backlist of callerids |
| 10:53.40 | StaRetji | so, I block callerid 1020 |
| 10:53.42 | kaldemar | and how is it implemented? |
| 10:54.19 | StaRetji | trough exten => |
| 10:54.28 | StaRetji | I mean, it blocks 1020, that works |
| 10:54.31 | kaldemar | how? |
| 10:54.41 | kaldemar | ~pb |
| 10:54.41 | 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. |
| 10:55.07 | StaRetji | well, matches 1020 and sends to context Hangup |
| 10:55.13 | kaldemar | are you using func BLACKLIST or some other means? |
| 10:55.23 | kaldemar | show the dialplan. |
| 10:55.31 | StaRetji | I am not near the server at the moment, can't access it :( |
| 10:55.41 | kaldemar | ask again when you can. |
| 10:56.01 | StaRetji | ok, i thought to discuss about if it will drop 10201 too |
| 10:56.06 | StaRetji | as I don't want that |
| 10:56.10 | StaRetji | need only 1020 |
| 10:56.25 | StaRetji | if that is possible by using db, that is enough info for me :) |
| 10:56.26 | kaldemar | that depends on how you have implemented your blacklist. if you use func BLACKLIST, then no. |
| 10:56.57 | StaRetji | Something lik GotoIf(blacklist/callerid.... |
| 10:57.19 | kaldemar | useless. come back when you know what your dialplan really has. |
| 10:57.25 | StaRetji | anyway, it is custom name, no BLACKLIST function |
| 10:57.35 | StaRetji | ok, no problem, thanks |
| 10:59.20 | bulkorok | what can I check if I have jitter when using Monitor() ? |
| 11:00.14 | bulkorok | seems liek jitter is on the whole machine :-/ |
| 11:01.04 | kaldemar | maybe you need a jitter buffer to get rid of the jitter before monitoring it. |
| 11:06.24 | *** join/#asterisk Greenlight (~email@cpc1-dund9-0-0-cust142.16-4.cable.virginmedia.com) |
| 11:08.40 | *** join/#asterisk mjordan (~mjordan@75.76.55.191) |
| 11:08.40 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 11:15.26 | *** join/#asterisk Natureshadow (nik@shore.naturalnet.de) |
| 11:16.55 | FireAndIce | Hi everyone!! |
| 11:17.14 | FireAndIce | I've asterisk server running with dynamic realtime on localhost. |
| 11:17.37 | FireAndIce | Now, I want to shift it to the cloud. |
| 11:18.03 | FireAndIce | Could anyone recommend any solutions, as I'm new to the cloud. |
| 11:24.02 | *** join/#asterisk davlefouAMD (~david@41.225.34.79) |
| 11:24.02 | FireAndIce | I'm building an android application with appengine as the backend. |
| 11:24.02 | FireAndIce | using Mysql for dynamic realtime. |
| 11:24.02 | StaRetji | hi kaldemar, I managed to get in the server ,here it is |
| 11:24.02 | StaRetji | exten => _X.,n,GotoIf(${DB_EXISTS(blacklist/${CALLERID(num)})}?a2billing,${EXTEN},1) |
| 11:24.03 | *** join/#asterisk mahlon (~mahlon@martini.nu) |
| 11:24.04 | *** join/#asterisk tzanger (tzanger@wallace.mixdown.ca) |
| 11:24.27 | *** join/#asterisk igcewieling (~igcewieli@ip98-183-25-31.pn.at.cox.net) |
| 11:24.32 | Greenlight | It really depends on your requirements. There are companies out there that offer preconfigured hosted asterisk systems, or you could rent a VPS or a full dedicated server |
| 11:24.46 | *** join/#asterisk jeev_ (~j@unaffiliated/jeev) |
| 11:25.14 | *** join/#asterisk ivan` (~ivan@unaffiliated/ivan/x-000001) |
| 11:26.10 | *** join/#asterisk ectospasm (~ectospasm@unaffiliated/ectospasm) |
| 11:26.26 | bulkorok | kaldemar: I shouldn't need a JB... it's a "local" lan with <30 ms to all endpoints |
| 11:28.39 | FireAndIce | Greenlight, is it possible to run asterisk server on Google Compute Engine? |
| 11:31.24 | Greenlight | In theory, yes, but it depends very much on your requirements. It's not something I've tried myself. |
| 11:31.43 | *** join/#asterisk Kobaz (~kobaz@its.kobaz.net) |
| 11:31.43 | *** join/#asterisk andross (~andrew@2610:1c0:0:2:8020:fd5:e185:dded) |
| 11:31.43 | *** join/#asterisk carrar (~tim@osburn.com) |
| 11:31.43 | *** join/#asterisk dr0ck (~dr0ck@c-75-70-59-2.hsd1.co.comcast.net) |
| 11:31.43 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 11:31.43 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 11:31.43 | *** join/#asterisk tris (tristan@2001:1868:a00a::4) |
| 11:31.43 | *** join/#asterisk phunguy (santas@dhcp.i-p.org.uk) |
| 11:31.43 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 11:31.43 | *** join/#asterisk pbxbrian (~pbxbrian@unaffiliated/brian98) |
| 11:31.43 | *** join/#asterisk leedm777 (~leedm777@nat/digium/x-vldotsqyjfhrmznt) |
| 11:31.43 | *** join/#asterisk SunTsu (miyamoto@unaffiliated/suntsu) |
| 11:32.07 | *** join/#asterisk bandroidx (~bandroidx@205.185.117.117) |
| 11:32.23 | *** join/#asterisk mahlon (~mahlon@martini.nu) |
| 11:37.34 | *** join/#asterisk Mission-Critical (~MissionCr@unaffiliated/missioncritical) |
| 11:43.30 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 11:49.40 | *** join/#asterisk Matthias (~Matthias@195.16.243.99) |
| 11:58.33 | *** join/#asterisk bogrd__ (~asteriskp@115.119.115.26) |
| 12:09.19 | *** join/#asterisk blee (~blee@68.204.217.123) |
| 12:09.46 | *** join/#asterisk [TK]D-Fender (~Joe@216-191-106-165.dedicated.allstream.net) |
| 12:10.29 | *** join/#asterisk FireAndIce (~FireAndIc@123.201.161.10) |
| 12:19.02 | vedic | Is AGI record command and dial plan record commands different? I see that for Asterisk 1.8 there are options defined for dialplan record command but in AGI there is no mention of options. (eg: a: to append to existing recording, q: quiet (do not play a beep tone)) etc. |
| 12:19.11 | vedic | http://www.voip-info.org/wiki/view/record+file |
| 12:19.19 | vedic | http://www.voip-info.org/wiki/view/Asterisk+cmd+Record |
| 12:19.33 | kaldemar | StaRetji: that checks if blacklist/${CALLERID(num)} exists. not if a key starting with ${CALLERID(num)} exists. |
