| 00:00.03 | nic_oz | eg: the receptionist is on the phone, 2 nd line rings, when she finished the 1st call I'd like to make her phone ring again |
| 00:02.28 | TokyoJimu | nic_oz: look at the application AgentLogin. Maybe that's some kind of ACD group. |
| 00:03.12 | paulc | yeah - have yuor inbound calls go to a queue and make her an agent.. works a treat :) |
| 00:03.40 | TokyoJimu | But her phone won't ring. It's really for agents who receive one call after another. |
| 00:04.22 | TokyoJimu | Can you use call queues without agents? |
| 00:04.27 | ChulJin | yup |
| 00:04.35 | ChulJin | AddQueueueMember |
| 00:04.49 | ChulJin | oops, extra ue there |
| 00:04.53 | ChulJin | AddQueueMember |
| 00:04.56 | _Vile | ueueueueueueue |
| 00:05.25 | ChulJin | AddddQueueueueueueueueMember |
| 00:05.34 | ChulJin | seriously, though...look at AddQueueMember. |
| 00:05.36 | _Vile | 190 channels on what HW? |
| 00:05.39 | ChulJin | does exactly what you want. |
| 00:05.44 | aminorex | FfuullDduupplleexx? |
| 00:05.52 | TokyoJimu | Tried both T400P and TE405P with same results. |
| 00:05.56 | _Vile | etthhheeeerrrrnnneeeeetttttt |
| 00:06.31 | _Vile | so you're trying 2 4 ports in a box filled w/ Ts and are getting issues? what cpu? |
| 00:06.37 | *** join/#asterisk kpfleming (~chatzilla@ip68-3-230-141.ph.ph.cox.net) |
| 00:07.02 | *** join/#asterisk bilo (nabeelj@69.158.61.207) |
| 00:07.17 | kpfleming | anyone know of a better way to exit from a macro than just putting a "break" in the priority sequence? i want to just "return" from the macro back to the original point based on a condition |
| 00:07.20 | TokyoJimu | We actually have 3 cards. Dual Athlon MP 2800+. Load average is about 2 so CPU looks OK. But the 191st call fails with errors like: |
| 00:07.41 | TokyoJimu | Dec 14 15:44:00 WARNING[1215]: Unable to start PBX on Zap/201-1 |
| 00:07.41 | TokyoJimu | Dec 14 15:44:00 WARNING[1215]: Failed to create update thread! |
| 00:07.41 | TokyoJimu | Dec 14 15:44:00 WARNING[1215]: Unable to start PBX on channel 0/9, span 9 |
| 00:07.43 | TokyoJimu | Dec 14 15:44:00 WARNING[1215]: Call specified, but not found? |
| 00:07.43 | TokyoJimu | Dec 14 15:44:00 WARNING[1215]: Hangup on bad channel 0/9 on span 9 |
| 00:08.02 | TokyoJimu | Some thread limit? |
| 00:08.06 | _Vile | interesting how many interrupts/second are you pushing? |
| 00:08.19 | _Vile | :) |
| 00:08.21 | twisted[work] | kpfleming, gotoif() |
| 00:08.27 | TokyoJimu | How to tell? |
| 00:08.33 | twisted[work] | pass the current context to the macro |
| 00:08.36 | _Vile | what OS? |
| 00:08.46 | TokyoJimu | Fedora Core 1. |
| 00:08.55 | paulc | delayed reply.. TokyoJimu: You can make her phone ring, using agentcallbacklogin |
| 00:08.57 | _Vile | check out mpstat -- sysstat |
| 00:09.00 | kpfleming | twisted: eww, that's icky... i'd rather just have an app that returned an appropriate value so that Macro() would drop out of its loop |
| 00:09.16 | _Vile | twisted, would you ever put 3 4 port cards on a box? |
| 00:09.24 | _Vile | and fill it with T1s? |
| 00:09.26 | twisted[work] | _Vile, sure, why not? |
| 00:09.28 | *** join/#asterisk Kumbang (~ecvs@167.205.22.54) |
| 00:09.52 | _Vile | hmm |
| 00:09.53 | TokyoJimu | Yes, we do it all the time. |
| 00:10.12 | twisted[work] | kpfleming, it's not in a loop |
