| 00:00.02 | zazzizza | :-( |
| 00:00.03 | terinjokes | does anyone know and provider that can get me a virtual phone number in australia? |
| 00:00.09 | OneBinary | ok |
| 00:00.10 | b11d|bbl | yeap |
| 00:00.14 | Ryanw | OneBinary, try sip debug ip 10.0.0.24 |
| 00:01.01 | Ryanw | terrinjokes, what do you mean by virtual phone number. VoIP only or dialable from Telstra too? |
| 00:01.07 | zazzizza | let's see. if you do csim start 7309 |
| 00:01.16 | zazzizza | does the phone ring? |
| 00:02.02 | *** join/#asterisk Marshall16 (n=Marshall@cpe-76-181-167-184.columbus.res.rr.com) |
| 00:03.28 | b11d|bbl | from where? |
| 00:03.30 | b11d|bbl | i'll try |
| 00:03.32 | zazzizza | from the cisco |
| 00:03.45 | b11d|bbl | i dont have that command |
| 00:03.47 | zazzizza | uh 'if' the vg supports csim |
| 00:03.47 | *** join/#asterisk TheCops (n=henri@206-248-136-187.dsl.teksavvy.com) |
| 00:03.50 | zazzizza | (n) |
| 00:04.09 | OneBinary | Ryanw, Phone has rebooted, and it is spitting out a lot of debug info. What am I looking for? |
| 00:04.36 | zazzizza | if you have asterisk 1.2+ i think you can do a dial from the console |
| 00:05.07 | b11d|bbl | yeah.. dialing 7309 from the console works |
| 00:05.20 | zazzizza | and dialing 5454? |
| 00:05.29 | Ryanw | OneBinary, just like HTML sip has response codes, 200 for OK, 404 not found, 302 permission denied |
| 00:05.42 | *** join/#asterisk h3x0r (n=hex@ip68-224-236-92.lv.lv.cox.net) |
| 00:05.43 | Ryanw | before the reboot you might've got an insight into why it was returning that error code |
| 00:06.02 | b11d|bbl | works |
| 00:06.02 | Ryanw | did the reboot fix it? |
| 00:06.02 | b11d|bbl | it rings through ok |
| 00:06.06 | zazzizza | i think it's more a config issue than a technical fault. if you call the TAC they scream at you |
| 00:06.12 | OneBinary | Ryanw, reboot did not fix it |
| 00:06.13 | zazzizza | ok |
| 00:06.25 | b11d|bbl | i know |
| 00:06.31 | b11d|bbl | i really dont want to call the TAC.. but i will.. |
| 00:06.34 | zazzizza | it's working man-in-the-middle sort of! |
| 00:06.35 | Ryanw | OneBinary, does sip show peers list it as unreachable now ? |
| 00:06.54 | OneBinary | Yes |
| 00:07.05 | zazzizza | now from the 7309 if you dial 5454 you get a fast busy? |
| 00:07.15 | b11d|bbl | yes |
| 00:07.37 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 00:08.00 | b11d|bbl | got 503 "Service Busy" from 10.0.5.5 (the vg) |
| 00:08.20 | Ryanw | OneBinary, assuming that 365 is on the same ip address after the reboot, try placing a call to 365 and see what the sip response says, if there is one at all. |
| 00:08.36 | *** join/#asterisk BZBW (n=wlwzhang@ip67-153-142-110.z142-153-67.customer.algx.net) |
| 00:08.58 | zazzizza | can you pastebin again? the post expired |
| 00:09.00 | BZBW | anyone knows 1.4 support SIP over TCP or not? |
| 00:09.15 | zazzizza | and i wanted to peek the runningconfig |
| 00:09.25 | b11d|bbl | ok.. you got it |
| 00:09.26 | b11d|bbl | give me a sec |
| 00:09.34 | b11d|bbl | i'll copy my latest.. |
| 00:09.35 | zazzizza | ok |
| 00:11.59 | b11d|bbl | http://pastebin.ca/235084 |
| 00:12.31 | *** part/#asterisk terinjokes (n=spader@adsl-11-150-123.mia.bellsouth.net) |
| 00:12.31 | BZBW | emm, anyone had issue on building 1.4, I kept getting errors, and my previous version is 1.2.10 |
| 00:12.59 | b11d|bbl | well, yeah. some have.. its beta.. |
| 00:13.48 | BZBW | it's beta3 already and I have yet to build it correctly:(. |
| 00:14.04 | b11d|bbl | have you reported your build errors? |
| 00:14.22 | BZBW | where should I report to? |
| 00:14.29 | b11d|bbl | bugs.asterisk.org ? |
| 00:14.30 | b11d|bbl | i think |
| 00:14.41 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 00:15.01 | BZBW | I'm just wondering if it's just some environment viarable that I didn't set correctly. |
| 00:15.27 | b11d|bbl | its possible.. |
| 00:16.38 | zazzizza | it should work |
| 00:16.43 | b11d|bbl | tell me about it :) |
| 00:16.45 | zazzizza | i dont know what else to look |
| 00:18.37 | b11d|bbl | well.. thanks just the same for taking the time |
| 00:18.45 | b11d|bbl | its appreciated |
| 00:19.13 | zazzizza | no problem |
| 00:19.21 | zazzizza | :-( |
| 00:19.36 | b11d|bbl | i'll likely have to call the TAC.. |
| 00:19.47 | b11d|bbl | sigh |
| 00:19.51 | zazzizza | hah |
| 00:20.09 | zazzizza | it has to be something *really* silly |
| 00:20.15 | b11d|bbl | yeah no doubt about it.. |
| 00:20.24 | b11d|bbl | i've been hoping it was something silly for the last 3 days ;) |
| 00:20.36 | zazzizza | oh |
| 00:20.47 | zazzizza | ok |
| 00:20.57 | zazzizza | is it something secret? |
| 00:21.02 | b11d|bbl | ? |
| 00:21.06 | b11d|bbl | what? |
| 00:21.07 | zazzizza | do you mind taking a look at sip.conf? |
| 00:21.12 | b11d|bbl | not at all |
| 00:21.22 | zazzizza | i dont know, maybe you work for the cia |
| 00:21.25 | b11d|bbl | you want to see my sip.conf ? |
| 00:21.34 | b11d|bbl | haha yeah i work for the CIA and im looking for help here ;) |
| 00:21.45 | zazzizza | yep and extensions.conf |
| 00:21.51 | b11d|bbl | ok.. give me a second. |
| 00:22.00 | zazzizza | so we sort it out |
| 00:22.11 | *** join/#asterisk saabo (n=saab_va@user-0c8hjld.cable.mindspring.com) |
| 00:24.03 | b11d|bbl | http://pastebin.ca/235095 |
| 00:24.06 | b11d|bbl | is extensions.conf |
| 00:24.17 | TheCops | Someone is using Polycom IP601 phone and headset? I have some difficulties with the volume and need some help. thanks |
| 00:24.29 | b11d|bbl | http://pastebin.ca/235096 |
| 00:24.31 | b11d|bbl | is sip.conf |
| 00:24.49 | b11d|bbl | i've XXXXX'd the secrets |
| 00:25.25 | zazzizza | no problem! |
| 00:30.33 | zazzizza | canreinvite = yes |
| 00:30.53 | zazzizza | and in [5454] |
| 00:31.07 | zazzizza | host=10.0.5.5 |
| 00:31.13 | b11d|bbl | canreinvite where? |
| 00:31.18 | b11d|bbl | in general? |
| 00:31.30 | zazzizza | at least in those two |
| 00:31.59 | zazzizza | i always use canreinvite=yes, because otherwise * keeps in the middle all the time |
| 00:32.01 | [hC] | so, * box with direct PRI, it is suggested to use hylafax instead of rxfax? (rxfax seems to not like 3+ page faxes) |
| 00:32.21 | b11d|bbl | fast busy.. one sec |
| 00:32.57 | zazzizza | reload both chan_sip.so |
| 00:33.01 | zazzizza | and pbx_config.so |
| 00:33.23 | zazzizza | sorry no changes made to pbx_config.so so far |
| 00:33.24 | b11d|bbl | nope |
| 00:33.27 | b11d|bbl | 400 bad request |
| 00:33.32 | b11d|bbl | Malformed / Missing URL |
| 00:33.39 | b11d|bbl | back from 10.0.5.5 |
| 00:34.05 | zazzizza | ok |
| 00:34.08 | zazzizza | in extensions.conf |
| 00:34.11 | zazzizza | change this |
| 00:34.17 | zazzizza | and this is my final shot |
| 00:34.37 | zazzizza | exten => 5454,1,Dial(SIP/5454) -> exten => 5454,1,Dial(SIP/5454/5454) |
| 00:34.43 | b11d|bbl | ok |
| 00:34.54 | zazzizza | you have to reload pbx_config.so |
| 00:35.27 | b11d|bbl | well, it worked.. but still.. one-way audio. |
| 00:35.38 | zazzizza | where? |
| 00:36.15 | b11d|bbl | the analog can not send audio back.. |
| 00:36.19 | b11d|bbl | the sip works normally.. |
| 00:36.41 | zazzizza | the analog sits on the vg? |
| 00:36.43 | b11d|bbl | yes |
| 00:37.09 | b11d|bbl | calling the sip from the analog, or calling the analog from the sip, both result in the exact same issue.. |
| 00:37.10 | zazzizza | what ip phone is it? |
| 00:37.13 | b11d|bbl | Polycom 501 |
| 00:37.15 | zamba | is g711 enabled in asterisk? |
| 00:37.26 | zazzizza | yes it is |
| 00:37.27 | b11d|bbl | it negotiates to g711 so yeah.. |
| 00:38.13 | zamba | Nov 3 01:38:04 WARNING[7630]: frame.c:1030 ast_parse_allow_disallow: Cannot allow unknown format 'g711' |
| 00:38.37 | b11d|bbl | g711 is ulaw.. maybe set that? |
| 00:38.37 | zazzizza | issue show codecs |
| 00:38.55 | zamba | ulaw, yeah.. that's the one |
| 00:39.03 | JT | g711 is ulaw or alaw |
| 00:39.11 | b11d|bbl | so i notice.. |
| 00:39.43 | zazzizza | ok, do a test just for yourself :S |
| 00:39.44 | *** join/#asterisk Strom_C (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 00:39.56 | zazzizza | it does not involve hammers so dont worry |
| 00:39.59 | b11d|bbl | :) |
| 00:40.11 | zazzizza | since the ip phone can register *anywhere* |
| 00:40.38 | zazzizza | put it on the 10.0.5.0/24 |
| 00:40.48 | zazzizza | that's the mother of all tests |
| 00:40.56 | zazzizza | if it *does* work |
| 00:41.01 | b11d|bbl | haha.. |
| 00:41.05 | zazzizza | then there's some routing issue |
| 00:41.08 | zazzizza | if not... |
| 00:41.17 | zazzizza | there's always a hammer... |
| 00:41.19 | zazzizza | :-D |
| 00:41.22 | b11d|bbl | :) |
| 00:41.25 | b11d|bbl | no doubt about it. |
| 00:41.26 | zazzizza | or 2! |
| 00:41.48 | b11d|bbl | ok.. im actually going to do that tomorrow morning, because it would take me like 30 mins to do that.. |
| 00:41.51 | b11d|bbl | i can assure you |
| 00:42.04 | b11d|bbl | and i've been ready to go home for the last 3 hours :) |
| 00:42.18 | zazzizza | ok, but make sure you do it! |
| 00:42.24 | b11d|bbl | its on my whiteboard.. first thing in the morning.. |
| 00:42.34 | zazzizza | sip is really shitty with that |
| 00:42.49 | zazzizza | i tell you i had a lot of shit coming from cisco and * |
| 00:42.56 | b11d|bbl | actually, i have a voip phone on that switch, but it's a snom.. |
| 00:43.08 | zazzizza | and i managed yesterday to go around something like *this* |
| 00:43.09 | zazzizza | :-S |
| 00:43.18 | b11d|bbl | :) |
| 00:43.22 | zazzizza | ok man |
| 00:43.32 | zazzizza | let's talk tomorrow |
| 00:43.35 | b11d|bbl | sounds good. |
| 00:43.38 | b11d|bbl | thanks again for everything. |
| 00:43.42 | zazzizza | i have an appointment at the pizza place :-D |
| 00:43.45 | b11d|bbl | :) |
| 00:43.48 | zazzizza | no problem, cheers! |
| 00:43.55 | b11d|bbl | goodnight everyone |
| 00:44.05 | zazzizza | goodnite! |
| 00:45.42 | *** join/#asterisk litage (n=nick@203.220.55.70) |
| 00:46.53 | JT | zamba: where is the asterisk system you're building, what country? |
| 00:50.50 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 00:51.14 | *** join/#asterisk nick125_lappy (n=nick@atarack/staff/nick125) |
| 00:51.58 | nick125_lappy | anyone here off hand know of a link that describes how to make a "guest" SIP account that anyone can connect to (I have a conference system and I want people to be able to call in though SIP)? |
| 00:52.44 | *** join/#asterisk conico (n=chatzill@88.224.252.153) |
| 00:53.08 | Ryanw | Nick...just set the context at the top of sip.conf all unauthenticated calls get routed there |
| 00:53.36 | nick125_lappy | fun.. |
| 00:54.18 | conico | hello i need physical layer info.if i want to replace my pbx with asterisk which pci cards do i need if i have pri telephone line to the company? |
| 00:55.06 | JT | T1? |
| 00:55.19 | conico | isdn pri |
| 00:55.27 | JT | yes |
| 00:55.31 | JT | now is it a T1? |
| 00:55.39 | conico | i only know this info about it |
| 00:55.42 | conico | say it t1 |
| 00:56.15 | JT | what country are you in? |
| 00:56.22 | conico | türkiye |
| 00:56.30 | Ryanw | T1 = US 24 channel ISDN, E1 = European / Australian 30 channel ISDN |
| 00:56.31 | JT | turkey? |
| 00:56.31 | zamba | JT: norway |
| 00:56.37 | zamba | JT: why do you ask? |
| 00:56.42 | conico | yes turkey |
| 00:56.57 | JT | zamba: the information provided to you to use ulaw was wrong, use Alaw |
| 00:57.19 | JT | heaps of people here say ulaw like USA is the only country in the world |
| 00:57.21 | *** join/#asterisk Winkie (n=urmom@gateway.duclicsic.com) |
| 00:57.32 | JT | the international standard is A-law |
| 00:57.33 | zamba | JT: ah, what's the difference? |
| 00:57.43 | JT | zamba: just a different companding table |