| 12:19.40 | Greenlight | voip info is usually very outdated, and in a lot of cases plain wrong |
| 12:20.09 | kaldemar | vedic: don't read those. |
| 12:20.28 | vedic | Greenlight: https://wiki.asterisk.org/wiki/display/AST/AGICommand_record+file |
| 12:20.37 | kaldemar | vedic: "core show application Record" in CLI shows the documentation for your version. |
| 12:20.42 | vedic | kaldemar: Here also I don't see mention of "Options" |
| 12:20.45 | [TK]D-Fender | vedic, You use AGI to call dialplan aps with the same options as norml. "EXEC ....." |
| 12:20.59 | vedic | [TK]D-Fender:ok |
| 12:21.03 | StaRetji | kaldemar: thx for reply. so, it matches whole string only. Any tip how to add another rule, I will make blacklist2? Thx |
| 12:21.19 | kaldemar | StaRetji: tip for what? |
| 12:21.22 | StaRetji | blacklist2 to consist of number starting with? |
| 12:21.33 | kaldemar | vedic: "agi show commands topic record" |
| 12:22.06 | kaldemar | oops, "agi show commands topic record file" |
| 12:22.39 | StaRetji | what should be changed in specific line, to tell asterisk match all numbers starting with 1020 |
| 12:22.47 | StaRetji | If I don't ask to much :/ |
| 12:23.40 | vedic | kaldemar: I see difference syntax of the commands (no mention of options vs mention of it). https://wiki.asterisk.org/wiki/display/AST/Application_Record https://wiki.asterisk.org/wiki/display/AST/AGICommand_record+file |
| 12:23.48 | kaldemar | StaRetji: do you want to compare 4 first digits of the caller id against keys in DB or really just 1020? |
| 12:24.30 | kaldemar | vedic: yes, they are different commands. |
| 12:24.30 | StaRetji | yes, compare first 3, for example, so that all numbers in database, that start with 102xxxxx would match |
| 12:24.51 | StaRetji | btw, really appreciate your effort |
| 12:24.52 | StaRetji | :) |
| 12:26.13 | vedic | [TK]D-Fender: EXEC is not advisable I think. Is it? Any way to use the Application command "Record" for AGI? |
| 12:26.22 | igcewieling | vedic: exec is fine |
| 12:26.23 | kaldemar | that makes no sense. if you want some digits only, why stuff whole caller id's in the DB in the first place? |
| 12:26.32 | [TK]D-Fender | vedic, What do you mean "I think"? What tells you this? |
| 12:26.50 | kaldemar | vedic: EXEC is the way. |
| 12:27.10 | [TK]D-Fender | vedic, AGI calls dialplan apps from within your script. That is its purpose |
| 12:27.33 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 12:27.42 | StaRetji | yes, I want to replace it, current blacklist was made with intention to match first digits, was not aware that I am doing it wrong |
| 12:28.15 | StaRetji | I thought if I put 102 in blacklist, it will match all numbers starting with 102 |
| 12:28.19 | vedic | [TK]D-Fender: In general many programming language avoid direct use of exec. That apart, AGI provides some direct commands which doesn't require EXEC. I doubt, why there are such direct commands and why not all be just used with EXEC |
| 12:28.20 | kaldemar | StaRetji: DB() gets you values from database. https://wiki.asterisk.org/wiki/display/AST/Selecting+Characters+from+Variables tells you how to select certain characters from variables and function return values. |
| 12:28.22 | StaRetji | I am aware that I was wrong :) |
| 12:28.40 | [TK]D-Fender | vedic, AGI EXEC.... not your LANGUAGE'S "exec" |
| 12:28.42 | [TK]D-Fender | ^^^^ |
| 12:28.57 | igcewieling | vedic: Asterisk's AGI exec is not like exec in other languages, it does not fork/exec and spawn a new process |
| 12:29.05 | [TK]D-Fender | vedic, read the chapter on AGI's commands. |
| 12:29.53 | vedic | [TK]D-Fender: Did I say they are languages? I me check, opps I didn't. I just said "I think" and then asked "Is it?" that I think enough to show that I am in doubt. and thanks that you have cleared it. |
| 12:29.55 | StaRetji | thx kaldemar, I will try to figure it out from the link you sent |
| 12:30.28 | vedic | igcewieling: Thanks |
| 12:31.38 | [TK]D-Fender | vedic, If you say you think it might be bad ... that would require a reason. "I think" isn't a question. Just a communication SNAFU..... |
| 12:32.15 | [TK]D-Fender | vedic, So AGI EXEC is the way for this. AGI only has a handful of little things that it can do that go beyond just calling * dialplan apps more or less naturally. |
| 12:32.22 | vedic | [TK]D-Fender: I don't that way. I raised a question after that. And I am continuously asking a query that show that I don't know certain things |
| 12:32.49 | [TK]D-Fender | vedic, The point of using AGI is for those tiny things (if you even need them), and so that you do complex things between each extra Asterisk related call. |
| 12:32.58 | vedic | [TK]D-Fender: thats is lovely. I got the concept now, will read the doc also as pointed. |
| 12:33.33 | [TK]D-Fender | vedic, So that you aren't calling 4 different scripts throughout your dialplan and breaking the flow for each and creating new overhead |
| 12:34.07 | vedic | I se |
| 12:34.09 | vedic | see |
| 12:36.30 | vedic | [TK]D-Fender: How can this be extended to FastAGI? is it possible? |
| 12:36.43 | [TK]D-Fender | Same thing really |
| 12:36.49 | vedic | ok |
| 12:37.33 | *** join/#asterisk RypPn_OuT (~RypPn@rosscom.co.uk) |
| 12:38.28 | [TK]D-Fender | vedic, Command-set is the same, just that it uses a socket instead of STDIN/SDOUT/SDERR |
| 12:39.02 | vedic | [TK]D-Fender: yea, I have set the tcp/ip socket and able to read agi variables. |
| 12:39.57 | [TK]D-Fender | vedic, Then you're already on your way. I typically like all my real processing to take place on the * server itself unless there are special load requirements otherwise. I don't like having 1 extra link that could break the call. |