| 00:10.16 | TokyoJimu | Worked on old asterisk with RH 7.3. |
| 00:10.31 | ZX81 | it's something -blank 0 |
| 00:10.32 | TokyoJimu | setterm -blank 0 |
| 00:10.35 | ZX81 | ta |
| 00:10.36 | ZX81 | :-) |
| 00:10.41 | kpfleming | twisted: uhh, yeah, app_macro uses a while loop to execute the macro contents, as best i can tell |
| 00:11.58 | *** join/#asterisk channan (~channan9@66.180.121.185) |
| 00:12.13 | twisted[work] | hell |
| 00:12.18 | twisted[work] | the dialplan uses a while loop |
| 00:12.24 | twisted[work] | if you wanna get into the code |
| 00:14.25 | kpfleming | right... so i want to be able to tell app_macro's while loop "hey, i'm done, just return back to where you were called from", without having to compute that location, since it's already known |
| 00:14.33 | kpfleming | i will make a patch and post it, in case anyone else wants it |
| 00:14.52 | twisted[work] | you can't. |
| 00:14.57 | twisted[work] | not currently |
| 00:15.01 | twisted[work] | so you will have to make a patch |
| 00:15.05 | twisted[work] | and i need to learn to read down lines |
| 00:15.09 | kpfleming | lol |
| 00:15.10 | xkev | GoSub() would be nice :) |
| 00:15.17 | xkev | but then I guess that's what Macro() is :) |
| 00:15.31 | twisted[work] | xkev, there's patches for gosub() stuffs on the bugtracker |
| 00:15.31 | kpfleming | it's almost the same thing, once you have MacroExit() (same as return) |
| 00:16.03 | xkev | twisted ah. I found macro to be synonymous so what's the purpose? |
| 00:16.23 | twisted[work] | go read the bugtracker, or ask corydon, as it's his patch |
| 00:18.10 | ZX81 | yay I got a 1:30am set for New Years party!!!! |
| 00:18.11 | ZX81 | yayayayaya |
| 00:18.13 | ZX81 | yay |
| 00:18.17 | ZX81 | It's gonna rock |
| 00:18.30 | ZX81 | And I got the highest pay of anyone in New Zealand for my set! |
| 00:18.32 | ZX81 | :-) |
| 00:19.09 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l01v-28-49.d4.club-internet.fr) |
| 00:21.28 | *** join/#asterisk MrEntropy (~entropy@177.004.dsl.sa.iprimus.net.au) |
| 00:21.42 | MrEntropy | yo |
| 00:22.03 | anthm | I have an app_while bug 2907 |
| 00:23.34 | *** join/#asterisk Ya_Ba (~YaBa@adslsapo-b5-161-116.telepac.pt) |
| 00:23.47 | *** join/#asterisk The_Duke (~the_duke@ppp-124-78.adsl.restena.lu) |
| 00:23.51 | *** join/#asterisk Ahewes (~rsb@67.113.12.162) |
| 00:24.01 | Corydon-w | xkev: yeah #2720, with a prereq of 2278 |
| 00:24.09 | The_Duke | Hello |
| 00:24.18 | Ahewes | anyone have good experience with a VoIP paging system? |
| 00:25.03 | The_Duke | how do I know if a toll free number is US48 TF Number or a US50+CA TF Number? |
| 00:25.08 | PoWeRKiLL | ~seen oej |
| 00:25.14 | jbot | oej <~oej@apollo.webway.se> was last seen on IRC in channel #asterisk, 1d 5h 17m 18s ago, saying: 'Corydon-w: I agree, that is a problem'. |
| 00:25.30 | ChulJin | dial it from Canada, Hawaii, and Alaska |
| 00:25.36 | *** join/#asterisk imonlinellc (~bclem@adsl-67-67-70-243.dsl.hstntx.swbell.net) |
| 00:25.38 | imonlinellc | hello |
| 00:25.40 | imonlinellc | quick question |
| 00:25.53 | imonlinellc | I am building a small in-house just to do voip pbx switching |
| 00:26.12 | imonlinellc | ie we are connecting our pbx to a t1 line and will be getting voip termination and origination |