| 00:57.58 | JT | same bandwidth otherwise |
| 00:58.22 | JT | conico: i'm not sure if Turkey uses T1 PRIs or E1 |
| 00:58.46 | conico | it should be t1 |
| 00:58.54 | conico | t1 pri |
| 00:58.54 | JT | are you sure? |
| 00:59.03 | JT | before you didn't know |
| 00:59.11 | conico | we talk about t1 lines everytime here |
| 00:59.16 | JT | hrm |
| 00:59.24 | conico | i never heard a e1 in turkiye |
| 00:59.37 | JT | well, a TE110P is the most basic card you can get that'll do both anyway |
| 00:59.44 | JT | you can get cards with more channels |
| 00:59.48 | JT | and echo cancellers |
| 01:00.13 | conico | more channels means more pri lines? |
| 01:00.26 | JT | yes |
| 01:01.01 | conico | ok now i bought a pci card and put the pri line in this then? |
| 01:01.23 | conico | is this all to replace a pbx? |
| 01:01.29 | JT | yes, that's it |
| 01:01.33 | conico | omg |
| 01:01.34 | JT | well if you're replacing the pabx |
| 01:01.40 | JT | you'll need phone extensions |
| 01:01.44 | conico | people will loose their jobs |
| 01:01.45 | JT | like SIP phones |
| 01:01.57 | JT | what people will lose their jobs? |
| 01:02.06 | conico | pbx people |
| 01:02.31 | conico | i do phone extensions in asterisk ? |
| 01:02.59 | conico | then put ip phones in the same ethernet segment with asteriks box? |
| 01:03.40 | EyeCue | hmm, is there any concept of PTT in asterisk, or any of its plugins ? |
| 01:04.12 | Ryanw | conico if you have no idea, checkout freepbx.org its a relatively simple intro into asterisk |
| 01:04.55 | conico | i have idea buut i want to be sure i have no pri card to try |
| 01:05.12 | conico | before buying i should be ok |
| 01:07.25 | conico | now i put t1 pri card in asteriks box put t1 pri telephone twisted pair to this card and make extensions in asterisk and put this asterisk box to ethernet and in ethernet ip phones.is that all ? |
| 01:07.51 | *** join/#asterisk slayer192 (n=slayer19@adsl-70-137-24-211.dsl.okcyok.swbell.net) |
| 01:10.17 | conico | if i want to use my existing pbx still what should i ?because we may use old phones still? |
| 01:10.43 | conico | another pci card for the connection the pbx? |
| 01:10.47 | *** join/#asterisk Winkie (n=urmom@cpc2-stre3-0-0-cust344.bagu.cable.ntl.com) |
| 01:10.48 | conico | to the pbx? |
| 01:11.09 | *** join/#asterisk NDT (n=noone@cpe-74-70-211-81.nycap.res.rr.com) |
| 01:12.39 | Ryanw | conico, depends how your current pbx is connected up atm. |
| 01:13.01 | Ryanw | conico, if it has a T1 then you can get 2 T1 interfaces for asterisk and make yourself a T1 crossover cable |
| 01:13.08 | TheCops | Someone using microbrowser of IP601? What about the load of asterisk with 15 phone making API request of Asterisk each 5 seconde^ |
| 01:13.33 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 01:13.39 | Qwell | TheCops: considering that rtp has a packet every 20ms or so... 5 seconds isn't gonna do squat |
| 01:14.10 | TheCops | Qwell, hehe |
| 01:14.32 | russellb | depending on what you request asterisk to do |
| 01:14.43 | TheCops | this is a park call list request |
| 01:14.43 | ManxPower | TheCops: there has in the past been issues with the manager interface handleing many connections at once, there is a manager proxy you can use to deal with this |
| 01:14.43 | russellb | if you say ... calculate PI to 120497u2039458102934 decimal places |
| 01:14.46 | russellb | then, you know ... |
| 01:14.56 | russellb | ManxPower: those issues should all be fixed now |
| 01:14.58 | TheCops | ManxPower, ho, nice |
| 01:15.13 | TheCops | ManxPower where can I find it ^ |
| 01:15.17 | *** join/#asterisk tengulre (n=tengulre@221.11.5.182) |
| 01:15.20 | russellb | but of course, some people still prefer the proxy |
| 01:18.02 | saabo | now do i get out of sip_nat.comf? |
| 01:18.13 | conico | Ryanw: from pbx to asterisk one t1 interface and from asterisk to PSTN pri one t1 interface?can these 2 t1 interface be on one pci card?should i do configuration on pbx ? |
| 01:20.21 | BZBW | hi, anyone knows 1.4 support SIP over TCP? |
| 01:22.54 | russellb | no, it does not |
| 01:23.26 | C6Vette | Does 1.4 support queue log in mysql? |
| 01:24.29 | ManxPower | TheCops: there was a talk about the proxy in the May 2005 Madrid Astricon |
| 01:24.42 | TheCops | ManxPower, I've got it, pretty nice |
| 01:24.44 | TheCops | thank you very much |
| 01:24.50 | ManxPower | TheCops: no problrm |
| 01:25.05 | TheCops | When for the Astricon in Quebec! |
| 01:28.45 | *** join/#asterisk ApEtc (i=apetc@ip70-162-197-214.ph.ph.cox.net) |
| 01:29.09 | *** join/#asterisk SwK (n=Silik0nJ@208-44-30-242.dia.static.qwest.net) |
| 01:29.45 | EyeCue | is there a protocol over which PTT works best? |
| 01:29.50 | reza_ | anyone know where i can hire a consultant to fine-tuen my asterisk setup? |
| 01:29.55 | EyeCue | or that is specific to PTT ? |
| 01:31.17 | BZBW | russellb: thx. |
| 01:31.49 | BZBW | anyone has issue on building 1.4 beta3? I just can't build it |
| 01:34.28 | JT | EyeCue: what devices are you looking to have PTT functionality between? |
| 01:34.32 | *** join/#asterisk carrar (i=tim@osburn.com) |
| 01:34.43 | JT | conico: that setup should work |
| 01:35.28 | conico | ok thank you so much |
| 01:35.32 | EyeCue | well, im just researching into ptt itself |
| 01:35.36 | EyeCue | i understand its a half-duplex technology |
| 01:35.44 | *** join/#asterisk haryv (i=keefejoh@gateway/web/cgi-irc/ircatwork.com/x-a36a7f81208633f2) |
| 01:35.53 | EyeCue | but the question on protocol is i suppose aimed around findind out open source implementations of PTT |
| 01:36.09 | EyeCue | obviously the client/server is only trasmitting packets when people ptt. |
| 01:36.37 | JT | between what devices though? :) |
| 01:37.24 | EyeCue | say in the teamspeak/ventrilo sense |
| 01:37.34 | JT | ah ok |
| 01:37.54 | EyeCue | i know ts/vent both use proprietary type server network thingies |
| 01:37.57 | tengulre | JT: nice to meet u! ;) |
| 01:38.01 | EyeCue | ventrilo uses tcp infact |
| 01:38.05 | EyeCue | which is strange :~) |
| 01:38.17 | JT | because i've noticed in the options for the Xten X-pro softphone some mention of PTT |
| 01:38.21 | *** join/#asterisk heison (n=heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 01:38.27 | JT | spose tcp works if you don't care about lag :P |
| 01:38.41 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 01:38.49 | JT | and there are implementations of PTT technology for asterisk, but not quite the same focus |
| 01:38.53 | tengulre | it is AM 9:38 here, I'm in office! and you? |
| 01:39.07 | JT | 1240pm here, in the office |
| 01:39.15 | tengulre | anybody know which IM tools can supported by asterisk? |
| 01:39.27 | haryv | sms |
| 01:39.43 | reza_ | fuck, traffic just stopped on the bridge. it's going to take hours getting home now. |
| 01:40.16 | JT | reza_: where's that? |
| 01:40.36 | haryv | In the morning it takes me 30 min just to drive 1/2 mile to the bridge which is about a mile from here. most of it is waiting time. |
| 01:40.36 | reza_ | san francisco bay bridge |
| 01:41.15 | JT | and i thought traffic was bad here... |
| 01:41.32 | haryv | I think cities need to have more control of out of control realestate cost so people should not have to move to the suburbs to afford a house. |
| 01:41.35 | *** join/#asterisk tengulre11 (n=tengulre@221.11.5.182) |
| 01:41.52 | JT | i think the planet needs more population control |
| 01:41.56 | JT | or at least education |
| 01:42.12 | haryv | Whistler/blackcomb did this and built a small residential area for its workers. |
| 01:42.23 | haryv | jt, and correct city planning. |
| 01:42.33 | haryv | poor planning results in long commute times. |
| 01:42.55 | carrar | JT, what do you have in mind? |
| 01:43.25 | JT | 6billion+ people is far too many as is |
| 01:43.33 | JT | people should reduce birth rates :) |
| 01:43.35 | haryv | Population control is becomming more of a issue as time goes on. |
| 01:43.40 | heison | hello everyone... |
| 01:44.16 | haryv | china is now in the process of buying 10 or more super tankers to keep up its thirst for more fuel. |
| 01:44.36 | heison | has anyone here configured Nortel Option 11C (Succession 3.0) to talk to a Cisco AS5300 via PRI? |
| 01:45.15 | haryv | heison thats a interesting configuration |
| 01:45.50 | reza_ | yeah, think so |
| 01:46.01 | heison | harvy: i have been searching all over... no one knows how to do this, not even the Nortel folks.. |
| 01:46.41 | tengulre11 | I come from CHINA. but I have not tankers. ;) |
| 01:46.46 | haryv | btw I need to get back to a voip engineer that is running a nortel mcs 5300 class iV switch and see what results thay had interfacing it with a asterisk box. |
| 01:47.05 | haryv | heison where are you at by chance |
| 01:47.14 | heison | harvy: i'm in Toronto |
| 01:47.29 | reza_ | yeah, all calls route to the 601 |
| 01:47.32 | *** join/#asterisk lule (i=lule@host11.201-253-76.telecom.net.ar) |
| 01:47.46 | haryv | If I was there would aid you in getting it configured. It would be a good learning expraince. |
| 01:47.55 | *** join/#asterisk BhaalWTF (n=bhaal@CPE-121-208-127-243.qld.bigpond.net.au) |
| 01:47.56 | *** part/#asterisk lule (i=lule@host11.201-253-76.telecom.net.ar) |
| 01:48.27 | *** join/#asterisk terinjokes (n=spader@adsl-11-150-123.mia.bellsouth.net) |
| 01:48.32 | terinjokes | hey! |
| 01:48.48 | terinjokes | i'm new to this voIP thing |
| 01:49.24 | tengulre11 | terinjokes: It is 'VoIP' not 'voIP'. |
| 01:49.47 | tengulre11 | ;) |
| 01:49.52 | russellb | no, it's VuIP |
| 01:49.55 | russellb | Voice under IP |
| 01:50.09 | reza_ | files |
| 01:50.28 | terinjokes | no, no CAStIP |
| 01:50.37 | reza_ | er |
| 01:51.06 | heison | harvy: i have tried 3 different Nortel vendors in Toronto but no one knows how to connect a media gateway to Nortel Option 11C... one of the vendors actually escallated the problem to Nortel, who claims it will not spend time to offer any help |
| 01:51.07 | terinjokes | Compressed Audio Streams through Internet Protocals |
| 02:08.52 | *** join/#asterisk stkn (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 02:13.17 | *** join/#asterisk inv_Arp (i=junya@c-71-206-88-100.hsd1.fl.comcast.net) |
| 02:18.29 | *** part/#asterisk terinjokes (n=spader@adsl-11-150-123.mia.bellsouth.net) |
| 02:21.21 | conico | i am looking at ip phones now and i saw that a model has 2 ethernet interfaces.why there is 2 ethernet interfaces on an ip phone? |
| 02:24.34 | [TK]D-Fender | conico : So you can plug it in-line with a PC so you don't need to take up another jank on your switch |
| 02:24.51 | [TK]D-Fender | jack |
| 02:25.10 | conico | thank you that is good |
| 02:26.06 | *** join/#asterisk nassy (n=nassy@207-38-197-201.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com) |
| 02:26.30 | conico | is this phone get power from the switch ? |
| 02:27.03 | [TK]D-Fender | conico : Many are capable of PoE (power over ethernet, depends on the model |
| 02:27.24 | conico | thanks again |
| 02:28.26 | [TK]D-Fender | file :y0 |
| 02:28.33 | file | wazzup? |
| 02:29.04 | [TK]D-Fender | file: back from martil arts followed by groceries and I'm just tired now.... |
| 02:29.13 | file | yay tired |
| 02:29.15 | [TK]D-Fender | file : I need a friggen vacation. |
| 02:29.26 | file | vacations are not allowed |
| 02:30.57 | *** join/#asterisk michaelo (n=michaelo@adsl-068-159-111-129.sip.gsp.bellsouth.net) |
| 02:31.03 | mcab | don't tell my boss that, I'm about to take off for two weeks... |
| 02:38.18 | *** join/#asterisk fholmes (n=fholmes@cpe-68-201-200-58.houston.res.rr.com) |
| 02:44.26 | [hC] | any of you guys use hylafax? |
| 02:45.58 | h0 | anyone have an ETA for 1.4 |