| 12:40.11 | [TK]D-Fender | vedic, But you use whatever you have to for your circumstances. |
| 12:44.02 | StaRetji | kaldemar: would this make any sense: exten => _X./_102,1,Hangup |
| 12:44.04 | StaRetji | thx |
| 12:44.43 | vedic | [TK]D-Fender: so if I have to execute this command from FastAGI: https://wiki.asterisk.org/wiki/display/AST/Application_Record, how should I build the syntax? As I have already tried something like: sock.write("RECORD FILE %s %s %s %d %d %s\n" %(filepath, format, escape_digits, timeout, offset, beep)) but it doesn't cover the options part mentioned in https://wiki.asterisk.org/wiki/display/AST/Application_Record |
| 12:45.46 | vedic | [TK]D-Fender: This part I built based on the AGI syntax: sock.write("RECORD FILE %s %s %s %d %d %s\n" %(filepath, format, escape_digits, timeout, offset, beep)) |
| 12:45.51 | vedic | doc |
| 12:46.45 | StaRetji | I mean, would this block all callerids starting with 102? |
| 12:46.55 | StaRetji | if yes, that is good enough for me :) |
| 12:47.41 | vedic | [TK]D-Fender: I think I just built the Application command syntax not the AGI syntax mentioned in the doc |
| 12:48.44 | igcewieling | vedic: very few people use AGI directly, most people use an AGI library for their language to avoid the silliness of having to worry about protocol specific things |
| 12:50.51 | igcewieling | vedic: did you clear out (read) stdin when your script starts? If not asterisk won't accept AGI commands, as you are violating the protocol by not reading stdin |
| 12:51.39 | vedic | igcewidling: yea, I read the agi variables first and then doing writing and reading cycle |
| 12:52.21 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.69) |
| 12:52.47 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.69) |
| 12:54.29 | jmetro | WTB fully documented processes so I can learn by example rather than fudgery |
| 12:54.34 | *** join/#asterisk bchia (~Adium@nat/digium/x-lhzvsmohrvbcmdqx) |
| 12:55.23 | *** join/#asterisk serafie (~erin@nat/digium/x-txyttwmnafsgtxgu) |
| 12:55.23 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 12:56.07 | kaldemar | StaRetji: _102 would not. otherwise yes. _ denotes a pattern and you don't have anything that would require it. |
| 12:56.08 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 12:57.33 | StaRetji | yes, I got it, all starting with 102 except 102 itself |
| 12:57.54 | StaRetji | right? :) |
| 12:58.00 | kaldemar | wrong. |
| 12:58.05 | StaRetji | ehm |
| 12:58.20 | StaRetji | exten => _X./102,1,Hangup |
| 12:58.23 | StaRetji | and this? |
| 12:58.29 | kaldemar | "_102" is the same as only having "102". |
| 12:58.44 | kaldemar | StaRetji: https://wiki.asterisk.org/wiki/display/AST/Pattern+Matching |
| 12:58.53 | *** part/#asterisk vedic (~V@183.82.27.174) |
| 12:59.52 | StaRetji | thx for that, will try to test on my own sip number |
| 13:05.26 | igcewieling | <PROTECTED> |
| 13:08.16 | StaRetji | thx igcewieling, I also so in documentation that _102 quote" - matches anything starting with 102 excluding 102 itself |
| 13:08.48 | StaRetji | so, I am confused now if this line is ok or no. exten => _X./_102,1,Hangup |
| 13:09.22 | StaRetji | or it need to be exten => _X./_102.,1,Hangup |
| 13:09.31 | StaRetji | with dot? |
| 13:09.45 | kaldemar | you need to define what "ok" is. |
| 13:09.45 | StaRetji | ahhh, is dot :( |
| 13:09.52 | igcewieling | With a . will match 102 plus any number of any characters |
| 13:10.09 | StaRetji | yes, I need to match any number starting with 102 |
| 13:10.20 | kaldemar | expect 102 plus zero characters. |
| 13:10.20 | StaRetji | so, 102xxxxxxxxxxxxxxx hangup :) |
| 13:10.20 | kaldemar | *except |
| 13:10.27 | igcewieling | . is one or more characters, not 0 or more characters |
| 13:10.56 | StaRetji | so will not block 1020 |
| 13:10.59 | igcewieling | READ THIS OR YOUR LIFE WILL BE POINTLESS AND MISERABLE (8:58:44 AM) kaldemar: StaRetji: https://wiki.asterisk.org/wiki/display/AST/Pattern+Matching |
| 13:11.01 | StaRetji | only 1021 |
| 13:11.08 | *** join/#asterisk k610 (~K610@cred.epid.ucl.ac.be) |
| 13:11.19 | igcewieling | StaRetji: yes it will match 1020 |
| 13:11.26 | igcewieling | but it will not match 102 |
| 13:12.07 | igcewieling | You are now asking questions answered in the documentation |
| 13:12.22 | StaRetji | well, I thought saw, only I was not sure, sorry for that |
| 13:13.33 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 13:16.31 | StaRetji | yes, that is it, just tested it. Thx for being patient with me ;) |
| 13:16.33 | StaRetji | Cheers |
| 13:19.53 | [TK]D-Fender | There is a pattern you can do to match "anything that has 102 in front" |
| 13:20.07 | [TK]D-Fender | Read the instructions on patterns |
| 13:20.57 | StaRetji | thx Fender dude |
| 13:21.04 | StaRetji | thx kaldemar |
| 13:21.07 | StaRetji | thx igcewieling |
| 13:21.09 | StaRetji | :) |
| 13:32.34 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-sfscerkjvjnzhqxb) |
| 13:32.34 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:51.35 | *** part/#asterisk StaRetji (~LittleAll@91.142.129.1) |
| 13:53.18 | *** join/#asterisk bchia (~Adium@nat/digium/x-zyafomymrylcgbqy) |
| 13:54.51 | *** join/#asterisk HolTech (~IceChat77@pool-74-107-104-2.bltmmd.fios.verizon.net) |
| 13:56.57 | igcewieling | Customers have a rich fantasy life. "PROBLEM: WHEN THEY TRY TO DIAL OUT - THEY HEAR A RECORDING URL IS DISABLED" Perhaps they actually SEE the message? |
| 13:57.22 | HolTech | the goto reference "asterisk: the future of telephony" seems to be cover an older verison of Asterisk before zaptel.conf became chan_dahdi.conf etc, is there an updated reference pdf ? |