| 00:26.17 | imonlinellc | very small footprint for voice mail |
| 00:26.21 | imonlinellc | I'm about to buy the machine |
| 00:26.24 | MrEntropy | where in kphone do i put the password (asterisk secret="foo")? |
| 00:26.42 | imonlinellc | if my calculations are correct...a dual pIII 800mhz with 1GB ram and 2x60GB ide drives should suffice |
| 00:26.44 | The_Duke | ChilJin: I'm located in LU (Luxembourg) and I don't know ITSPs in CA, Alaska or Hawai.... |
| 00:26.50 | imonlinellc | anybody want to challenge that calculation? |
| 00:27.34 | Ya_Ba | hi [[]]] |
| 00:28.28 | kpfleming | Corydon-w: i like your Gosub patch, it's a cleaner implementation than Macro (doesn't duplicate the extension processing loop from pbx.c), but it's not moving while it has the local variables stuff in it... could you post a version that only adds Gosub/Return/etc and get that part merged first? |
| 00:32.09 | MrEntropy | surely someone here used kphone and sent a password somehow |
| 00:33.23 | imonlinellc | asterisk rocks guy |
| 00:33.24 | imonlinellc | guys |
| 00:33.43 | imonlinellc | it's allowed us to sell products and services to our customers that have helped our bottom line |
| 00:33.50 | imonlinellc | is there a way to donote some money to the project? |
| 00:34.09 | Ya_Ba | i'm sorry.... can anyone tell me so kind of GUI to configure my new asterisk machine?! i'm a newbie, saw some stuff on wiki but some doesn't work the way i want. |
| 00:34.28 | markw | Ya_Ba: xterm or gvim. |
| 00:34.37 | slightly | heh |
| 00:34.39 | crash3m | hehe good answer |
| 00:34.41 | Ya_Ba | Markw :D :D ssh :D |
| 00:34.46 | slightly | harsh |
| 00:34.47 | Ya_Ba | good old command line |
| 00:34.53 | imonlinellc | use command line |
| 00:34.57 | slightly | tsk |
| 00:35.02 | slightly | gvim is for wimps |
| 00:35.05 | markw | xterm is a gui... :) |
| 00:35.07 | slightly | real men use ed |
| 00:35.12 | markw | copy con. :) |
| 00:35.16 | xkev | slightly, cat > |
| 00:35.17 | crash3m | real men use dd |
| 00:35.28 | Ya_Ba | ok, so recomend me a GOOD tutorial about konfiguring the dialplan, extensions and sip phones |
| 00:35.38 | modulus_ | ed > vi > pico |
| 00:35.44 | slightly | theres a good pdf tutorial on the asterisk homepage |
| 00:36.13 | slightly | says slight who's banging his head against a configuration brick wall ;) |
| 00:36.19 | Ya_Ba | this is the scenario: 1 PC, 4 phones, 2 konected directly to the switch, 2 connected via internet... |
| 00:36.47 | xkev | spend about a week staring at extensions.conf and you'll be a pro |
| 00:37.06 | Ya_Ba | xkev :D |
| 00:37.13 | xkev | don't forget -vvvv too :) |
| 00:37.28 | Ya_Ba | that was the way i do it when programming |
| 00:38.01 | Ya_Ba | well |
| 00:38.03 | Ya_Ba | thanks |
| 00:38.13 | *** join/#asterisk Tester2 (~Cyber@modemcable174.150-203-24.mc.videotron.ca) |
| 00:38.14 | Ya_Ba | i'll go read the pdf @ asterisk.org |
| 00:38.28 | Ya_Ba | just one more kuestion.... |
| 00:38.57 | imonlinellc | whats the deal with t he "kuestion" |
| 00:39.03 | imonlinellc | and "konfiguring"? |
| 00:39.18 | Ya_Ba | is it enough a machine P3 with 256bm ram, ADSL 512k/128k connection to handle at least 5 sip phones ?! |
| 00:39.21 | *** join/#asterisk Umaro (~umaro@c-24-22-76-14.client.comcast.net) |