| 02:48.17 | [hC] | h0: less than 30 days. |
| 02:48.30 | h0 | k thanks |
| 02:49.38 | Ryanw | i've used hylafax once, not sure i'll be much help. |
| 02:49.39 | h0 | Am I correct in my understanding that 1.4 will help to do a lot of the configuration that is needed itself |
| 02:49.54 | *** join/#asterisk aXanaXa (n=m@ppp-69-219-149-17.dsl.chcgil.ameritech.net) |
| 02:50.41 | aXanaXa | Hey anyone here installed TrixBox on a Dell Dimension E520? |
| 02:50.59 | *** join/#asterisk pipipi (n=suedoh@adsl-68-125-32-90.dsl.irvnca.pacbell.net) |
| 02:51.02 | pipipi | Nov 2 18:46:11 NOTICE[16048] chan_sip.c: -- Registration for '2138050567@sip.broadvoice.com' timed out, trying again (Attempt #7) |
| 02:51.05 | pipipi | Nov 2 18:46:11 DEBUG[16048] chan_sip.c: Stopping retransmission on '6977fc8476a7e7c20e724ea641c67e9c@sip.broadvoice.com' of Request 108: Match Found |
| 02:51.08 | pipipi | Nov 2 18:46:11 DEBUG[16048] chan_sip.c: Scheduled a registration timeout for sip.broadvoice.com id #20 |
| 02:51.11 | pipipi | anybody know what is going on here? |
| 02:51.23 | pipipi | i am fairly new with asterisk and I am using Trixbox w/ BroadVoice |
| 02:51.38 | pipipi | for incoming/outgoing phone calls using SIP |
| 02:52.46 | nassy | aXanaXa: try channel freepbx if no one here has |
| 02:53.13 | aXanaXa | k thanks |
| 02:53.43 | *** join/#asterisk brc_ (n=brc___@pdpc/supporter/basic/brc) |
| 03:10.18 | SuPrSluG | pipipi:try another broadvoice server. it happens with them regularly. keep 2 in your config and switch when they suck |
| 03:10.30 | *** join/#asterisk MIXX941 (n=mark@unaffiliated/mixx941) |
| 03:10.45 | *** join/#asterisk hoobastooba (n=ckwall@c-67-169-248-217.hsd1.ut.comcast.net) |
| 03:12.10 | *** join/#asterisk trelane (i=trelane@unaffiliated/trelane) |
| 03:13.39 | hoobastooba | here is my dilemma... to use ChanIsAvail to keep subsequent calls from the queue from ringing queue members phones I have to use channel Local. If I use Channel Local, the Queue does not record the call. For the call to be recorded with the in queue monitoring i have to use channel SIP. I need both features. What other options are there? |
| 03:17.02 | *** part/#asterisk Agrajag- (n=filip@c211-30-4-5.artrmn1.nsw.optusnet.com.au) |
| 03:17.56 | trelane | hoobastooba, set each phone in the queue to only allow one cal path |
| 03:19.01 | hoobastooba | i should have mentioned that I am using addqueuemember for dynamic agents. Is this possible still? |
| 03:20.35 | *** join/#asterisk Un1x (n=x@74.96.130.207) |
| 03:20.54 | Un1x | ;? |
| 03:22.10 | *** join/#asterisk teknoprep (n=tekon@unaffiliated/teknoprep) |
| 03:24.22 | *** join/#asterisk nsgn (n=nsgn@cpe-66-69-197-178.austin.res.rr.com) |
| 03:24.40 | *** join/#asterisk riddlebox (n=james@75-132-205-166.dhcp.stls.mo.charter.com) |
| 03:24.49 | nsgn | evenin' all. can someone throw out the name of a very cheap VoIP provider they are pleased with? |
| 03:25.05 | nsgn | as far as getting a pstn number, that is |
| 03:26.50 | *** join/#asterisk osiris (n=osiris@c-71-205-27-131.hsd1.mi.comcast.net) |
| 03:27.13 | nsgn | :-/ anyone alive? |
| 03:29.49 | *** join/#asterisk topping (n=topping@207.47.6.182.static.nextweb.net) |
| 03:34.06 | Ryanw | why would asterisk delay for 2 seconds before executing exten => 1,1 in an IVR? |
| 03:34.33 | nsgn | anyone? i'm looking for a recommendation for a SIP pstn service provider |
| 03:36.31 | *** join/#asterisk hohum (n=dcorbe@69-175-203-11.chvlva.adelphia.net) |
| 03:36.33 | hohum | hello |
| 03:36.57 | hohum | when I'm reading the SIP RFC I saw a reference to [H14.43] with regards to User-Agent handling |
| 03:37.03 | hohum | what is [H14.43] |
| 03:37.07 | hohum | and how do I find that document? |
| 03:37.18 | hohum | I tried googling for it and all I found was more references to it |
| 03:45.24 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 03:45.24 | *** mode/#asterisk [+o mog] by ChanServ |
| 03:52.45 | *** join/#asterisk bmg505 (n=leon@c1-253-14.rndf.isadsl.co.za) |
| 03:52.54 | [TK]D-Fender | Ryanw : Can you pastebin the call and your dialplan |
| 04:01.23 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 04:05.54 | Ryanw | D-Fender, i found the problem, direct dialing from the main menu, extensions in the 100-199 range and 3 second digit timeout. |
| 04:06.31 | *** join/#asterisk SwK (n=Silik0nJ@c-71-56-122-62.hsd1.ga.comcast.net) |
| 04:11.34 | *** part/#asterisk fx0 (n=fx0@cypher.punk.net) |
| 04:12.56 | [TK]D-Fender | Ryanw : Yup.. overlap ial for the win... |
| 04:12.59 | [TK]D-Fender | dial* |
| 04:15.01 | *** join/#asterisk BhaalWK (i=bhaal@freenode/unconfirmed/bhaal) |
| 04:16.39 | ManxPower | Ryanw: noob 8-) |
| 04:17.55 | Ryanw | MaxnPower: anyone experienced at anything knows that sometimes a simple solution is overlooked. |
| 04:18.01 | ManxPower | What I do is reserve a menu option on the main IVR menu for "if you know your party's extension dial 1 |
| 04:18.38 | ManxPower | Ryanw: what country are you in? |
| 04:18.51 | Ryanw | Maxn, yeah unfortunately i've already got all the recordings professionaly done and it would have to be re-recorded to include "if you know the extension dial 4" |
| 04:18.58 | Ryanw | I'm an Aussie. |
| 04:19.12 | ManxPower | ah. Don't know the dialing rules for down there. |
| 04:20.52 | Ryanw | Australia: all numbers are 10 digits the first 2 digits are either a mobile / state prefix. 0011 prefix for international |
| 04:21.09 | ManxPower | also 0 for outside line and 999 for emergency services? |
| 04:21.31 | Ryanw | whats this outside line crap? and in Australia 000 for emergency services |
| 04:22.06 | ManxPower | outside line is what larger pbxs use to avoid using dialing timeouts. |
| 04:22.22 | Ryanw | 0118 999 88199 9119 7253....hehe |
| 04:22.55 | Ryanw | no need to tell it about an outside line, if its more then 5 digits and not 000, then its external |
| 04:23.31 | ManxPower | and if it is 3 digits how long do you wait for the call to another internal extensions to start? |
| 04:23.59 | Ryanw | ok, point taken. |
| 04:24.03 | ManxPower | i.e. how do you know the user did not just pause to look at the number somewhere. |
| 04:24.48 | [TK]D-Fender | ManxPower : load chan_psychic.so of course! |
| 04:25.21 | ManxPower | Aussieland has a similar design to the USA, just with different key digits. In the USA it is common for this: |
| 04:25.26 | heison | does anyone know of a good Nortel person who I can rely on with integrating Nortel M1 Option 11C with AS5300 over PRI NI2?? |
| 04:26.27 | ManxPower | Dial 9 for an outside line, never start extensions with digit 9. 911 is emergency services, the international access is 011 and incountry non-local calls start with a 1 |
| 04:28.05 | ManxPower | So on my systems, extens 2000-7999 can be extensions, 911 or 9911 will go to emergenct services, local call are 9+7-digits, toll are 9+1+10-digits |
| 04:29.58 | Ryanw | ManxPower, whats your OS for *? |
| 04:30.07 | ManxPower | Ryanw: linux 8-) |
| 04:30.33 | Ryanw | what distro, i'm using FC5 |
| 04:31.03 | ManxPower | I'm a fan of Mandrake for too many reasons to go into. |
| 04:36.58 | *** join/#asterisk SwK (n=Silik0nJ@c-71-199-179-121.hsd1.ga.comcast.net) |
| 04:38.53 | NDT | go away SwK |
| 04:44.55 | *** join/#asterisk gchaix (n=gchaix@osuosl/staff/gchaix) |
| 04:45.35 | *** part/#asterisk gchaix (n=gchaix@osuosl/staff/gchaix) |
| 04:45.52 | JT | Ryanw: not all australian numbers are 10 digits |
| 04:46.10 | JT | standard landlines with the area code included, and mobile numbers are |
| 04:46.20 | JT | but there's stuff like 13xxxx |
| 04:46.49 | JT | and people don't dial numbers with area codes unless they have to :) |
| 04:48.43 | Juggie | 10digit dialing is forced in alot of places now |
| 04:49.19 | JT | in .au? |
| 04:56.00 | *** join/#asterisk obiwanmikenolte (n=obiwanmi@71-10-182-240.dhcp.stls.mo.charter.com) |
| 04:57.39 | hads | Not in .au |
| 04:58.13 | hads | You didn't even have to dial 10 digits on your mobile for the local area last time I was in .au |
| 05:01.20 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 05:01.38 | *** join/#asterisk Guadamu1 (n=edgar@201.195.22.6) |
| 05:02.12 | *** part/#asterisk hoobastooba (n=ckwall@c-67-169-248-217.hsd1.ut.comcast.net) |
| 05:02.28 | Guadamu1 | hello everybody. I'm new in the channel, so nice to meet you. I'm guadamux, from Costa Rica |
| 05:02.50 | obiwanmikenolte | Howdy |
| 05:03.46 | JT | hads: i know, i live here, you don't :) |
| 05:03.57 | JT | not sure if Juggie was refering to some corporate pabxes |
| 05:04.31 | Guadamu1 | I'm looking for the best way to have load-balancing redundant asterisk servers. Any suggestion?? I believe that I have to use LVS (Linux Virtual Server) o something like that... |
| 05:04.51 | JT | there's only 5 landline area codes |
| 05:11.47 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 05:12.32 | Ryanw | Guadamu1, i'm not sure if there is something already constructed or not but if there is not you could try heartbeat, lvs & asterisk-realtime sip |
| 05:14.40 | Guadamu1 | thank you, Ryanw |
| 05:16.07 | JT | hads: oh, and before when i said "you don't" i was refering to having to dial area codes, i wasn't being snobbish :P |
| 05:17.02 | hads | JT: I know, I was backing you up! :) |
| 05:17.59 | hads | JT: I'm coming to visit you .au's in January for LCA |
| 05:18.07 | *** join/#asterisk aadilismail (n=adilisma@202.38.55.114) |
| 05:24.06 | JT | cool |
| 05:24.10 | JT | in my city too |
| 05:24.36 | Guadamu1 | qdk, are you there? |
| 05:26.11 | *** join/#asterisk linlin (n=linlin@71.194.70.13) |
| 05:33.10 | *** part/#asterisk Ryanw (n=cableguy@ge0-0-15-lns0.207alg.qx21.net) |
| 05:34.39 | crosslimits | hi guys |
| 05:41.13 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
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| 05:53.07 | *** join/#asterisk DocHolliday (i=RogerRab@gateway/gpg-tor/key-0x0E4F6D6C) |
| 05:59.32 | *** join/#asterisk BhaalWK (i=bhaal@freenode/unconfirmed/bhaal) |
| 06:03.29 | *** part/#asterisk Guadamu1 (n=edgar@201.195.22.6) |
| 06:07.39 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.72.24) |
| 06:08.26 | *** part/#asterisk obiwanmikenolte (n=obiwanmi@71-10-182-240.dhcp.stls.mo.charter.com) |
| 06:09.28 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 06:09.48 | *** join/#asterisk cbm11211 (n=Administ@66.28.182.170) |
| 06:17.18 | DarKnesS_WolF | morning |
| 06:23.03 | *** join/#asterisk VibroMax (n=phisto@83.229.70.154) |
| 06:23.42 | *** join/#asterisk slayer192 (n=slayer19@adsl-70-137-24-211.dsl.okcyok.swbell.net) |
| 06:28.05 | *** join/#asterisk jaike (n=jaike@125.5.144.90) |
| 06:28.56 | jaike | anyone using aastra sip phones? |
| 06:30.14 | *** join/#asterisk wmandra (n=wmandra@c-68-36-74-49.hsd1.nj.comcast.net) |
| 06:39.07 | *** join/#asterisk svenna (n=svenna@p548D42C7.dip0.t-ipconnect.de) |
| 06:39.09 | *** join/#asterisk sjobeck (n=sjobeck@208-151-246-203.dq1sn.easystreet.com) |
| 06:41.10 | *** join/#asterisk xtr-II (n=94752345@S0106000c41ed11e1.vf.shawcable.net) |
| 06:45.04 | *** join/#asterisk JunK-Y (n=junky@modemcable198.14-83-70.mc.videotron.ca) |
| 06:47.08 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 06:47.11 | Chris-NB | hi |
| 06:47.37 | Chris-NB | someone got experience with regex for enum entries? (bind dns server) |
| 06:48.19 | Chris-NB | i've to enter 4 digit extensions, but there are fax extensions mixed in it : / |
| 06:48.30 | Chris-NB | so i've to spare them out |
| 06:50.53 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 07:06.45 | *** join/#asterisk HaMYaI (n=HaMYaI@ppp-58.8.4.69.revip2.asianet.co.th) |
| 07:09.28 | *** join/#asterisk nentis (n=krisa@mail.opensourcery.com) |