| 13:59.40 | igcewieling | HolTech: there are several updated editions |
| 14:00.43 | igcewieling | start at http://www.asteriskdocs.org/ |
| 14:00.45 | HolTech | igcewieling: thanks. I though I got my copy off aterisk.org, I'll double check. |
| 14:01.34 | HolTech | <PROTECTED> |
| 14:02.17 | igcewieling | you should still read the UPGRADE-*.txt files included in the Asterisk source code. |
| 14:03.42 | *** join/#asterisk SpenglerMBP (~spengler@dns1.aacc.edu) |
| 14:04.24 | HolTech | Thanks, I'm actually starting out on asterisk 1.8.18, I was using the 2nd edition pdf which was throwing me off due to so many changes in asterisk. Looks like the UPGRADE-*.txt mostly cover things to know when going from one ver of asterisk to another ? |
| 14:04.51 | WIMPy | Both are pretty old. |
| 14:05.20 | WIMPy | The 4th edition of the book is online as well as Asterisk 11. |
| 14:08.27 | HolTech | I see that. I'm pretty new so I'll switch to 11 before I get too deep. Anyone have a good reference for the sipura 3000 and asterisk, my web crawling hasn't provided much with an overview of the device's config and the manufacturer docs are slim |
| 14:12.28 | [TK]D-Fender | HolTech, There's plenty of guides to Google up |
| 14:21.08 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 14:24.26 | *** join/#asterisk [TK]D-Fender (~Joe@216-191-106-165.dedicated.allstream.net) |
| 14:27.57 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 14:37.21 | *** join/#asterisk bchia (~Adium@nat/digium/x-zvcwgbkdpjgmaoet) |
| 14:39.38 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 14:39.38 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 14:41.32 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 14:41.53 | *** join/#asterisk DarthExpeditor (~IceChat9@71.46.209.2) |
| 14:42.11 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 14:42.59 | *** join/#asterisk thecardsmith (~quassel@pdpc/supporter/student/thecardsmith) |
| 14:45.06 | *** join/#asterisk fling (~fling@fsf/member/fling) |
| 14:51.47 | *** join/#asterisk teknoprep (~Chris@unaffiliated/teknoprep) |
| 14:51.48 | teknoprep | hi all |
| 14:52.11 | teknoprep | i was wondering... with a queue and autopause=yes... how long does that agent become paused if the agent does not answer the call ? |
| 14:52.15 | teknoprep | is it just for that one call ? |
| 14:54.10 | *** join/#asterisk areski (~areski@81.184.35.151.dyn.user.ono.com) |
| 14:55.02 | *** join/#asterisk davlefouAMD (~david@197.15.210.67) |
| 14:57.16 | leifmadsen | teknoprep: indefinitely until they unpause |
| 14:57.22 | teknoprep | how do they unpause ? |
| 14:57.38 | leifmadsen | use the dialplan application |
| 14:57.42 | teknoprep | gotcha |
| 14:57.52 | leifmadsen | build a star code or something like that |
| 14:58.03 | leifmadsen | or use a web interface to connect to AMI and do it that way |
| 14:58.11 | leifmadsen | (if you're building a web interface) |
| 14:58.17 | teknoprep | next question.. if someone doesn't log out of the queue... i am using LeastRecent... and the 15seconds pass's... isn't it supposed to ring a different Agent ? |
| 14:58.26 | teknoprep | or will it keep rining the same agent ? |
| 14:58.55 | leifmadsen | I can't remember if LeastRecent means least recent answered or dialed -- I think answered |
| 14:58.58 | leifmadsen | so same agent unless you pause |
| 14:59.13 | leifmadsen | easy test to make though |
| 14:59.21 | leifmadsen | try it and find out |
| 14:59.30 | teknoprep | thats whats happening |
| 14:59.34 | teknoprep | it keeps ringing the phone |
| 14:59.43 | leifmadsen | there you go then |
| 14:59.46 | teknoprep | should i use rrmemory ? |
| 14:59.48 | leifmadsen | pause the agent so it moves on |
| 14:59.51 | leifmadsen | I don't know... should you? |
| 15:00.09 | teknoprep | well i can't have it keep rinigng someone that didn't rememeber to log out |
| 15:00.18 | teknoprep | and i can't for now use autopause |
| 15:00.20 | leifmadsen | that's what autopause is fore.... |
| 15:00.25 | leifmadsen | then use a different method |
| 15:00.43 | teknoprep | will rrmemory move to the next agent if the first agent doesn't pick up after 15seconds ? |
| 15:00.48 | teknoprep | since i ring agents for 15seconds |
| 15:00.53 | teknoprep | with the Q |
| 15:01.42 | leifmadsen | testing will be your savior |
| 15:01.46 | leifmadsen | that's a simple test to do |
| 15:05.36 | teknoprep | its on an active system |
| 15:05.39 | igcewieling | teknoprep: a technical solution to user stupidity may not work well. |
| 15:05.41 | teknoprep | i added an asterisk Q to our current SIP PBX |
| 15:05.52 | teknoprep | i agree |
| 15:05.58 | teknoprep | i am going to use rrmemory to |
| 15:06.02 | teknoprep | tho and see what happens |
| 15:06.18 | igcewieling | or you could just fire people who repeatedly don't log out. |
| 15:06.36 | *** join/#asterisk LiuYan (~LiuYan@211.154.128.171) |
| 15:07.16 | *** join/#asterisk HmdP_Mobile (~HmdP_Mobi@D9799130.cm-3-2c.dynamic.ziggo.nl) |
| 15:08.31 | teknoprep | not my company |
| 15:08.34 | teknoprep | just a consultant |
| 15:08.44 | teknoprep | and this gig isn't a business consulting situation |
| 15:08.45 | teknoprep | lol |
| 15:08.55 | leifmadsen | wait, you're consulting for these guys? |
| 15:08.57 | leifmadsen | interesting. |
| 15:09.06 | teknoprep | already done the job |
| 15:09.15 | teknoprep | switched it to RRMEMORY and everything worked |
| 15:09.27 | teknoprep | 120hours billable |
| 15:09.29 | leifmadsen | where do I sent my invoice to? |
| 15:09.32 | teknoprep | lol |
| 15:09.37 | leifmadsen | no seriously. |
| 15:09.57 | leifmadsen | sounds like you just billed for my answer. |