| 00:39.31 | Ya_Ba | 2 in the same LAN, the others via internet |
| 00:39.40 | Ya_Ba | sorry my bad english :P I'm Portuguese |
| 00:39.46 | Umaro | Hey guys, having a small problem with IAX2.. getting one-way audio |
| 00:40.56 | Godsey | Ya_Ba I think 5 phones on a p3 is easily doable |
| 00:41.01 | nic_oz | what does the panalty mean when you define a member of a queue ? |
| 00:41.21 | Godsey | I have a p4 2.8 w/ 512meg and have had 12 sip calls at once w/ 0.00 load average |
| 00:41.24 | markw | Ya_Ba: all real phone systems use cli. |
| 00:42.18 | Slimey | yay! |
| 00:42.23 | Slimey | Solaris patch applied to CVS |
| 00:42.32 | *** join/#asterisk juice (~juice@mo-205-240-40-86.dyn.sprint-hsd.net) |
| 00:42.42 | markw | Slimey: asterisk on solaris? |
| 00:42.56 | Slimey | yup |
| 00:42.59 | markw | Slimey: you know the guys that developed that digium/zapata card dumped all the solaris boxes. |
| 00:43.06 | Strom_TM | i thought the whole point was to be able to use commodity intel stuff |
| 00:43.13 | Strom_TM | oh wait, i forgot about solaris x86 |
| 00:43.20 | Tester2 | x-lite -> router -> internet -> router -> asterisk -> x100p -> telephone line => is this config suppose to be stable? because everytime I try it the connection (telephone) drop after a random time (+/- 5 minutes) any idea why? By the way nothing special in /var/log/asterisk/messages and on the asterisk term it is written " == Spawn extension (sip, 0XXXXXXXX, 2) exited non-zero on 'SIP/COP-4c31'" |
| 00:43.26 | Slimey | This is solaris on sparc |
| 00:43.44 | Slimey | The nice thing about asterisk on solaris is that sun make lots of NEBS compliant boxes, so telcos like them |
| 00:44.08 | markw | Slimey: toddtel, aka nccom, aka jartel, aka half the guys listed on the zapata page, used to be solaris snobs until they saw the commodity hardware light. |
| 00:44.10 | Corydon-w | kpfleming: You misunderstand why 2720 isn't moving. Yes, it has a prerequisite, but the real issue is that Mark disagrees with me on implementing subroutines as Gosub/Return |
| 00:44.24 | Slimey | markw: heh, full circle time, then |
| 00:44.30 | kpfleming | ahh too bad... i like the way you implemented them |
| 00:44.36 | Slimey | I'm working on getting the zaptel stuff working on solaris |
| 00:45.04 | Corydon-w | Until Mark has a veritable army of people saying "We want this more than Macro", 2720 isn't going anywhere |
| 00:45.10 | Slimey | dev.slimey.org-root-20: modinfo | grep -i zap |
| 00:45.10 | Slimey | <PROTECTED> |
| 00:45.10 | Slimey | <PROTECTED> |
| 00:45.22 | Slimey | that much works on solaris ;) |
| 00:45.30 | kpfleming | well, i'll add my voice, but i'm also posting a tiny patch to add MacroExit so that macros can return early :-( |
| 00:45.40 | Strom_TM | do sparcs use PCI interfaces now? the last time i stuck my nose in a Sun box it was all sbus |
| 00:45.54 | Slimey | it's all PCI now |
| 00:46.05 | *** join/#asterisk Legend (~legend@24.244.142.134) |
| 00:46.12 | Strom_TM | ah ok |
| 00:46.14 | Slimey | The move was about 5 years ago? :) |
| 00:46.24 | Slimey | I still have sbus boxes lying around, tho |
| 00:46.33 | Strom_TM | Slimey, the sun boxes I have are five years old :) |
| 00:46.39 | Strom_TM | probably more |
| 00:47.02 | Godsey | I just took a bunch of old systems to the dump :) |