| 07:14.12 | nentis | Any reason why dial by directory would not hit VM if unanswered? Punching in the extension works, and the directory works for internal users. Only POTS trunks exibit this behavior. |
| 07:14.41 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 07:21.21 | DarKnesS_WolF | nentis: seems ur dialplan has something wrong |
| 07:22.25 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 07:23.26 | nentis | What is the best method of debugging the dial plan? Starting asterisk with -vv? |
| 07:23.40 | nentis | or is there a useful console debug command |
| 07:28.38 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 07:45.20 | crosslimits | new born ...i have been delete all the sounds file from sounds dir ... how can i reload or download again ??? |
| 07:48.14 | *** join/#asterisk aadilismail (n=adilisma@202.38.55.114) |
| 07:48.34 | aadilismail | sorry got dc .. . |
| 07:48.39 | hads | new born? |
| 07:48.59 | hads | make install should install the sounds. |
| 07:49.23 | aadilismail | i have been deleted all the sounds.. file... no file exist in /var/lib/asterisk/sounds... how can i reload and download again ?/? |
| 07:49.35 | aadilismail | ok |
| 07:50.06 | *** join/#asterisk TheBleh (n=thebleh@ip68-224-138-154.lv.lv.cox.net) |
| 07:50.18 | aadilismail | where shud i enter this command? |
| 07:50.54 | *** join/#asterisk Rhizome (n=Rhizome@host-81-191-147-145.bluecom.no) |
| 07:50.55 | DarKnesS_WolF | aadilismail: also there is asterisk-sounds in asterisk.org |
| 07:51.03 | DarKnesS_WolF | nentis: or from CLI set verbose 99999 |
| 07:51.08 | aadilismail | ok |
| 07:52.53 | nentis | thx |
| 07:53.11 | DarKnesS_WolF | nentis: always good idea to enable the full logger |
| 07:56.20 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 07:56.29 | *** join/#asterisk sahafeez (n=sahafeez@ip68-6-223-92.sd.sd.cox.net) |
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| 07:58.13 | *** join/#asterisk Sasch (n=Admin@host102-30-static.107-82-b.business.telecomitalia.it) |
| 07:58.23 | Sasch | can help me .... |
| 07:59.05 | Sasch | i have a tdm400p .... with one FXO for my pstn (telecom italia Spa) and one FXS for my fax ... |
| 07:59.27 | DarKnesS_WolF | Sasch: go on |
| 07:59.51 | Sasch | i'm italian and is difficult to translate :-P |
| 08:00.04 | Sasch | i want when in pstn asterisk found a fax |
| 08:00.20 | *** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 08:00.34 | Sasch | redirect it in FXS channel |
| 08:00.40 | Sasch | it is possible ?? |
| 08:01.07 | Sasch | beacause i realize a dial plan that WaitExten |
| 08:01.57 | sahafeez | the fax calls is coming in on the tdm400p? |
| 08:02.20 | *** join/#asterisk blueneon (i=hfklows@dsl-146-30-49.telkomadsl.co.za) |
| 08:02.35 | Sasch | yes .. |
| 08:02.38 | DarKnesS_WolF | Sasch: yes |
| 08:02.42 | DarKnesS_WolF | i have the same setup |
| 08:02.48 | DarKnesS_WolF | one line gose directly to the fax machine |
| 08:03.06 | Sasch | asp i link my dial plan |
| 08:03.42 | DarKnesS_WolF | Sasch: in ur extentions.conf go to teh incoming calls context to ur phone line and let it dial the FXS channel ;-) |
| 08:04.28 | Sasch | http://papinicomputer.homelinux.org/ |
| 08:05.45 | blueneon | i have fedora core 3 and asterisk 1.2.12 and zaptel 1.2.10 installed with a TDM400P .. everything is setup and is working perfectly... we currently only have 1 fxo and 1 fxs all incoming calls come via the fxo and are then routed to the internal zap on fxs, i would like to start playing with routing of calls via voip to a remote person. How do I do that, can asterisk forward calls to skype? |
| 08:07.06 | Sasch | <DarKnesS_WolF> my telephone is 0577 807317 ... i make a line in my extension.conf that is exten => 7,1,dial(Zap/2) |
| 08:07.08 | shellshark | not to skype |
| 08:07.21 | bigjb | blueneon: if you want to route to skype you are going to need to use a piece of software like uplink |
| 08:07.31 | shellshark | blueneon: you can use a real VoIP provider, however (not Vonage, not Skype) |
| 08:07.43 | shellshark | blueneon: I recommend ShellShark Networks, https://voip.shellshark.net/ |
| 08:08.10 | blueneon | i live in South Africa |
| 08:08.14 | blueneon | :( |
| 08:08.22 | Sasch | but when i lunch 0577 807317 7 asterisk don't send call in Zap/2 |
| 08:08.22 | blueneon | real voip providers are hard to come by |
| 08:08.56 | blueneon | i dont want it to call a phsyical line, i want it to use voip to voip so its a free call |
| 08:09.01 | shellshark | blueneon: where are you wanting to make the majority of your calls to? .za? |
| 08:09.18 | blueneon | does that mean both me and the remote user would need to subscribe to that shellshark provider? |
| 08:09.22 | shellshark | err, you normally can not call the PSTN for free :p |
| 08:09.22 | nentis | I've had good experience with voicepluse (connect.voicepulse.com) |
| 08:09.46 | blueneon | i dont want to call a PSTN for free, or at all |
| 08:09.47 | shellshark | blueneon: no, a VoIP provider gives you PTSN access over IP |
| 08:09.57 | shellshark | ah |
| 08:10.01 | blueneon | now im confused |
| 08:10.09 | blueneon | we have a normal analog line |
| 08:10.13 | blueneon | clients call that |
| 08:10.16 | blueneon | asterisk answers |
| 08:10.17 | shellshark | then you can have your friend register to your asterisk server directly, and call him as an extension |
| 08:10.23 | blueneon | and forwards to an internal extension |
| 08:10.26 | shellshark | we understand your setup ;) |
| 08:10.34 | blueneon | if busy i want to forward to a voip |
| 08:10.41 | blueneon | (a mate down the road) |
| 08:10.41 | blueneon | heh |
| 08:10.59 | blueneon | aaah |
| 08:11.00 | shellshark | ah, ok, set your mate up an extension on your asterisk box |
| 08:11.01 | blueneon | SIP |
| 08:11.06 | blueneon | ye |
| 08:11.11 | blueneon | doh didnt think of that |
| 08:11.11 | blueneon | lol |
| 08:11.20 | shellshark | SIP or IAX2 or MGCP or SCCP or UNISTIM.... however you want to do it :) |
| 08:11.22 | *** join/#asterisk sjobeck (n=sjobeck@208-151-246-203.dq1sn.easystreet.com) |
| 08:11.56 | Sasch | CtRiX can help me ??? |
| 08:12.05 | nentis | hey sjobeck (Kris from OpenSourcery) |
| 08:12.19 | sjobeck | hi |
| 08:12.36 | sjobeck | kris: do I owe you an email? |
| 08:13.02 | sjobeck | kris: oh, no, sorry, wrong kris, right, yeah, yeah, hey, how are you? |
| 08:13.05 | nentis | uh.. no. Just saying hi. :) (Thanks for the Jay referal.. it worked out great) |
| 08:13.22 | blueneon | whats the best free software that my mate can use to connect/register to asterisk via IP |
| 08:13.24 | sjobeck | nentis: grt. call on us any time. pleasure |
| 08:13.32 | blueneon | IAX2 would be best protocol no? |
| 08:14.13 | sjobeck | bluenon: IAX is not a standard, SIP is, SIP doesnt do all types of NAT, IAX does, far more ITSP's do SIP |
| 08:14.31 | blueneon | but isnt asterisks native IAX? |
| 08:15.03 | sjobeck | bluenon: native sip, native iax, native pstn, native others as well |
| 08:15.22 | blueneon | ok |
| 08:15.41 | blueneon | well i've used x-lite via sip and it seems rather crappy, i get break up in the line etc |
| 08:15.50 | blueneon | the software seems buggy |
| 08:16.00 | blueneon | whats the best free SIP software ? |
| 08:16.11 | monsted | blueneon: asterisk of course ;) |
| 08:16.35 | blueneon | i mean for the client side |
| 08:16.35 | blueneon | :P |
| 08:16.43 | blueneon | ie. instead of x-lite |
| 08:17.06 | monsted | i use idefisk and iax, dunno about sip clients |
| 08:17.56 | blueneon | ok so whats the best iax client? |
| 08:18.51 | _Vile | where's my muffin? |
| 08:19.16 | qdk | blueneon: sjphone |
| 08:19.24 | qdk | blueneon: for SIP. |
| 08:20.04 | blueneon | ta |
| 08:20.10 | jaike | blueneon: are you sure the breakup you hear on the line isnt caused by congestion of you internet connection? we use sip with xlite but we dont have those problems |
| 08:20.37 | *** join/#asterisk EyeCue (n=eyecue@unaffiliated/eyecue) [NETSPLIT VICTIM] |
| 08:20.46 | blueneon | its LAN |
| 08:20.49 | blueneon | so no |
| 08:20.50 | blueneon | ;) |
| 08:21.11 | jaike | were using it in our LAN also, QA listening to agents calls |
| 08:21.22 | qdk | jaike: xlite IS bad... it adds stupid latency. |
| 08:21.31 | blueneon | agreed |
| 08:21.32 | blueneon | :) |
| 08:21.44 | jaike | qdk: well weve not had problems so far |
| 08:21.58 | EyeCue | miranda-im + iax.dll :D |
| 08:22.00 | EyeCue | :D |
| 08:22.14 | qdk | jaike: never said that, but you probably never looked at the latency. |
| 08:22.35 | blueneon | hmm what port do i open on my firewall to allow remote sip clients to register with my asterisk box? |
| 08:24.39 | hads | 5060 UDP |
| 08:24.56 | hads | You'd want to know that you secured your box first of course. |
| 08:25.21 | EyeCue | laf at box. |
| 08:26.24 | Sasch | CALLERID is the num of a persona that call me ... for look wath number compose it ... |
| 08:26.36 | hads | ? |
| 08:26.46 | Sasch | exusme |
| 08:26.48 | sjobeck | bluenon: sjphone |
| 08:27.00 | Sasch | CALLERID is the number of a person thatt call me |
| 08:27.45 | sjobeck | bluenon: udp/5060 as well as udp/10000-20000 |
| 08:28.05 | Sasch | there is a viarible that explain the number that digit a person form example my number is 456 if the person when call me digit 456 789 (for look 789) |
| 08:28.16 | Sasch | i don't speak very well english I'm italian ... |
| 08:28.23 | Sasch | excusme -.-' |
| 08:29.47 | *** join/#asterisk inspired (n=mikael@85.221.7.59) |
| 08:30.49 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust944.leed.cable.ntl.com) |
| 08:35.10 | hads | Sasch: EXTEN |
| 08:35.27 | *** join/#asterisk blueneon` (n=blueneon@dsl-146-30-49.telkomadsl.co.za) |
| 08:36.15 | hads | Sasch: http://www.voip-info.org/wiki-Asterisk+variables#PredefinedChannelVariables |
| 08:37.25 | Sasch | thanks |
| 08:40.27 | Sasch | i have want to realize a dial plan that when a person call me at my number (0577807317) start a menu (1 for sales etc......) and when digit 05778073177 call redirect to my Zap/2 |
| 08:40.31 | Sasch | can help me ... |
| 08:41.00 | Sasch | excusme when call 05778073177 call redirect to my Zap/2 |
| 08:49.22 | *** join/#asterisk NDT (n=noone@cpe-74-70-211-81.nycap.res.rr.com) |
| 08:50.03 | nentis | . |
| 08:50.26 | NDT | heh...I am either getting old or I forget...But wasn't there a way to make an extension so say a sip friend using xlite when they dialed an extension could create a mailbox? |
| 08:50.52 | NDT | Instead of having the box already predefined |
| 09:00.54 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 09:05.02 | *** join/#asterisk FTexcom (n=FTexcom@14.Red-80-26-4.staticIP.rima-tde.net) |
| 09:05.12 | FTexcom | <PROTECTED> |
| 09:05.15 | *** join/#asterisk Skept (n=melancho@202.65.153.254) |
| 09:07.47 | *** join/#asterisk zigman (i=zigman@irc.zigman.de) |
| 09:11.21 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 09:11.40 | *** part/#asterisk Skept (n=melancho@202.65.153.254) |
| 09:11.52 | *** join/#asterisk blueneon (n=blueneon@dsl-146-30-49.telkomadsl.co.za) |
| 09:12.20 | blueneon | hmm, i am trying to let an friend register on asterisk using sjphone (SIP) |
| 09:12.29 | blueneon | for some reason its not working |
| 09:12.36 | blueneon | and asterisk isnt giving any msg's |
| 09:12.59 | blueneon | but if i do it from the LAN it connect |
| 09:13.00 | blueneon | s |
| 09:13.18 | blueneon | i have opened my firewall to allow all ports incoming (just to test) |
| 09:15.49 | *** join/#asterisk fenlander (n=fenlande@82.152.81.57) |
| 09:20.09 | DarKnesS_WolF | blueneon: pastbin ur sip.conf |