| 15:10.24 | teknoprep | it was a project job.. i already was paid up front |
| 15:10.32 | igcewieling | things Google would have been cheaper. |
| 15:11.10 | *** join/#asterisk Rico29 (~rico@oceanet-telecom-fttb-129-2.olm.fr) |
| 15:11.13 | Rico29 | hi all |
| 15:11.19 | teknoprep | why would google be cheaper ? |
| 15:11.47 | Rico29 | can anybody tell me why this line does not play the "beep" after the first message ? |
| 15:11.49 | Rico29 | $AGI->exec('Read','num_abo,/home/asterisk_sounds/svi/eforsports/accueil1,6,,,10'); |
| 15:11.49 | WIMPy | Because Leif isn't cheap. |
| 15:12.05 | Rico29 | first sound file is well played |
| 15:12.13 | Rico29 | but beep is not |
| 15:12.20 | *** join/#asterisk kontinuity (~kontinuit@122.167.101.120) |
| 15:12.25 | leifmadsen | WIMPy: +1 |
| 15:12.49 | WIMPy | Rico29: I have no idea what language you're using there, but are you sure the parameters go into the quotes? |
| 15:12.58 | Rico29 | WIMPy> yes |
| 15:13.02 | Rico29 | (it's in perl agi) |
| 15:13.34 | *** join/#asterisk davlefouAMD (~david@197.15.210.67) |
| 15:19.28 | *** join/#asterisk Natureshadow (nik@shore.naturalnet.de) |
| 15:22.04 | igcewieling | Rico29: look at "core show application read" and pay careful attention to the syntax shown |
| 15:22.36 | [TK]D-Fender | Rico29, https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Read |
| 15:22.46 | Rico29 | thakns, Ive read it |
| 15:22.54 | [TK]D-Fender | Rico29, Nowhere do I see a beep being played as even being an option |
| 15:22.57 | Rico29 | my problem was not on this line.....; |
| 15:23.08 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 15:23.08 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 15:23.20 | igcewieling | heh, [TK]D-Fender the syntax on that page has a typoe |
| 15:23.24 | Rico29 | $AGI->exec('Read','num_abo,/home/asterisk_sounds/svi/eforsports/accueil1&beep,6,,,10'); works fine |
| 15:23.26 | Rico29 | thanks |
| 15:24.11 | *** join/#asterisk hexanol (~bibi@modemcable094.94-70-69.static.videotron.ca) |
| 15:26.38 | [TK]D-Fender | igcewieling, As long as entire options aren't lost in the typo ... ok/fine/sure :) |
| 15:32.23 | *** join/#asterisk HmdP_Mobile (~HmdP_Mobi@D9799130.cm-3-2c.dynamic.ziggo.nl) |
| 15:36.02 | *** join/#asterisk HmdP_Mobile (~HmdP_Mobi@D9799130.cm-3-2c.dynamic.ziggo.nl) |
| 15:51.01 | *** join/#asterisk SuperNull (~FreeManof@24-148-101-238.ip.mhcable.com) |
| 15:54.33 | teknoprep | i have another question... call comes into SBC --- openUC --- IVR ---> call option is selected which is a call Q on the Asterisk Box --- call is reinvited from OpenUC to Asterisk box through SBC... call q picks up on asterisk... 2nd ext on phones for q is run on those agent phones... the agent answers... the agent transfers back to a caller on the openuc box... i was wondering if there was a way to reinvite from asterisk to the OpenUC box so th |
| 15:54.34 | teknoprep | at asterisk isn't another B2BUA... this way we don't eat up 4sessions on the SBC |
| 15:59.08 | mjordan | igcewieling: yes it does |
| 15:59.12 | *** join/#asterisk kontinuity (~kontinuit@122.167.101.120) |
| 15:59.56 | mjordan | looks like a quirk in the XML => wiki script. |
| 15:59.56 | WIMPy | Did you already pay for the last answer? |
| 16:07.08 | igcewieling | teknoprep: even if you reinvite that only takes Asterisk out of the media path, not the signaling path. |
| 16:07.59 | teknoprep | yeah thats what i wanted |
| 16:08.04 | teknoprep | out of the Signaling path |
| 16:08.06 | teknoprep | np |
| 16:08.28 | teknoprep | this is only a temp solution to get rid of the Avaya PBX they currently have |
| 16:08.40 | teknoprep | waitig on an Update from there new PBX vendor for the call queuing they require |
| 16:08.48 | teknoprep | just temporarily setup a call q system on Asterisk |
| 16:11.24 | Greenlight | Whos the new PBX vendor ? |
| 16:12.02 | *** join/#asterisk classix (~salven@silenceisdefeat.com) |
| 16:14.27 | teknoprep | ezuce / openuc |
| 16:14.59 | teknoprep | its just the guys that took over SipXecs |
| 16:15.28 | teknoprep | under 200 phones we setup asterisk... larger phones systems we use openUC |
| 16:15.52 | teknoprep | if a customer wants UC.. integration to Exchange... and HA... no matter what their size we use OpenUC |
| 16:33.48 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.69) |
| 16:34.22 | *** join/#asterisk bchia (~Adium@nat/digium/x-ximsexjzytmtrdix) |
| 16:35.39 | HolTech | would a channel be "active" in "core show channels" if no call is currently going on ? |
| 16:37.52 | igcewieling | HolTech: no. |
| 16:38.15 | igcewieling | by "active" I assume you mean "shows up" as there is no call state "active" in my core show channels |
| 16:39.26 | HolTech | hm, my CLI ( on 11.2.2) output to the command "core show channels" has 3 lines, active channels, active calls, calls processed |
| 16:43.17 | igcewieling | you mean collumns. I have these: Channel Location State Application(Data) |
| 16:44.16 | HolTech | negative, i have the same columns as you mention, but 3 rows as I mentioned |
| 16:44.40 | igcewieling | ah, at the END |
| 16:45.02 | igcewieling | in any case, active calls are active calls |
| 16:45.17 | HolTech | understood, thanks |
| 16:49.51 | *** part/#asterisk bchia (~Adium@nat/digium/x-ximsexjzytmtrdix) |
| 16:52.20 | chuckf | mmmm... breakfast for lunch |
| 16:57.40 | Katty | eyes Chotaire |
| 16:57.41 | Katty | oh |
| 16:57.45 | Katty | eyes chuckf |
| 17:01.23 | *** part/#asterisk jsjc (~Adium@227.Red-2-136-107.dynamicIP.rima-tde.net) |
| 17:01.41 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 17:02.27 | chuckf | eyes Katty |