| 00:47.04 | Slimey | Ultra 1 was sbus, Ultra 2 was sbus, stuff after that was pci |
| 00:47.06 | Godsey | IPC and Sparc2 |
| 00:47.24 | Godsey | 730ish pounds :) |
| 00:47.44 | *** join/#asterisk elkcsr (elkcsr@CPE-24-208-95-252.neb.rr.com) |
| 00:47.59 | Strom_TM | the Ultra Enterprise 2s I have under one of my desks used to be evite.com :) |
| 00:48.32 | Slimey | I think some of the enterprise xx00's you got a choice of PCI or sbus per I/O module |
| 00:49.45 | *** join/#asterisk kimosabe (~kimosabe@201.129.75.111) |
| 00:50.23 | Tester2 | no one has an answer for my qestion? |
| 00:51.03 | kimosabe | question when u run asterisk -vvvgc does it tend to stop runnin after a few days |
| 00:51.18 | markw | Slimey: Todd was so impressed with kevin's card (zapata) that he dumped all his solaris boxes and runs everything on linux now. |
| 00:51.47 | markw | Slimey: he's a big CLEC btw. |
| 00:51.51 | Umaro | If I call a PSTN number and connect it to a local extension like the echo app, it works great. I can hear echo just fine. But if I try to get my firefly working, only the person I called can hear me, I can't hear them. |
| 00:52.49 | Umaro | calling from firefly to a local extension, such as echo, works fine also. |
| 00:53.34 | Slimey | markw: I still live in the world where solaris boxes are nice ;) |
| 00:54.03 | markw | Slimey: he's looking at craming as much minutes into a box as he can. |
| 00:54.11 | markw | Slimey: 1 box == 250000 minutes a day. |
| 00:54.30 | markw | Slimey: and at a fraction of a cent per minute that he bills for terminating the call, that starts to add up to real money. |
| 00:54.48 | *** join/#asterisk ClayReiche1 (fwuser@acxexch1.accxx.com) |
| 00:54.50 | Slimey | nice |
| 00:55.15 | agronqui | anybody know if it is possible to have * decide which codec to used based on matching the one that it is going to be connecting to, so that it doesn't have to do conversion?? e.g. I have a UA that does 711/729 or sometimes only one or the other and I want * to pick the codec for the other side of the connection based on what the UA advertises in it's SDP OK message..... |
| 00:55.19 | *** join/#asterisk wsmith (~wsmith@adsl-68-73-230-240.dsl.chcgil.ameritech.net) |
| 00:55.26 | Slimey | where it provides a nice simple PBX with call routing for my ISP |
| 00:55.51 | wsmith | do all digium/digum-look-alikes cards have the same pci manufacturer ids? |
| 00:56.18 | markw | wsmith: the digium card is based off Kevin Dixon's design at zapatatelephony.org |
| 00:56.27 | markw | wsmith: if you want to build one you can too. |
| 00:57.55 | wsmith | i'm just trying to figure out a way to script zaptel configuration, and am looking for an easy way out. any ideas? |
| 00:58.25 | markw | wsmith: no, haven't had to do that part. :) |
| 00:58.59 | agronqui | like, if I have 711 and 729 allowed on all interfaces, but 729 higher priority, and a UA advertises it only does 711, * tries to do the 729 on one side and fails to find a converter, why can't it just pick 711 on both sides then?? |
| 01:00.49 | implicit | markw: worry more about phone guys doing computer systems |
| 01:01.08 | markw | implicit: seeing how I was phone before computer, I don't worry too much. :) |
| 01:01.50 | implicit | markw: a lot of the guys were into telephony before voip as well as computers |