| 09:20.26 | infinity1 | i just updated from 1.2.10 to 1.2.13 and i'm getting errors when dialing. dial_exec_full: Unable to create channel of type 'IAX2' (cause 3 - No route to destination) |
| 09:20.27 | DarKnesS_WolF | blueneon: and edit /etc/asterisk/logger.conf and enable full loger |
| 09:20.49 | DarKnesS_WolF | infinity1: iax2 show peers |
| 09:20.58 | DarKnesS_WolF | and make sure the person ur calling is registered |
| 09:21.27 | FTexcom | <PROTECTED> |
| 09:21.42 | infinity1 | DarKnesS_WolF: 4 iax peers. |
| 09:22.01 | infinity1 | DarKnesS_WolF: i'm trying to use voipjet. looks fine |
| 09:22.16 | infinity1 | something changed from .10 to .13 hmm |
| 09:22.47 | DarKnesS_WolF | infinity1: no route to destination means it can't reach the extentions / context ur dialing. |
| 09:23.28 | *** join/#asterisk A-Tuin|work (n=A-Tuin@212.41.185.81) |
| 09:25.25 | infinity1 | i removed the dns name of voip jet and put in the IP addr of their server |
| 09:25.25 | infinity1 | strange |
| 09:25.25 | infinity1 | seems to work now. |
| 09:26.47 | infinity1 | strange. why isn't dns working inside asteirks |
| 09:27.33 | *** join/#asterisk jeremy_g (n=jerms@static-213-115-44-90.sme.bredbandsbolaget.se) |
| 09:29.31 | jaike | infinity1: nslookup resolving properly? |
| 09:29.39 | *** join/#asterisk Arno[Slack] (n=hellSOUN@master.infinityperl.org) |
| 09:30.09 | *** join/#asterisk eivindtr (n=eivindtr@062016176152.customer.alfanett.no) |
| 09:30.59 | *** join/#asterisk chapeaurouge (n=chapeaur@80.92.83.35) |
| 09:32.30 | infinity1 | jaike: yea. strange. after " == Refreshing DNS lookups." appeared on the console, it started working |
| 09:36.19 | aadilismail | hi |
| 09:36.43 | aadilismail | new born... WARNING[12036]: app_voicemail.c:2412 leave_voicemail: No entry in voicemail config file for '123'.. where voice mail conf??? |
| 09:37.11 | infinity1 | /etc/asterisk/voicemail.conf |
| 09:37.40 | aadilismail | ok |
| 09:47.27 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43-static.72-81-b.business.telecomitalia.it) |
| 09:52.16 | DarKnesS_WolF | aadilismail: and every time u say new born go read the book or check voip-info.org |
| 09:52.50 | jeremy_g | DarKnesS_WolF: :) dont be hard on him |
| 09:53.05 | DarKnesS_WolF | jeremy_g: this is the 3rd new born thing ;-) |
| 09:53.21 | FTexcom | mine's the kind of problem everyboyd avoids |
| 09:53.22 | jeremy_g | DarKnesS_WolF:oh!! |
| 09:53.26 | FTexcom | *everybody |
| 09:53.55 | DarKnesS_WolF | jeremy_g: hehe it's okay ;-) |
| 09:54.16 | DarKnesS_WolF | FTexcom: what is ur problem ? |
| 09:54.34 | FTexcom | SPA941 can't transfr calls |
| 09:54.49 | DarKnesS_WolF | FTexcom: i don't know what is SPA941 |
| 09:54.51 | DarKnesS_WolF | :-) |
| 09:54.55 | FTexcom | it's a sip phone |
| 09:54.58 | FTexcom | <PROTECTED> |
| 09:55.54 | DarKnesS_WolF | FTexcom: i don't know but for ex in my giptel phones i press hold then dial 215 then push transfaer and i hangup |
| 09:55.57 | DarKnesS_WolF | and it's working fine |
| 09:56.15 | DarKnesS_WolF | FTexcom: so i think u need to read ur phone manuals |
| 09:56.27 | FTexcom | I'm doing it the way the phone manual says |
| 09:56.34 | FTexcom | I even updated firmware |
| 09:57.07 | DarKnesS_WolF | i don't have one to help u sorry |
| 09:57.29 | FTexcom | no problem, I can still blame on the phones :P |
| 09:58.09 | DarKnesS_WolF | good :P |
| 09:59.36 | FTexcom | it's a very strange problem...Thomson sip phones can transfer perfectly, ZAp extension can't, SPA can't...damn.. |
| 09:59.44 | DarKnesS_WolF | FTexcom: |
| 09:59.45 | DarKnesS_WolF | tell me |
| 09:59.55 | DarKnesS_WolF | u press xfer right? |
| 09:59.57 | DarKnesS_WolF | then u get a tone |
| 10:00.01 | FTexcom | correct |
| 10:00.01 | DarKnesS_WolF | u dial 215 |
| 10:00.03 | DarKnesS_WolF | right? |
| 10:00.22 | DarKnesS_WolF | after he pics up / answer u click xfer again ? |
| 10:00.23 | *** join/#asterisk blueneon` (n=blueneon@dsl-146-30-49.telkomadsl.co.za) |
| 10:00.28 | FTexcom | correct |
| 10:00.44 | FTexcom | after that, my phone hangs up and 215 can only hear silence |
| 10:01.19 | DarKnesS_WolF | FTexcom: hmmmm |
| 10:01.23 | DarKnesS_WolF | there is 2 ways in the manual |
| 10:01.24 | DarKnesS_WolF | this is one |
| 10:01.28 | DarKnesS_WolF | did u try the other ? |
| 10:02.18 | *** join/#asterisk moon06 (n=michael@82.228.240.97) |
| 10:02.48 | FTexcom | the blind transfer method? |
| 10:03.14 | DarKnesS_WolF | nop |
| 10:03.21 | *** join/#asterisk apardo (n=apardo@87.217.144.170) |
| 10:03.27 | DarKnesS_WolF | 2 mothods for attanding transfer |
| 10:03.40 | DarKnesS_WolF | FTexcom: http://www.sipura.com/Documents/SPA941AdminGuide.pdf check the transfer |
| 10:04.07 | DarKnesS_WolF | FTexcom: also tell me something all 3 phones are in the same network "LAN "? |
| 10:05.27 | FTexcom | yes |
| 10:05.43 | FTexcom | xferLx dosn't work |
| 10:06.35 | FTexcom | hum... |
| 10:06.46 | FTexcom | I can't believe |
| 10:06.50 | FTexcom | this is like going to the docto |
| 10:07.05 | FTexcom | doctor* |
| 10:10.24 | FTexcom | everything's working correctly now |
| 10:10.43 | *** join/#asterisk bkw_ (n=brian@adsl-64-149-40-112.dsl.tul2ok.sbcglobal.net) |
| 10:10.57 | FTexcom | I just don't get it |
| 10:11.10 | bkw_ | give it time |
| 10:12.08 | RoyK | bkw_: you don't get it either, do you? |
| 10:12.14 | DarKnesS_WolF | FTexcom: the transfer works ? |
| 10:13.44 | FTexcom | yes, now it works |
| 10:15.28 | FTexcom | I'm confused.. |
| 10:15.44 | bkw_ | RoyK, apparently I don't... |
| 10:15.56 | FTexcom | I think you scared the shit out of the phoen DarKnesS_WolF |
| 10:16.14 | bkw_ | the Asterisk Architecture blows.. thats about the extent of what I get. |
| 10:18.10 | RoyK | bkw_: it's too great for a small man like you to grasp |
| 10:18.48 | bkw_ | RoyK, maybe so :( |
| 10:19.27 | RoyK | a complex network of bugs supporting oneanother and filling into each other, forming a jelly-like substance claiming to be stable |
| 10:19.33 | bkw_ | this dialplan thing you speak of... I wasn't aware dialplans could have logic.. |
| 10:19.39 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 10:20.07 | bkw_ | I thought a dial plan routes calls from one place to another... at least thats the impression I got |
| 10:21.33 | monsted | one nice hack i've seen is to have the dialplan check if a bluetooth enabled cell phone is within reach, then choose not to dial it if it's sitting next to a wired phone anyway |
| 10:22.20 | bkw_ | yes thats hackish at best |
| 10:23.55 | monsted | pretty neat, i thought :) |
| 10:36.38 | *** join/#asterisk juice (n=juice@mo-76-0-43-187.dhcp.embarqhsd.net) |
| 10:37.34 | Simplix | is there any way to put call farwarding in odbc ? |
| 10:40.12 | bkw_ | look for app_dbodbc |
| 10:41.44 | *** join/#asterisk creativx (n=creadure@226.62-97-205.bkkb.no) |
| 10:42.26 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 10:43.15 | pif | wet dreams.. |
| 10:45.54 | *** join/#asterisk Strom_M (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 10:49.01 | *** join/#asterisk |Johny| (n=gomesper@bacus.corp.fccn.pt) |
| 10:49.10 | |Johny| | Hello |
| 10:49.24 | |Johny| | have you guys tried the Cisco 7970 ? |
| 10:49.29 | *** join/#asterisk Eliran_Itzhak (n=eliran@bzq-82-81-63-217.red.bezeqint.net) |
| 10:49.31 | |Johny| | Im making the config file |
| 10:49.39 | |Johny| | and I dont know what to put in: |
| 10:49.51 | |Johny| | <PROTECTED> |
| 10:49.51 | |Johny| | <PROTECTED> |
| 10:50.13 | |Johny| | whats this "versionStamp"? |
| 10:51.04 | *** join/#asterisk ScottyTM (n=ScottyTM@marlin.42h.de) |
| 10:51.51 | ScottyTM | hi |
| 10:52.25 | ScottyTM | is there a possibility to include other contexts in Asterisk realtime extensions? |
| 10:52.53 | ScottyTM | like include => othercontext |
| 10:53.40 | aadilismail | new born... in sip.conf or extensions.conf... suddenly my finger touched key " V " and i saw " VISUAL" ,,,wats it ??? |
| 10:54.47 | DarKnesS_WolF | aadilismail: what ediror ur using ?vim ? |
| 10:54.57 | DarKnesS_WolF | aadilismail: and this is not a freaking asterisk thing ! it's a vim thing |
| 10:54.58 | ScottyTM | it's a vim feature |
| 10:55.19 | aadilismail | ok let me check |
| 10:55.22 | aadilismail | thanx |
| 10:55.24 | ScottyTM | for the use of copy and paste for example |
| 10:55.54 | DarKnesS_WolF | aadilismail: /j #vim |
| 10:56.01 | aadilismail | ok |
| 10:56.13 | *** join/#asterisk kuto (n=f3fsa@125.60.241.24) |
| 10:59.46 | DarKnesS_WolF | FTexcom: cool let me scare some more phones :P |
| 11:03.58 | *** join/#asterisk creativx (n=creadure@226.62-97-205.bkkb.no) |
| 11:05.56 | Simplix | thanx bkw_ for app_dbodbc |
| 11:10.44 | *** join/#asterisk ghenry (n=ghenry@82-69-192-46.dsl.in-addr.zen.co.uk) |
| 11:15.35 | *** join/#asterisk MrChimpy (n=MrChimpy@212.158.8.162) |
| 11:18.00 | DarKnesS_WolF | RoyK: what is htcpcp ? |
| 11:18.13 | RoyK | ~htcpcp |
| 11:18.17 | jbot | [htcpcp] the 'Hyper Text Coffee Pot Control Protocol' defined in RFC2324 |
| 11:18.45 | *** join/#asterisk neoalex (n=neoalex@user-12ldt1n.cable.mindspring.com) |
| 11:19.12 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 11:19.36 | neoalex | hello... can asterisk and stun server run properly on the same machine |
| 11:20.38 | neoalex | bassically I need to connect a client that is behind a nat to an asterisk, and I suppose stun is the way to do it |
| 11:21.13 | DarKnesS_WolF | neoalex: yes |
| 11:21.24 | DarKnesS_WolF | neoalex: actually u can try without stun |
| 11:21.31 | DarKnesS_WolF | i have some clients behind nat and it's working |
| 11:21.42 | DarKnesS_WolF | in sip use canreinvite = no nat = yes |
| 11:21.50 | DarKnesS_WolF | or using iax and fwd one port from there router ;-) |
| 11:22.02 | neoalex | I have that and it is not working, at least for one particular provider |
| 11:22.16 | neoalex | they did the nat, and I suspect they even blocked some ports |
| 11:22.40 | neoalex | so the client won't connect, but normally it works |
| 11:23.10 | neoalex | so for the clients using that ISP I would need to use stun or something of the sort |
| 11:23.14 | neoalex | Xtunnels maybe |
| 11:23.51 | neoalex | xtunnels.org is not working atm so that's why I would try stun |
| 11:24.35 | DarKnesS_WolF | neoalex: stund |
| 11:26.18 | neoalex | right, and that will work fine on the same machine as asterisk |
| 11:26.43 | neoalex | because I read some post on some forum that they would not work because they listen on the same port |
| 11:27.29 | DarKnesS_WolF | neoalex: i think u can control that |
| 11:33.27 | *** join/#asterisk Ebola (n=Ebola@host86-136-130-111.range86-136.btcentralplus.com) |
| 11:36.19 | aadilismail | what is HAYES COMMANDS??? |
| 11:37.38 | monsted | aadilismail: AT commands for modems |
| 11:37.39 | *** join/#asterisk ToyMan (n=stuq@74-32-36-35.dsl1.mdl.ny.frontiernet.net) |
| 11:37.52 | monsted | (and there's no need to yell) |
| 11:38.45 | *** join/#asterisk zazzizza (n=gabriel@syrah.cespi.unlp.edu.ar) |
| 11:41.55 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 11:47.35 | *** join/#asterisk spaghetty (n=user@lugbari/people/spaghetty) |
| 11:57.16 | blueneon` | is there somewhere in asterisk where one can up the volume for internal lines? |
| 11:57.19 | blueneon` | (zap) |
| 11:58.42 | blueneon` | the internal phone is on max vol yet its still very soft when i talk to a caller i can hardly hear them |
| 11:59.09 | *** part/#asterisk jaike (n=jaike@125.5.144.90) |
| 12:00.48 | FTexcom | blueneon` rxgain, txgain values |
| 12:04.00 | blueneon` | in which file? |
| 12:05.00 | *** join/#asterisk zeppelin_ (n=zeppelin@201-34-96-24.paemt700.dsl.brasiltelecom.net.br) |
| 12:05.33 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.72.24) |
| 12:07.12 | blueneon` | zaptel.conf i imagine |