| 17:04.43 | chuckf | I'm actually quite surprised at how many people around here were weired out by the concept |
| 17:06.56 | Katty | breakfast is delicious anytime! |
| 17:07.06 | Katty | because breakfast is... delicious. |
| 17:07.15 | chuckf | yes it is |
| 17:07.47 | igcewieling | breakfast is not a time it is a philosophy. |
| 17:07.48 | chuckf | eggs, bacon, sausage, ham, pancakes, french toast. They had it all |
| 17:08.34 | chatran | who have this breakfast ? |
| 17:08.44 | chuckf | $day_job |
| 17:08.53 | chuckf | they supplied it:) |
| 17:08.57 | chatran | we have only coffee |
| 17:09.18 | *** join/#asterisk navaismo (~navaismo@189.241.122.125) |
| 17:09.40 | chuckf | They feed us lunch once a week and donuts once a week |
| 17:10.03 | chatran | :( i want donuts |
| 17:10.12 | chatran | send some to brazil |
| 17:10.29 | igcewieling | I wonder if sourdough donuts exist. |
| 17:10.36 | chuckf | They'll be sent out this afternood |
| 17:10.46 | WIMPy | That sounds perverted |
| 17:10.54 | igcewieling | indeed they do. |
| 17:10.59 | igcewieling | I'll have to try making them |
| 17:11.45 | chuckf | http://nighttrekkersguild.org/prajantr/?p=1377 |
| 17:11.56 | chuckf | ^^ sourdough donuts |
| 17:12.04 | teknoprep | is there anyway to get asterisk to "reload" without "restarting or dropping calls" when i need to get an updated sip_nat.conf ? |
| 17:12.13 | *** join/#asterisk threesome (~threesome@ip-94-113-12-74.net.upcbroadband.cz) |
| 17:13.01 | [TK]D-Fender | teknoprep, First ... that isn't even an Asterisk config file |
| 17:13.03 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 17:13.16 | [TK]D-Fender | teknoprep, Second, a reload never drops any kind of calls. |
| 17:13.20 | teknoprep | yeah i know.. its an #include i have been using for years |
| 17:13.33 | [TK]D-Fender | teknoprep, The only thing that would is if a module was directly reloaded, not just a config |
| 17:13.40 | teknoprep | i just never had the need to change nat settings.. and then reload asterisk while calls where running |
| 17:13.49 | teknoprep | gotcha |
| 17:15.36 | igcewieling | chuckf: I make sourdough bread from scratch, might try donuts sometime soon |
| 17:16.25 | chuckf | igcewieling: it sounds like something interesting to try |
| 17:20.07 | *** join/#asterisk edong23 (~quassel@mptc-dhcp-50-220.mptelco.com) |
| 17:23.00 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:23.00 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 17:23.28 | *** join/#asterisk Free99 (~Free99@cpe-66-108-105-10.nyc.res.rr.com) |
| 17:23.44 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 17:24.51 | Free99 | hey guys. I tried putting "tlsbindport=5039" into my manager.conf, but for some reason asterisk is saying it is an invalid keyword |
| 17:25.03 | Free99 | under the [general] section btw |
| 17:26.15 | teknoprep | is there a way to log out of the q like so |
| 17:26.19 | teknoprep | 9601**201 |
| 17:26.28 | teknoprep | if i dial that should it log out 201 from 9601 q |
| 17:27.18 | [TK]D-Fender | Free99, Becauese it isn't a valid parameter name |
| 17:27.20 | [TK]D-Fender | Free99, http://svnview.digium.com/svn/asterisk/branches/11/configs/manager.conf.sample?revision=371121&view=markup |
| 17:27.40 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:27.40 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 17:27.40 | [TK]D-Fender | teknoprep, STILL the wrong channel.... |
| 17:28.08 | *** join/#asterisk HolTech (~IceChat77@pool-74-107-104-2.bltmmd.fios.verizon.net) |
| 17:28.15 | Free99 | weird. O'reilly has a mistake in it then |
| 17:28.47 | [TK]D-Fender | Always refer to the * docs direct first |
| 17:29.40 | Free99 | thanks [tk] |
| 17:30.03 | igcewieling | Free99: all the sample config files are included with the asterisk source |
| 17:34.21 | *** join/#asterisk tallest_red (~CNZ@ip98-169-197-57.dc.dc.cox.net) |
| 17:44.25 | *** join/#asterisk jermey_g (jermey_g@c-9b9b72d5.021-183-73746f3.cust.bredbandsbolaget.se) |
| 17:44.27 | jermey_g | hi |
| 17:45.12 | *** join/#asterisk t (tom@freenode/staff/tomaw) |
| 17:45.40 | *** join/#asterisk thecardsmith (~quassel@pdpc/supporter/student/thecardsmith) |
| 17:50.54 | *** join/#asterisk angler (~angler@pdpc/sponsor/digium/angler) |
| 17:50.55 | *** mode/#asterisk [+o angler] by ChanServ |
| 17:52.46 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 18:03.09 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 18:09.03 | *** join/#asterisk Rahail (~Rahail@67.214.121.189) |
| 18:09.23 | *** join/#asterisk thecardsmith (~quassel@pdpc/supporter/student/thecardsmith) |
| 18:12.16 | *** join/#asterisk protocoldoug (~quassel@pdpc/supporter/student/thecardsmith) |
| 18:16.20 | *** join/#asterisk pcAngel (yoink@S0106602ad072bdf0.vc.shawcable.net) |
| 18:17.57 | pcAngel | I have a client on a remote PBX that is trunking through me, and I need to change things based on a ref tag in the From part of the message, for example the 124 in From: "iConnected" <sip:iconnected@neptune.xencall.com>;tag=xxxx;ref=124 |
| 18:18.19 | pcAngel | Is there a way in the dialplan to look at the ref=124 part of that ? |
| 18:20.33 | pcAngel | or alternatively, to match it to a separate sip peer entry based on ref=XXX ? |
| 18:26.12 | pcAngel | or a way to run a regex on ${CHANNEL(from)} to locate the the 124 ? |
| 18:32.13 | leifmadsen | pcAngel: You could pass it to an AGI() to do real regex, or match using dialplan regex |
| 18:32.20 | leifmadsen | REGEX() function |
| 18:32.29 | leifmadsen | it's just for matching though and returns true/false |
| 18:32.54 | protocoldoug | core show function CUT |
| 18:33.00 | protocoldoug | you can cut apart a string by delimiters |