| 01:02.34 | *** join/#asterisk Brixius (Brixius@c-24-118-4-197.mn.client2.attbi.com) |
| 01:02.36 | ClayReiche1 | Anyone have any experience with the RealTime features? |
| 01:02.44 | markw | implicit: it's amazing what these guys are doing. |
| 01:03.10 | markw | implicit: basically a bunch of old school 2600 guys. |
| 01:03.32 | markw | implicit: now they're CLEC's and legally screwing the phone company. :) |
| 01:03.46 | implicit | markw: eh, except for most of them had no idea what they were doing in terms of phreaking |
| 01:03.50 | Brixius | Good evening to all you fine astmasters (er asterisk masters) here. HAHA |
| 01:03.56 | markw | implicit: not these guys. |
| 01:03.57 | kimosabe | is there a reason why my asterisk stops running all the suden it just exits |
| 01:04.07 | wsmith | great. like i don't get enough astmaster jokes at work... |
| 01:04.07 | implicit | markw: i know what i am talking about |
| 01:04.17 | markw | implicit: where you located? |
| 01:04.33 | implicit | california |
| 01:05.27 | Moc | hi again |
| 01:05.32 | implicit | hi moc |
| 01:05.52 | wsmith | kimosabe, plenty. are you running asterisk in the background? |
| 01:07.50 | ClayReiche1 | Can anyone help me out with a realtime problem? |
| 01:08.24 | wsmith | maybe later |
| 01:08.31 | wsmith | ; ) |
| 01:09.04 | kimosabe | wsmith im runni asterisk -vvvgc |
| 01:10.05 | kimosabe | wsmith its my first asterisk solution and it never shuts down but on mondays they call im down and there i go service is stoped |
| 01:10.34 | wsmith | every monday? |
| 01:11.42 | kimosabe | yes |
| 01:11.58 | kimosabe | place is closed sat and sunday |
| 01:12.46 | wsmith | do you have the dialplan (extensions.conf) play a message during the non-business hours? |
| 01:12.50 | *** join/#asterisk ctooley ([U2FsdGVkX@199.89.146.18) |
| 01:14.19 | wsmith | the easiest way to fix this problem (without finding a cure) is to use safe_asterisk. it constantly restarts asterisk if it ever dies. |
| 01:17.01 | kimosabe | wsmith how doi do that |
| 01:17.45 | kimosabe | wsmith i have interoffice link fxs fxs into pbx for interconecting offices over seas or over border |
| 01:20.40 | *** join/#asterisk RestLessGemini (~RestLessG@202.142.189.86) |
| 01:22.39 | ds2 | would anyone know of a good summary of teh difference between MGCP and SGCP? |
| 01:23.25 | Chotaire | teh diff! |
| 01:23.50 | _Vile | teh! |
| 01:24.07 | ClayReiche1 | I’m having trouble with the Realtime setup. I’ve followed the instructions on voip-info using odbc but I get this message during asterisk boot: |
| 01:24.07 | ClayReiche1 | [chan_sip.so] => (Session Initiation Protocol (SIP)) |
| 01:24.07 | ClayReiche1 | <PROTECTED> |
| 01:24.07 | ClayReiche1 | Dec 14 19:55:11 NOTICE[9642]: chan_sip.c:8462 reload_config: Unable to load config sip.conf, SIP disabled |
| 01:24.07 | ClayReiche1 | <PROTECTED> |
| 01:24.09 | ClayReiche1 | <PROTECTED> |
| 01:24.11 | ClayReiche1 | <PROTECTED> |
| 01:24.39 | RestLessGemini | check if you have /etc/asterisk/sip.conf |
| 01:25.01 | ClayReiche1 | I removed it intentionally. |
| 01:25.14 | ClayReiche1 | for realtime. |
| 01:25.15 | RestLessGemini | Unable to load config sip.conf, SIP disabled |
| 01:25.19 | ClayReiche1 | do I need it? |
| 01:25.22 | RestLessGemini | <PROTECTED> |