| 12:07.16 | FTexcom | blueneon` yes |
| 12:07.25 | blueneon` | tires* |
| 12:07.27 | blueneon` | :) |
| 12:07.31 | zazzizza | /etc/asterisk/zapata.conf |
| 12:07.42 | FTexcom | you can try the ztmonitor command to check the values of the rxgain and txgain |
| 12:07.43 | blueneon` | line 8: Unknown keyword 'rxgain' |
| 12:07.43 | blueneon` | line 9: Unknown keyword 'txgain' |
| 12:07.44 | blueneon` | :/ |
| 12:08.36 | FTexcom | it's on zapata.conf, sorry |
| 12:08.48 | blueneon` | lol ok |
| 12:08.49 | blueneon` | ta |
| 12:10.15 | FTexcom | how much would you paid for a box with a 4 channel rdsi digium card and a functional asterisk? |
| 12:10.56 | DarKnesS_WolF | what is rdsi ? |
| 12:11.25 | FTexcom | ISDN...RDSI it's on spanish |
| 12:11.57 | DarKnesS_WolF | digium makes ISDN cards ? |
| 12:13.09 | FTexcom | http://keison.co.uk/digium/digium_b410p.htm yes |
| 12:13.19 | RoyK | DarKnesS_WolF: you may say that, but I'd recommend using sangoma's cards instead. less buggy |
| 12:13.45 | DarKnesS_WolF | RoyK: i don't use ISDN cards |
| 12:13.55 | DarKnesS_WolF | i have analog TDM400P and it works very good ! |
| 12:14.41 | RoyK | pots is evil |
| 12:14.47 | FTexcom | I'm making a budget for a client based on a ISDN 4 channels card with asterisk |
| 12:15.24 | *** join/#asterisk zotz (n=zotz@24.244.133.107) |
| 12:15.26 | DarKnesS_WolF | RoyK: what ? |
| 12:15.32 | DarKnesS_WolF | FTexcom: i see i have one of this but T1/E1 |
| 12:15.42 | *** join/#asterisk zamba (i=marius@flage.org) |
| 12:17.18 | zazzizza | pots = Plain Old Telephone System/Service depending on the teacher :-S |
| 12:18.09 | monsted | we usually refer to our POTS-enabled colleagues as telegraph operators... they don't seem to like that ;) |
| 12:18.26 | blueneon` | I have a TDM400P too and its perfect |
| 12:18.38 | DarKnesS_WolF | blueneon`: i have 5 of it :P mawhahahhahaha |
| 12:18.46 | monsted | (i work for one of those evil telco monopolies, doing VoIP for larger customers) |
| 12:18.46 | blueneon` | nice |
| 12:18.51 | DarKnesS_WolF | blueneon`: at work none is mine :P but soon i'll get my own ;-) |
| 12:18.57 | blueneon` | i have 1 lol with only 1 fxo and 1 fxs |
| 12:19.12 | DarKnesS_WolF | blueneon`: that will be like mine |
| 12:19.16 | blueneon` | but im going to be expanding to two fxo's and two fxs soon |
| 12:19.17 | DarKnesS_WolF | how much did it cost? |
| 12:19.25 | FTexcom | Here we have a 6 fxo and 6 fxs |
| 12:19.45 | blueneon` | mine was about $250 |
| 12:19.50 | blueneon` | incl the mods ofc |
| 12:20.01 | DarKnesS_WolF | ofc ? |
| 12:20.07 | blueneon` | ofcourse |
| 12:20.12 | DarKnesS_WolF | yes it's around 216 onlin ;-) |
| 12:20.12 | blueneon` | :P |
| 12:20.31 | blueneon` | *shrug* oh well |
| 12:20.32 | DarKnesS_WolF | but that almost a one month salary for me :-s |
| 12:20.37 | DarKnesS_WolF | in egypt |
| 12:20.44 | blueneon` | i live in South Africa |
| 12:20.44 | blueneon` | :P |
| 12:20.53 | FTexcom | Spain.. |
| 12:21.00 | DarKnesS_WolF | blueneon`: ahh rich :P |
| 12:21.01 | FTexcom | and it's almost a salary here too |
| 12:21.09 | DarKnesS_WolF | blueneon`: i have a friend thinking of moving to sa |
| 12:21.21 | blueneon` | is he mad? |
| 12:21.23 | blueneon` | :P |
| 12:21.24 | DarKnesS_WolF | FTexcom: lets move to blueneon` he is inviting us :P |
| 12:21.28 | DarKnesS_WolF | see ? |
| 12:21.45 | FTexcom | south africa? no thanks |
| 12:21.45 | DarKnesS_WolF | blueneon`: invitation accept |
| 12:21.45 | blueneon` | exactly |
| 12:21.45 | blueneon` | im looking to leave |
| 12:21.45 | DarKnesS_WolF | FTexcom: the country of dimons :P |
| 12:21.51 | blueneon` | but i run my own biz here |
| 12:21.53 | DarKnesS_WolF | blueneon`: really ? advice stay awya from egypt |
| 12:21.54 | blueneon` | *shrug* |
| 12:21.58 | FTexcom | I have an angolan friend...he dosn't like south africans a lot |
| 12:22.03 | blueneon` | i would never live in egypt |
| 12:22.13 | blueneon` | i wanna go to Australia |
| 12:22.55 | DarKnesS_WolF | blueneon`: ah cool same here or any other country :-) |
| 12:23.02 | blueneon` | haha |
| 12:23.09 | blueneon` | im actually scottish |
| 12:23.11 | FTexcom | I'm pretty fine in Spain... |
| 12:23.16 | blueneon` | my parents moved here when i was small |
| 12:23.40 | blueneon` | tho i moved to the UK when i was 17 and lived there on my own for about 2 years.. got sick of the weather and came back |
| 12:23.44 | blueneon` | i wish i never came back tho |
| 12:23.47 | DarKnesS_WolF | blueneon`: lol small ? |
| 12:23.49 | DarKnesS_WolF | how small :P? |
| 12:23.55 | blueneon` | baby |
| 12:24.02 | DarKnesS_WolF | ah young :P |
| 12:24.06 | blueneon` | :P |
| 12:24.33 | blueneon` | ye any way in the last year alone, i have been pickpocketed so has my wife, and we were robbed while asleep in our beds!! |
| 12:24.34 | blueneon` | :( |
| 12:24.41 | blueneon` | so ye im looking to get out of this shit hole |
| 12:24.51 | blueneon` | but not to go back to UK |
| 12:24.55 | blueneon` | so its a catch 22 |
| 12:24.56 | blueneon` | hehe |
| 12:24.59 | FTexcom | I know SA it's a pretty dangerous place |
| 12:25.24 | DarKnesS_WolF | blueneon`: how old are u !? |
| 12:25.27 | blueneon` | only ever since they did away with apartheid, and i dont mean that in a rasist way at all |
| 12:25.30 | blueneon` | 25 |
| 12:25.43 | DarKnesS_WolF | blueneon`: same here |
| 12:25.45 | blueneon` | :) |
| 12:29.54 | *** join/#asterisk glitch- (i=1045@unaffiliated/glitchz) |
| 12:30.08 | glitch- | any tips for learning SIP protocol for corporate telephony? |
| 12:30.49 | zazzizza | does anybody know what is the issue with iaxtel? i wanted to place some calls yesterday, and found it dead. i guess it's been so for long... |
| 12:31.19 | DarKnesS_WolF | glitch-: read the RFC |
| 12:32.02 | DarKnesS_WolF | zazzizza: yes iaxtel is down since yesterday as i think. not sure but i got another provider .. voip.shellshark.net |
| 12:32.02 | zazzizza | glitch-: how deep? rfcs are *the ultimate* guide, but not for the unwary |
| 12:32.44 | zazzizza | DarKnesS_WolF: do the reach the same numbers? i just wanted to call digium :-S |
| 12:33.11 | *** join/#asterisk yogurt2ungue (n=yogurt2u@24-48-231-201.fibertel.com.ar) |
| 12:33.35 | zazzizza | anyway, any of you most probably know. i bought a tdm400, is it 3.3v or 5v? cause i just have 5v slots on the box asterisk is now running |
| 12:33.44 | glitch- | DarKnesS_WolF:yes i read rfc 3261 |
| 12:33.52 | zazzizza | and i dont want to do something i may regret lol |
| 12:34.09 | DarKnesS_WolF | zazzizza: check the website |
| 12:34.31 | zazzizza | DarKnesS_WolF i did. there's nothing in it |
| 12:34.35 | zamba | how can i set priorities of codecs to use? |
| 12:34.40 | zazzizza | DarKnesS_WolF not a mention |
| 12:34.54 | DarKnesS_WolF | zamba: disallow=all allow=codecs u want |
| 12:35.09 | zamba | DarKnesS_WolF: how is the priority set? |
| 12:35.25 | DarKnesS_WolF | zamba: i think the 1st one is used 1st |
| 12:35.30 | zamba | ah, got it :) |
| 12:35.37 | DarKnesS_WolF | don't know if there is an option to use the client perfreed |
| 12:36.06 | glitch- | zazzizza:i start study two days ago, befor i use cisco callmanager |
| 12:36.59 | zazzizza | glitch-. if you want to go cisco, there's 2 courses/material i may consider. cvoice for voice over ip, and cipt for ip telephony |
| 12:37.40 | zazzizza | glitch-. they're not quite the same. cvoice is architectural voip, and ipt is call manager/poe stuff |
| 12:38.16 | inspired | hmm, when using Originate from AMI billsec is always 0, although duration is set and disposition is ANSWERED. has anyone seen this? |
| 12:40.56 | *** join/#asterisk brif8 (n=brif8@67.78.24.178) |
| 12:41.28 | glitch- | zazzizza:cool |
| 12:41.30 | glitch- | =) |
| 12:42.02 | zazzizza | glitch-. there's a cisco press book... let me see |
| 12:42.26 | brif8 | I was given the "latest" firmware for the CG-410 from Clipcomm now I CAN'T receive any calls I get "chan_sip.c:10468 handle_request_invite: Failed to authenticate user 3000" Any ideas how to fix this ? |
| 12:53.14 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 12:53.14 | *** topic/#asterisk is Asterisk: The Open Source PBX -=- http://www.asterisk.org -=- New Asterisk 1.0, 1.2, 1.2-netsec, 1.4beta, Asterisk-addons 1.2, 1.4beta, Zaptel 1.2, 1.4beta, and Libpri 1.2 releases now available! (Oct. 18, 2006) -=- Join #freepbx for freepbx/trixbox support. -=- Join #asterisk-gui to learn about the new Asterisk GUI framework |
| 12:53.48 | glitch- | i've been working with cisco since 2000 |
| 12:53.55 | glitch- | still want to get ccie :-S |
| 12:54.10 | *** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk) |
| 12:54.26 | zamba | how do i set up conference rooms in asterisk? |
| 12:54.45 | monsted | glitch-: with cisco products or do you actually work for cisco? |
| 12:55.17 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 12:58.18 | glitch- | cisco product |
| 12:59.23 | DarKnesS_WolF | Update: Asterisk 1.4 will include a 'whisper' feature as part of ChanSpy(): A third party may speak to only one of the two parties of a bridged call. i love this !!!! |
| 13:00.40 | *** join/#asterisk juanjoc (n=juanjoc@201.216.212.113) |
| 13:00.55 | monsted | glitch-: damn, i need someone to go smack the cat6500 dev team upside the head |
| 13:01.37 | zazzizza | monsted. what happened? troubles? (a) |
| 13:02.42 | monsted | zazzizza: debugging on WS-X6608-E1 cards in a cat6500 running hybrid (IOS/CatOS) is a pain |
| 13:03.20 | zazzizza | monsted. not a pain. is sick :-D |
| 13:05.40 | glitch- | 25000 SIP telephon network on openser in plan |
| 13:08.47 | *** join/#asterisk roxy_ (n=user@203.249.97-84.rev.gaoland.net) |
| 13:10.30 | roxy_ | The O'Reilly Asterisk book gives some hardware requirement depending of the number of channels. In this context is a channel a "conversation", so a conference between 5 peoples is considered 1 channel ? |
| 13:10.50 | *** join/#asterisk lorinc (n=ang@caracas-3779.adsl.interware.hu) |
| 13:10.53 | *** join/#asterisk bXi (i=bluepunk@irssi.co.uk) |
| 13:11.49 | monsted | roxy_: that usually counts as five channels |
| 13:12.00 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 13:12.18 | monsted | (five people calling the conference service) |
| 13:12.20 | roxy_ | monsted: then conversation between 2 people is 1 or 2 ? |
| 13:12.30 | monsted | 1 |
| 13:12.37 | roxy_ | monsted: thanks |
| 13:12.56 | monsted | unless it needs to transcode, i suppose it might be two :) |
| 13:13.09 | *** join/#asterisk nicchap (n=nicchap@216.209.85.2) |
| 13:14.03 | roxy_ | Is there a white paper to grab somewhere on the install of asterisk in a 1500 persons company ? |
| 13:15.29 | brif8 | Hi All I was given the "latest" firmware for the CG-410 from Clipcomm now I CAN'T receive any calls I get Failed to authenticate user 3000" when the call is made Any Ideas ?? I see that the Call comes in |
| 13:15.49 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 13:16.01 | brif8 | user 3000<sip:3523029577@10.10.10.10>;tag=3102468673 where 3000 is the PSTN/VoIP Port and 3523029577 is the phone calling in ? |
| 13:16.20 | brif8 | I thought that exten => s, covered all numbers ? |
| 13:18.03 | *** join/#asterisk MGSsancho (n=user@adsl-68-120-231-8.dsl.irvnca.pacbell.net) |
| 13:18.06 | MGSsancho | http://ask.slashdot.org/askslashdot/06/11/03/069231.shtml |
| 13:18.21 | MGSsancho | n00b should have read the voip-info site |
| 13:18.38 | zamba | can anyone help me compiling the zaptelrtc module? |
| 13:18.54 | zamba | i'm getting so many errors it's incredible :) |
| 13:20.37 | *** join/#asterisk Arnar (n=arnarb@landi.oddi.is) |
| 13:21.56 | *** join/#asterisk RyanW (n=cableguy@cor8-ppp3862.hay.dsl.connect.net.au) |