| 18:36.05 | *** join/#asterisk protocoldoug (~quassel@pool-72-92-156-9.burl.east.myfairpoint.net) |
| 18:37.11 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 18:38.49 | *** part/#asterisk pigpen (~mark@fw.seamans.cc) |
| 18:39.02 | *** join/#asterisk protocoldoug (~quassel@pool-72-92-156-9.burl.east.myfairpoint.net) |
| 18:39.05 | *** join/#asterisk Dovid (~Dovid@static-173-63-105-210.nwrknj.fios.verizon.net) |
| 18:41.00 | *** join/#asterisk tonyclewis (uid6025@gateway/web/irccloud.com/x-rhzksqtckcfarpfg) |
| 18:41.52 | HolTech | a softphone such as Zoiper would be connecting to my asterisk pbx to register on what port ? 5060 ? |
| 18:42.13 | leifmadsen | whatever port you tell it |
| 18:42.21 | leifmadsen | asterisk normally listens on 5060/UDP |
| 18:42.51 | *** join/#asterisk GameGamer43 (uid5533@gateway/web/irccloud.com/x-ayhzsbujmhbbkppp) |
| 18:46.56 | protocoldoug | if I'm using Zoiper, I like to use IAX (typically 4569), fwiw |
| 18:48.12 | *** join/#asterisk evil-man (~evil-man@insider-mail.icf.org.ru) |
| 18:48.23 | HolTech | I'm trying to connect with Zoiper from a windows machine, but watching the netstat on my pbx to see if it's connecting... but not seeing anything. Firewall is already punched. |
| 18:48.32 | HolTech | for 5060 at least... and zoiper is set for SIP |
| 18:49.24 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.95) |
| 18:50.23 | *** part/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 18:52.27 | *** join/#asterisk Sorcier_FXK (~nssystem@unaffiliated/sorcierfxk) |
| 19:10.02 | *** join/#asterisk blee (~blee@68.204.217.123) |
| 19:10.44 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 19:10.44 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 19:17.02 | slav3_kitten | i entirely read that nick wrong |
| 19:17.18 | slav3_kitten | runs away before self incrimination |
| 19:20.02 | *** join/#asterisk protocoldoug (~quassel@unaffiliated/protocoldoug) |
| 19:29.58 | *** join/#asterisk vlad_sta_ (~vlad_star@83.220.236.33) |
| 19:41.11 | *** topic/#asterisk by mjordan -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.3.0 (2013/03/28), 10.12.2 (2013/03/27), 1.8.21.0 (2013/03/28), DAHDI-linux 2.6.2 (2013/03/08), DAHDI-tools 2.6.2 (2013/03/08), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 19:43.40 | chatran | slav3_kitten what nick |
| 19:44.17 | slav3_kitten | wouldn't you like to know |
| 19:44.32 | chatran | areski ? |
| 19:45.12 | slav3_kitten | nah |
| 19:45.20 | chatran | free69 ? |
| 19:45.25 | slav3_kitten | nope |
| 19:45.29 | chatran | hum |
| 19:45.35 | chatran | chanserv? |
| 19:45.37 | slav3_kitten | 2 down, 192 left to go |
| 19:45.54 | chatran | protocoldoug ? |
| 19:46.14 | slav3_kitten | neg |
| 19:46.32 | chatran | vlad_sta_* ? |
| 19:46.52 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 19:46.52 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 19:46.55 | sp00kz | ε/̵͇̿̿/’̿’̿ ̿ ̿̿ ̿ ̿””(•̪●) |
| 19:47.00 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.33) |
| 19:47.35 | chatran | sp00kz you goon on asterisk ? |
| 19:47.40 | chatran | *good ? |
| 19:48.07 | chatran | i need configurations for pap2t works well |
| 19:48.23 | chatran | i have one but the other side do no t listen me |
| 19:50.47 | chatran | -mquin- [Global Notice] Seems we're having a few network problems, please bear with us while we look into it ? |
| 19:51.04 | chatran | they want me drink a beer ? |
| 19:51.05 | newtonr | chatran: your SIP calls connect, but you only have audio in one direction? |
| 19:51.18 | chatran | newtonr |
| 19:51.25 | chatran | one way do not listen |
| 19:51.37 | chatran | if i call... we can talk good |
| 19:51.49 | newtonr | chatran: so you get audio in both directions, but the guy on the other end just refuses to listen to you? |
| 19:52.05 | chatran | yes |
| 19:52.15 | chatran | look what error its give me |
| 19:52.55 | chatran | <PROTECTED> |
| 19:53.04 | chatran | this is nat problem |
| 19:53.09 | igcewieling | chatran: looks to be a nat problem |
| 19:53.13 | chatran | but i see doc, maybe not is |
| 19:53.25 | igcewieling | fix your setup and it should work |
| 19:53.30 | chatran | maybe is firewall block something |
| 19:53.46 | chatran | igcewieling man, im trying fix this... like one weak |
| 19:54.04 | chatran | someone have basic configs to give me ? |
| 19:54.23 | chatran | i did try a lot configs but no one did right |
| 19:54.41 | igcewieling | chatran: the configs depend on your setup. 1) is the ASTERISK server behind NAT? 2) are one or more of the SIP CLIENTS behind NAT? 3) are they behind a different NAT than ASTERISK? |
| 19:54.52 | navaismo | chatran, fix your nat settings in asterisk & pap2t |
| 19:54.57 | newtonr | chatran: in that kind of situation, basic configs are not likely to help you. you'd have to detail your whole network environment and show pcaps for someone to help |
| 19:55.01 | chatran | igcewieling this is my config: |
| 19:55.18 | *** join/#asterisk Dovid (~Dovid@static-173-63-105-210.nwrknj.fios.verizon.net) |
| 19:55.19 | chatran | asterisk is direct on internet, no firewalls |
| 19:55.29 | chatran | client is pap2t behind common modem |
| 19:55.54 | chatran | its the most common possible config |
| 19:56.05 | chatran | and i still having this "nat" problem |
| 19:56.26 | igcewieling | Make sure in [general] sip.conf you have nat=yes directmedia=no and do NOT have localnet= or externip= settings. Remove any port forwarding on the NAT firewall, turn off any NAT related stuff on the PAP |
| 19:57.04 | igcewieling | then do a "module reload chan_sip.so" in Asterisk then reboot the PAP |
| 19:57.06 | chatran | ok i will try this |
| 19:58.15 | chatran | i did factory reset |