| 01:25.25 | RestLessGemini | dont remove it |
| 01:25.26 | RestLessGemini | keep it there |
| 01:25.34 | RestLessGemini | and then test |
| 01:25.51 | ClayReiche1 | 10-4 thanks |
| 01:25.51 | wsmith | kimosabe, instead of asterisk -vvvvc, you use safe_asterisk. i think the script is in the source directory (anybody?) |
| 01:26.35 | kimosabe | thanksman |
| 01:27.12 | *** join/#asterisk Johaan (~johaan@ip24-56-24-181.ph.ph.cox.net) |
| 01:27.15 | ClayReiche1 | Kimo: /usr/sbin/safe_asterisk |
| 01:27.22 | kimosabe | thanks man |
| 01:30.25 | Godsey | is it musicclass=blah in sip.conf to set music on hold? |
| 01:31.21 | Godsey | I must be doing it wrong :) |
| 01:33.57 | ClayReiche1 | RestLessGemini: Now I'm getting registraion failures. |
| 01:34.02 | ClayReiche1 | Dec 14 20:29:26 NOTICE[9722]: chan_sip.c:7531 handle_request: Registration from '5555 <sip:5555@65.77.37.186:5060>' failed for '65.77.37.2' |
| 01:34.02 | ClayReiche1 | Dec 14 20:29:26 NOTICE[9722]: chan_sip.c:7531 handle_request: Registration from '5556 <sip:5556@65.77.37.186:5060>' failed for '65.77.37.2' |
| 01:34.46 | ClayReiche1 | heheh sorry! |
| 01:35.08 | ZX81 | lol |
| 01:35.22 | ClayReiche1 | That was for the... "I'll help you later with your realtime problem" crack.... :) |
| 01:35.28 | ZX81 | jradford: :-) new releases are always newsworthy |
| 01:35.30 | RestLessGemini | Clay: Actually i've not setup realtime yet, check your db for sip stuff |
| 01:35.38 | wsmith | lol. |
| 01:35.44 | RestLessGemini | Godsey : read http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20SetMusicOnHold |
| 01:36.09 | Godsey | I had to restart asterisk and it worked |
| 01:36.11 | Godsey | sip reload didn't work |
| 01:36.26 | _Vile | god damnit I hate Microsoft sometimes |
| 01:36.27 | Godsey | that way is easy :) |
| 01:36.32 | Godsey | I was working on setting it for outbound calls |
| 01:36.39 | *** join/#asterisk gushi (danm@prime.gushi.org) |
| 01:36.52 | gushi | hey guys, what's the zaptel tool that will tell me the status of my PRI? |
| 01:37.09 | gushi | i.e. that will show alarms and stuff? |
| 01:37.54 | RestLessGemini | there is i guess one tool only, zttool |
| 01:44.50 | Godsey | I admin an isp |
| 01:45.00 | Godsey | a women came in wanting to change some details on their account |
| 01:45.04 | Godsey | who the husband setup |
| 01:45.08 | Godsey | er which |
| 01:45.17 | Godsey | and we're really anal like a cell phone company :) |
| 01:45.38 | Godsey | and she got pissed and yelled I DONT LIVE WITH MY HUSBAND I JUST WANT TO READ EMAIL IN OUR ACCOUNT |
| 01:45.41 | Godsey | haha :) |
| 01:45.47 | RestLessGemini | :) |
| 01:46.07 | Godsey | working with the public < * |
| 01:46.16 | wsmith | Godsey, hey i work at a cell phone company and were are absouletly anal-less (sic?) |
| 01:47.25 | Moc | ok... time to work on some video editing.. |
| 01:47.30 | Moc | bbl |
| 01:48.11 | RestLessGemini | ok me out for the day |
| 01:48.13 | RestLessGemini | take care all |
| 01:48.16 | RestLessGemini | bbye for today |
| 01:48.22 | *** part/#asterisk RestLessGemini (~RestLessG@202.142.189.86) |
| 01:53.22 | nestAr | Godsey: technically, if they are married, she has just as much right to change something as he does.. remember, in a court, her name is Mrs. His Name |
| 01:53.24 | nestAr | ;) |