| 13:26.24 | brif8 | Did Asterisk 1.2.10 have problems with caller ID ? by any chance ? |
| 13:29.12 | brif8 | when sending caller id in the sip packet shouldn't it be From: caller_ID<sip:asterisk number@asterisk ip>;tag=..... and NOT From: asterisk_number<sip:caller_ID@asterisk ip>;tag= ?? |
| 13:30.12 | FTexcom | ha! |
| 13:30.15 | FTexcom | phone unlocked |
| 13:30.17 | *** join/#asterisk apardo (n=apardo@87.217.144.170) |
| 13:32.23 | *** join/#asterisk BASEman (i=keefejoh@gateway/web/cgi-irc/ircatwork.com/x-e82e0468329c6bef) |
| 13:35.18 | BASEman | I was told that asterisk could use any good old modem to connect your phone to it, interally; this way, my linux router would serve as an ATA. Is that correct? |
| 13:35.49 | *** join/#asterisk h3x0r4t0r (n=hex@ip68-224-236-92.lv.lv.cox.net) |
| 13:38.02 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 13:39.28 | RyanW | BASEman, i doubt its possible. |
| 13:39.44 | tzanger | BASEman: it can |
| 13:39.59 | tzanger | BASEman: it has to be one of a very specific series of WinModems |
| 13:40.09 | tzanger | do some research on wcfxo |
| 13:40.25 | tzanger | that is for an FXO port though, I am not aware of any WinModem that Asterisk supports for FXS. |
| 13:42.50 | BASEman | tzanger: I am not sure to follow you... You mean, my good old USR won't help, right? |
| 13:43.04 | tzanger | no |
| 13:43.23 | tzanger | it must be one of a specific series of either Motorola or Intel WinModems, IIRC (It's been qutie some time since I've been involved in this) |
| 13:46.00 | *** join/#asterisk skirmisha (n=vk@a82-93-113-154.adsl.xs4all.nl) |
| 13:46.05 | skirmisha | hello guys |
| 13:46.27 | skirmisha | does anyone know how can i redirect calls coming from sip to another sip provider? |
| 13:46.40 | BASEman | tzanger, but finding one of these will surely cost me less than an ATA and offer me more possibilities, right? Or would you not recommand this way of doing it? |
| 13:47.30 | tzanger | depends. some work well, some don't work well. Interfacing to the PSTN is always a crapshoot because there are so many variables, and the quality of the hybrid plays a key role. |
| 13:47.47 | tzanger | poor quality hybrid = audio issues, echo, problems detecting digits, you name it |
| 13:52.31 | *** join/#asterisk seele_ (n=seele@208.35.117.195) |
| 13:52.36 | seele_ | hello ! |
| 13:53.11 | *** join/#asterisk lupino3 (n=lupino3@217-133-98-121.b2b.tiscali.it) |
| 13:53.42 | seele_ | I'm seeing "channel.c: Nobody there, continuing..." in the asterisk full.log. This error is repeated 20+ times per second when it occurs .... please help ! |
| 13:54.30 | anonymouz666 | I saw this once |
| 13:55.15 | anonymouz666 | It was because I applied a patch |
| 13:57.02 | skirmisha | does anyone know how can i redirect calls coming from sip to another sip provider? |
| 13:57.04 | zazzizza | Do not worry. Your TDM400P is dual slotted to work in both a 3.3V and 5.0V PCI slot that is PCI 2.2 compliant. Have a great day! |
| 13:57.18 | zazzizza | :-D at least a good one |
| 13:57.19 | DarKnesS_WolF | zazzizza: hehe perfect answer ;-) |
| 13:57.42 | zazzizza | indeed! better impossible! ;-) |
| 13:57.55 | DarKnesS_WolF | zazzizza: have a good day setting it up;-) |
| 13:57.58 | DarKnesS_WolF | skirmisha: more information ? |
| 13:58.15 | seele_ | skirmisha, add sip provider like a trunk |
| 13:58.53 | seele_ | skirmisha, http://www.voip-info.org/wiki/view/Asterisk+settings+Net2phone |
| 14:00.09 | skirmisha | i did |
| 14:00.26 | skirmisha | u don't get me |
| 14:00.29 | *** join/#asterisk [hC] (n=hardcore@66.119.187.139) |
| 14:00.36 | skirmisha | calls coming from another sip server to asterisk |
| 14:00.46 | skirmisha | i want to redirect all calls to third party sip gw |
| 14:00.55 | skirmisha | all incoming calls |
| 14:01.06 | DarKnesS_WolF | skirmisha: so u want all incoming calls gose to another server |
| 14:01.13 | skirmisha | yes |
| 14:01.24 | DarKnesS_WolF | add context to the sip proverder where the calls are comming |
| 14:01.25 | DarKnesS_WolF | and do |
| 14:01.32 | DarKnesS_WolF | in that extention |
| 14:01.43 | DarKnesS_WolF | exten => s,1,Dial(SIP/theoterhserver) |
| 14:01.59 | DarKnesS_WolF | i hope i got it right and and i hope it will work adn do what u want . " there is no warranty :P |
| 14:02.05 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:02.13 | skirmisha | thanks |
| 14:02.31 | *** join/#asterisk Assid (i=assid@59.183.27.142) |
| 14:02.43 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 14:02.50 | BASEman | tzanger, thank you, I'll investigate the topic more when I have time... |
| 14:02.56 | b11d|bbl | morning lads |
| 14:03.20 | *** join/#asterisk dasenjo (n=dasenjo@201.228.128.10) |
| 14:04.04 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:04.45 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 14:05.28 | *** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com) |
| 14:05.40 | zazzizza | DarKnesS_WolF thanks! i have already an asterisk box trunking calls via a c2600 + 2 clone fxs to get in&out of the existing pbx, so putting this to work will be much easier ;-) |
| 14:05.45 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 14:05.59 | *** join/#asterisk Ebola (n=Ebola@host86-136-130-111.range86-136.btcentralplus.com) |
| 14:07.47 | skirmisha | DarKnesS_WolF there is one more thing, all incoming calls are not coming from phones registered with asterisk |
| 14:08.22 | brif8 | SIP Header From: should it not read "From: Caller_ID<sip:extension@Asterisk_IP_Address>;tag=...." Yet mine is showing "From: extension<sip:callerID@Asterisk_IP_Address>;tag=...." and thus resulting in Failed to Authenticate user ERROR ? |
| 14:08.40 | brif8 | is this a 1.2.10 Bug or what ? |
| 14:09.07 | *** join/#asterisk A-Tuin|work (n=A-Tuin@212.41.185.81) |
| 14:10.26 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 14:11.36 | DarKnesS_WolF | skirmisha: so ? |
| 14:13.46 | skirmisha | will this do the job |
| 14:14.22 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 14:16.41 | skirmisha | i am not sure how asterisk will know where to route the calls |
| 14:17.01 | DarKnesS_WolF | skirmisha: routeing hte calls called a dialplan |
| 14:17.06 | DarKnesS_WolF | u do ur own dialplan |
| 14:17.13 | *** part/#asterisk brif8 (n=brif8@67.78.24.178) |
| 14:17.22 | skirmisha | yes that's ok |
| 14:17.36 | *** join/#asterisk dil (n=sdf@227-174.CLCOM.cgocable.ca) |
| 14:18.12 | dil | hi |
| 14:18.26 | dil | where can i get business edition support? |
| 14:18.51 | zazzizza | dil be@digium.com -> www.digium.com -> contact |
| 14:20.38 | bXi | can somebody recommend a way of capturing sip data before it enters asterisk? |
| 14:20.51 | bXi | trying to troubleshoot some issues on my isdn line |
| 14:20.57 | *** join/#asterisk p1p (n=p1p@mail.comp911.com) |
| 14:20.58 | Rhizome | tshark port 5060 |
| 14:21.00 | skirmisha | bXi what issues |
| 14:21.08 | bXi | skirmisha: basicly |
| 14:21.17 | bXi | some sip phones produce stuttering on the isdn lines |
| 14:22.14 | bXi | it was really bad in the beginning |
| 14:22.55 | bXi | but i fixxed that by putting the RTP payload size from 0.03 (standard setting on SPA3000) to 0.020 (standard according to manual AND rfc) |
| 14:24.48 | bXi | music on hold works perfectly on the isdn |
| 14:24.57 | bXi | so its something with communicating towards asterisk |
| 14:24.59 | skirmisha | ahh that's most probably is codec problem or bandwidth |
| 14:25.41 | skirmisha | does your asterisk have isdn cards? |
| 14:25.44 | bXi | yeah |
| 14:25.54 | bXi | some el cheapo isdn card |
| 14:25.56 | bXi | with the visdn driver |
| 14:26.13 | skirmisha | what codec do u use? |
| 14:26.27 | bXi | on my sipphones i have G711u or G711a |
| 14:26.32 | bXi | both produce the same effects |
| 14:26.56 | skirmisha | this is only on incoming or both incoming and outgoing |
| 14:27.29 | bXi | when i call to a cell phone the cell phones has most issues with stuttering |
| 14:28.12 | skirmisha | how many concurrent call do u have? |
| 14:28.23 | bXi | concurrent == at the same time right? |
| 14:28.31 | skirmisha | yes |
| 14:28.58 | bXi | no other calls going on at the same time |
| 14:29.04 | bXi | its in testing stage now |
| 14:29.07 | skirmisha | spa3000 is sipura right? |
| 14:29.11 | bXi | yeah |
| 14:29.18 | skirmisha | can u test with g729 codec |
| 14:29.22 | *** join/#asterisk gaspiz (n=gaspiz@86.35.34.63) |
| 14:29.37 | bXi | i think asterisk doesnt have g729 but i'll try |
| 14:30.09 | skirmisha | it relays g729 |
| 14:30.19 | *** join/#asterisk sb_mx (n=sb_mx@200.78.229.18) |
| 14:30.20 | skirmisha | if both parties support it |
| 14:30.49 | bXi | is g729a the same as g729 ? |
| 14:30.52 | FTexcom | too see if you have g729 support enabled do this on the cli...show translation...if you see numbers on the g729 line...it's avaliable |
| 14:31.18 | gaspiz | Hi, in asterisk 1.2.12 when from dead-agi you playback a file or wait for a digit and meanwhile the user hangs up the scripts halt. Did anyone experience this issue |
| 14:31.19 | skirmisha | bXi try with g729 only |
| 14:31.26 | bXi | g729 - - - - - - - - - - - |
| 14:31.48 | FTexcom | it's not available |
| 14:32.05 | FTexcom | <PROTECTED> |
| 14:32.12 | FTexcom | that's mine |
| 14:32.19 | bXi | wasnt g729 the codec you should pay for? |
| 14:32.26 | FTexcom | yes it is |
| 14:33.08 | bXi | hmmm i do have g726 according to show translation |
| 14:33.32 | bXi | g729 doesnt work on the spa3000 |
| 14:33.37 | *** join/#asterisk oa (n=oal@62.61.133.90.generic-hostname.arrownet.dk) |
| 15:02.28 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 15:02.28 | *** topic/#asterisk is Asterisk: The Open Source PBX -=- http://www.asterisk.org -=- New Asterisk 1.0, 1.2, 1.2-netsec, 1.4beta, Asterisk-addons 1.2, 1.4beta, Zaptel 1.2, 1.4beta, and Libpri 1.2 releases now available! (Oct. 18, 2006) -=- Join #freepbx for freepbx/trixbox support. -=- Join #asterisk-gui to learn about the new Asterisk GUI framework |
| 15:03.03 | *** join/#asterisk GaVak (n=denniso@adsl-074-228-124-003.sip.sav.bellsouth.net) |
| 15:04.14 | MGSsancho | id reccomend the asterisk buisisness version from digium. and i know of a few guy who work for them. im just wondering if they are awake andhave their computers on |
| 15:04.33 | MGSsancho | damn i make no sense when i dont sleep for a few days. |
| 15:04.37 | *** join/#asterisk _polto_ (n=polto@d83-189-153-155.cust.tele2.ch) |
| 15:04.42 | _polto_ | hello all |
| 15:05.04 | MGSsancho | hi |
| 15:05.37 | _polto_ | do somebody know if i can use IAXy as a trunk to my analog line ? if yes, how to configure it ? |
| 15:07.40 | *** join/#asterisk foxxtrot (n=craig@c-67-185-55-194.hsd1.wa.comcast.net) |
| 15:08.49 | GaVak | When I set up my external SIP phone users and set externip= for their phones, the system starts treating their IP address as their user name. (Instead of the username set in the phone.) Is there a way to avoid this behavior? |
| 15:12.15 | *** join/#asterisk DrkShdw (n=scorpio@unaffiliated/drkshdw) |
| 15:12.35 | *** join/#asterisk Chicago (n=Chicago@c-67-186-94-7.hsd1.in.comcast.net) |
| 15:12.58 | Chicago | What resources are available for free incoming lines/ free outgoing lines? |
| 15:16.05 | *** join/#asterisk kuto (n=ftft@125.60.241.24) |
| 15:16.19 | *** join/#asterisk amdtech (n=amd011@ss-5-100.shsu.edu) |
| 15:19.16 | EFI-VJ | free incomming sip voicestick free outgoing voipstunt |
| 15:21.08 | b11d | how does that free stuff work? wheres their income? |
| 15:24.40 | zamba | when trying to load ztdummyi get the following error: ztdummy: no version for "struct_module" found: kernel tainted. |