| 19:58.30 | chatran | now i go search for nats configs on pap2t |
| 19:58.56 | igcewieling | no! do NOT set up NAT on the PAP2 |
| 19:59.07 | igcewieling | if you do so it will interfere with asterisk's nat=yes setting. |
| 19:59.11 | *** join/#asterisk FireAndIce (~FireAndIc@123.201.161.4) |
| 19:59.22 | chatran | im searching to find if exist something about nat.... if have i will remove |
| 19:59.36 | igcewieling | chatran: I just provided you with all the information you need |
| 19:59.40 | newtonr | chatran: just try them at the default first |
| 19:59.53 | chatran | newtonr i already tryed |
| 19:59.59 | igcewieling | The most common issue with Asterisk and NAT is people make it overly complicated. |
| 20:00.00 | chatran | but gives same message |
| 20:00.09 | chatran | i think is on server the problem |
| 20:00.17 | igcewieling | chatran: looks like you do NOT have directmedia=no on your asterisk box |
| 20:00.21 | chatran | i will search if have nat config on server soon |
| 20:00.22 | igcewieling | (3:56:26 PM) igcewieling: Make sure in [general] sip.conf you have nat=yes directmedia=no and do NOT have localnet= or externip= settings. Remove any port forwarding on the NAT firewall, turn off any NAT related stuff on the PAP |
| 20:00.28 | chatran | is |
| 20:00.30 | chatran | yes |
| 20:00.36 | chatran | wait a little |
| 20:00.37 | chatran | ... |
| 20:00.53 | igcewieling | chatran: so it is the EXACT OPPOSITE of what I told you to set it. |
| 20:01.08 | chatran | ? |
| 20:01.20 | chatran | nat = no and directmedia = yes ? |
| 20:01.28 | igcewieling | Me: Looks like you do not have directmedia=no. You: is yes |
| 20:02.00 | igcewieling | chatran: did I say anyhwere to set it to yes? I told you several times to set directmedia=no. If you do not do that you will not have working audio |
| 20:02.00 | chatran | igcewieling keep calm im bad english writer |
| 20:02.27 | chatran | i will search to find if i have directmedia setting |
| 20:02.29 | chatran | ... |
| 20:02.31 | chatran | you too fast |
| 20:03.56 | *** part/#asterisk hexanol (~bibi@modemcable094.94-70-69.static.videotron.ca) |
| 20:05.20 | chatran | ok |
| 20:05.28 | chatran | i will try |
| 20:08.28 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 20:21.24 | igcewieling | stops channeling [TK]D-Fender |
| 20:26.38 | *** join/#asterisk MrMeek (~meekhime@172-4-223-5.lightspeed.toldoh.sbcglobal.net) |
| 20:30.30 | [TK]D-Fender | heads out for a while |
| 20:31.01 | igcewieling | Customers make me grumpy. |
| 20:33.12 | paulc | Life would be great if it wasn't for (certain) customers.. |
| 20:40.30 | igcewieling | Part of the problem is that it seems most of the people who change phone companies are generally unhappy people anyway. If they were not chronically unhappy they would not be switching phone companys |
| 20:45.43 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-sfscerkjvjnzhqxb) |
| 20:48.53 | HolTech | ok, this truly newbie question, so thanks in advance and hold the insults please. So, in a dialplan the line exten => s,2,Dial(SIP/123,60,r) would send the call to a SIP device with extension 123 correct? now where is the 123 defined... is it what i named it between the brackets in my sip.conf ie [ ... ] ? |
| 20:51.50 | *** join/#asterisk BrokenArrow (~BrokenArr@unaffiliated/brokenarrow) |
| 20:56.53 | pcAngel | Is there any functions in asterisk that will get the value of a key, from a string encoded like key1=value1;key2=value2;key3=value3 ? |
| 20:57.07 | pcAngel | that format is the format of ${SIP_HEADER(from)} |
| 20:58.06 | igcewieling | HolTech: It would send the call to the device registered to the [123] section of sip.conf. that is not an extension, that is a peer. N00bs often make the peer name the same as the extension, but it is purely cosmetic |
| 20:58.21 | pcAngel | HolTech: yes, that is correct, the terminology is a bit difference though -- Dial(SIP/123) will dial the SIP device named 123, and in sip.conf the device name is stored between the brackets |
| 21:11.45 | igcewieling | pcAngel: nothing directly does that but see "core show function CUT" and the FIELD* functions (core show functions) |
| 21:12.51 | igcewieling | Does anyone happen to know what prodding messages are? [Mar 28 17:12:17] WARNING[3377][C-00025b9c]: channel.c:4810 ast_prod: Prodding channel 'SIP/vzb-0004e3d9' failed |
| 21:25.04 | igcewieling | Google seems to think it is an SCCP thing |
| 21:39.02 | *** join/#asterisk kontinuity (~kontinuit@122.167.101.120) |
| 21:52.40 | *** join/#asterisk Sorcier_FXK (~nssystem@unaffiliated/sorcierfxk) |
| 21:53.45 | HolTech | trying to Playback(hello-world), CLI says its "in new stack" but I'm not hearing anything... calling into pbx from landline. any suggestions? |
| 21:58.55 | HolTech | f'it, gonna take NAT out of the equation |
| 22:05.58 | *** join/#asterisk k610 (~K610@cable-78.29.241.186.coditel.net) |
| 22:07.22 | igcewieling | HolTech: what pstn interface card are you using? |
| 22:08.42 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 22:08.42 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 22:23.56 | *** join/#asterisk Chotaire (chotaire@chotaire-home.vipri.net) |
| 22:34.17 | *** part/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 22:50.34 | *** join/#asterisk ghost75 (~trechber@dslb-188-105-019-055.pools.arcor-ip.net) |
| 22:53.10 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
| 23:14.13 | *** join/#asterisk Gugge (gugge@kriminel.dk) |
| 23:22.25 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 23:40.30 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 23:40.30 | *** mode/#asterisk [+o pabelanger] by ChanServ |