| 15:26.53 | *** join/#asterisk pifiu (n=someone@216.5.79.1) |
| 15:27.15 | pifiu | morning everyone! |
| 15:27.20 | b11d | morning |
| 15:27.30 | GaVak | morning. |
| 15:27.44 | b11d | pifiu.. i missed you at the gang bang last night.. where were you/ |
| 15:28.06 | *** join/#asterisk saftsack (n=saftsack@pD9E056A0.dip.t-dialin.net) |
| 15:28.09 | GaVak | Zamba: http://www.clarkconnect.com/forums/showthreaded.php?Cat=0&Number=73539&page=&vc=1 suggests that you don't have the right links set for your kernel source. |
| 15:28.20 | zamba | yeah, but i do :p |
| 15:28.37 | zamba | i just rechecked that |
| 15:30.31 | pifiu | lol |
| 15:30.37 | pifiu | who did you guys bang? |
| 15:30.37 | pifiu | lmao |
| 15:30.57 | *** join/#asterisk jtexter3 (n=jtexter3@COX-66-210-197-46-static.coxinet.net) |
| 15:31.11 | b11d | well.. jenny for sure.. but there were two others there that i've never seen before |
| 15:31.12 | jtexter3 | Anyone here having DTMF issues with a TE411P and Zaptel 1.2.10? |
| 15:31.13 | b11d | they were asking for oyu |
| 15:31.18 | b11d | but I was like.. i havent seem him |
| 15:31.54 | *** join/#asterisk cbm11211 (n=Administ@66.250.98.174) |
| 15:32.05 | *** part/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 15:33.02 | *** join/#asterisk Strom_C (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 15:33.14 | *** join/#asterisk juanjoc (n=juanjoc@201.216.212.113) |
| 15:33.28 | *** join/#asterisk jcims (n=jcims@rrcs-24-172-217-2.central.biz.rr.com) |
| 15:33.32 | *** join/#asterisk qdk (n=qdk@0x535eae17.boanxx9.adsl-dhcp.tele.dk) |
| 15:33.33 | *** join/#asterisk skirmisha (n=vk@213.144.225.179) |
| 15:34.54 | jcims | hey folks, anyone know why my cdr's would show the src as 'Anonymous' for all calls from an internal phone to the outside? |
| 15:35.02 | pifiu | shit you should have called me! |
| 15:35.07 | pifiu | you know i got the polycom's! |
| 15:35.17 | b11d | sorry man.. next time I will for sure! |
| 15:35.35 | pifiu | lol good |
| 15:35.41 | b11d | just bring that funny sock you wore that last time.. |
| 15:35.44 | b11d | it cracked everyone up |
| 15:35.48 | pifiu | lol ok? |
| 15:35.50 | b11d | :P |
| 15:36.00 | *** join/#asterisk _alex_mx_ (n=alex@200.78.229.18) |
| 15:36.14 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 15:37.34 | jcims | any fonality folks on? |
| 15:38.10 | Qwell[] | jcims: Surely they have their own support channels |
| 15:38.26 | jcims | of course |
| 15:38.41 | jcims | just trying to help troubleshoot something for them |
| 15:38.53 | b11d | for their support people? |
| 15:38.55 | b11d | wow |
| 15:38.56 | Qwell[] | best to go through their support dept |
| 15:38.59 | jcims | i am |
| 15:39.01 | Qwell[] | and yeah... |
| 15:39.05 | Qwell[] | b11d: I agree :p |
| 15:39.07 | b11d | :) |
| 15:39.15 | jcims | usually they are very good, this is just an odd thing |
| 15:39.47 | jcims | i've googled it quite a bit, nobody here has responded to the issue, it's just unusual |
| 15:40.03 | Qwell[] | because nobody here is able to support fonality |
| 15:40.09 | Qwell[] | the whole "closed source" thing |
| 15:40.36 | jcims | ok... didn't think it was that much of a deviation from the main product |
| 15:42.48 | GaVak | Why would my * server read the Userid for phones inside the firewall, but for the phones with externip=xxx.xxx.xxx.xxx set, It reads the incoming IP as the user name? |
| 15:42.53 | GaVak | chan_sip.c:11131 handle_request_register: Registration from '<sip:4103@savpbx.tpisoft.com>' failed for '74.228.124.3' - Username/auth name mismatch |
| 15:43.15 | GaVak | If I remove the externip= entry, it grabs the 4103. |
| 15:44.15 | GaVak | I would use the ips as the user name, but they are dynamic.... |
| 15:44.36 | *** part/#asterisk jcims (n=jcims@rrcs-24-172-217-2.central.biz.rr.com) |
| 15:48.32 | skirmisha | anyone who knows good sip redirect server? |
| 15:48.47 | ManxPower | GaVak: perhaps your localnet= is wrong? |
| 15:49.22 | ManxPower | GaVak: I assume you have a [4103] section of sip.conf? |
| 15:49.23 | jeremy_g | :D |
| 15:49.35 | jeremy_g | what is callerid? is it what is in the From: field |
| 15:49.39 | jeremy_g | :P |
| 15:49.51 | ManxPower | skirmisha: you mean SIP Proxy. Try SER "SIP Express Router" |
| 15:50.01 | jeremy_g | what is display name, caller presentation text, |
| 15:50.20 | jeremy_g | skirmisha:dont try that. he is misguiding you. |
| 15:50.24 | ManxPower | jeremy_g: no idea, I always override the callerid info in sip.conf |
| 15:50.27 | jeremy_g | :P |
| 15:51.46 | *** join/#asterisk Katty (n=copirite@64.82.199.210) |
| 15:51.53 | Katty | hihi |
| 15:51.57 | jeremy_g | haha |
| 15:52.02 | Qwell[] | hoho? |
| 15:52.07 | jeremy_g | aOOOOOO |
| 15:52.27 | jeremy_g | this kinda signalling is way better than sip |
| 15:52.35 | jeremy_g | builtin sigcomp |
| 15:52.56 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 15:52.56 | ManxPower | Good morning, Katty |
| 15:52.57 | *** join/#asterisk DrkShdw (n=scorpio@unaffiliated/drkshdw) |
| 15:53.19 | Katty | hey Manx (= |
| 15:53.46 | jeremy_g | hey Katty, u got cam n pics lolz |
| 15:53.53 | Katty | ... |
| 15:53.58 | file | Katty: I see you! |
| 15:54.02 | Katty | file: there you are! |
| 15:54.10 | file | I also _heard_ you a short time ago |
| 15:54.16 | Katty | file: can you believe i had 12 enteries in my extensions.conf and you were the only one that answers still )= |
| 15:54.23 | file | Katty: crazy that |
| 15:54.26 | Katty | i know. |
| 15:54.29 | Katty | they just /died/ |
| 15:54.32 | Katty | no voicemail |
| 15:54.35 | Katty | NOTHING. |
| 15:54.43 | file | :( |
| 15:55.03 | *** join/#asterisk SwK (n=Silik0nJ@208.44.30.242) |
| 15:55.07 | Katty | file: we're nearly almost moved now :> |
| 15:55.15 | Katty | file: all the cat5 is ran... |
| 15:55.21 | Katty | file: and two of our servers are up and going. |
| 15:55.26 | file | yay |
| 15:55.33 | Katty | SwK: hey you (= |
| 15:55.50 | Katty | file: do you think there'd be any drawback in putting a demo server on a dmz port.. |
| 15:56.00 | Katty | file: and then taking phones to..wherever. |
| 15:56.09 | file | yes, it will catch on fire |
| 15:56.13 | *** join/#asterisk svenna (n=svenna@p548D42C7.dip0.t-ipconnect.de) |
| 15:56.15 | Katty | whyfor? |
| 15:56.50 | file | Neutron Moon Rays will travel through the DMZ port |
| 15:56.58 | Katty | i see i see, well that's quite a pickle. |
| 15:57.06 | Chicago | Any gurus care to nurture my asterisk n00bishness? I am on gentoo and have asterisk compiled... but I have a major problem. I was planning on using some US Robotics voice modems which I have learned are absolutely not supported. |
| 15:57.14 | Katty | i shall have to put my neutron ray filtermication system on it first. |
| 15:57.41 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 15:57.42 | Chicago | In my environment (home) I want to route and record/monitor calls from the home phone line and the cell phone... I was planning on having the lines passthrough asterisk and have asterisk dial out to reach me for all my incoming calls. |
| 15:57.55 | Katty | file: can you find me an arrow? |
| 15:58.03 | Chicago | And then possibly get another verizon wireless phone and bind it to my asterisk to ensure all the calls to my mobile are in-network. |
| 15:58.08 | Katty | file: i want to put a company wide directory on the phone |
| 15:58.12 | Chicago | Can somebody discuss this with me please. |
| 15:58.37 | Katty | file: course i don't know where to start )= |
| 15:58.44 | Katty | file: documentation is always good! |
| 15:58.48 | JunK-Y | hey katty, long time! |
| 15:58.53 | Katty | JunK-Y: hey hun! |
| 15:58.59 | JunK-Y | whats up? |
| 15:59.02 | Katty | JunK-Y: yeah, we moved... |
| 15:59.10 | JunK-Y | where? |
| 15:59.12 | Katty | JunK-Y: been doing dirty work like running cable )= |
| 15:59.19 | Katty | JunK-Y: just 10 miles east |
| 15:59.24 | monsted | hmm, playing with least-cost call routing using various providers all over the world would be so much more interesting if i didn't have a free phone line already ;) |
| 15:59.45 | *** join/#asterisk _santiago_ (n=santiago@debian/developer/santiago) |
| 15:59.51 | Katty | JunK-Y: i finally got a desk again :P |
| 16:00.30 | JunK-Y | yay :) |
| 16:00.38 | *** join/#asterisk rbd (n=rbd@adsl-074-229-183-112.sip.rmo.bellsouth.net) |
| 16:01.43 | rbd | anyone on here know the SCCP (Cisco Skinny) signalling protocol? I was wondering if it supports port address translation (i.e. if in effect I could have multiple SCCP phone connections looking to be from an endpoint with a single IP) |
| 16:02.09 | *** join/#asterisk santiago (n=santiago@debian/developer/santiago) |
| 16:04.02 | *** join/#asterisk Katty (n=copirite@64.82.199.210) |
| 16:06.52 | X-Rob | anyone want any alison recordings done? |
| 16:07.33 | b11d | she'll just record anything for people eh |
| 16:08.20 | *** join/#asterisk brc_ (n=brc___@pdpc/supporter/basic/brc) |
| 16:11.30 | *** join/#asterisk nortex (n=breeves@snapper.titanspecialties.com) |
| 16:13.22 | *** join/#asterisk alerios (n=alerios@190.24.97.148) |
| 16:13.36 | roxy_ | is there a place where I can find doc on ~1500 users asterisk system ? |
| 16:15.27 | *** join/#asterisk andresmujica (n=andresmu@201.245.231.252) |
| 16:15.58 | andresmujica | hello, |
| 16:16.30 | zazzizza | roxy_ search voip-info.org. depending on provisioning you should be able to asset the reqs |
| 16:16.51 | nortex | roxy_, You mean like examples or just a "is it possible" |
| 16:17.42 | *** join/#asterisk brc_ (n=brc___@pdpc/supporter/basic/brc) |
| 16:17.55 | andresmujica | anyone knows if the iaxy s101 would or could support an additional codec different form g726 an alaw? |
| 16:18.11 | ManxPower | andresmujica: it does not and will not. |
| 16:18.29 | ManxPower | andresmujica: it does not have enough processing power to support other codec |
| 16:18.31 | ManxPower | s |
| 16:18.40 | andresmujica | yeap that's the question. |
| 16:18.44 | andresmujica | ok thanks |
| 16:18.52 | ManxPower | It doesn't even support DNS |
| 16:19.04 | andresmujica | yeap that's a bad thing. |
| 16:19.43 | andresmujica | the codec issue is a problem also for a remote deployment over poor bandwidth links... |
| 16:19.55 | andresmujica | but anyway.. thanks |
| 16:19.58 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 16:20.40 | Katty | mister fender (= |
| 16:26.05 | *** join/#asterisk _Vile (i=vile@198.175.14.242) |
| 16:29.35 | *** join/#asterisk SplasPood (n=jwb@nat-loopback.lga4.us.corp.voxel.net) |
| 16:31.16 | [TK]D-Fender | Katty: Mew. |
| 16:37.36 | *** join/#asterisk Splat (n=Splat@220-253-134-37.TAS.netspace.net.au) |
| 16:39.17 | *** join/#asterisk ruskie (n=ruskie@sourcemage/mage/ruskie) |
| 16:41.17 | *** join/#asterisk Splas (n=jwb@nat-loopback.lga4.us.corp.voxel.net) |
| 16:42.13 | *** join/#asterisk droops (n=droops@adsl-074-245-001-031.sip.jan.bellsouth.net) |
| 16:42.15 | *** join/#asterisk hmmhesays (n=hmmhesay@24-117-135-28.cpe.cableone.net) |
| 16:42.43 | hmmhesays | being sick really sucks |
| 16:42.50 | Katty | hey hun |
| 16:43.05 | hmmhesays | Hey Katty |
| 16:43.12 | hmmhesays | I've been doing nothing but laying in bed for days |
| 16:43.24 | Katty | aww. |
| 16:43.40 | hmmhesays | I woke up in a hug puddle of sweat twice last night |
| 16:44.22 | Katty | eww. |
| 16:45.14 | Katty | you really should get your tail to the doctor. |
| 16:46.06 | *** join/#asterisk Mw3 (i=mw3@national.t-error.hu) |
| 16:46.12 | hmmhesays | no, just a ND sore throat |
| 16:46.48 | roxy_ | zazzizza: thanks |
| 16:47.05 | Katty | hmmhesays: well maybe they can do /something/ |
| 16:47.54 | roxy_ | nortex: example. Needed know-how and stuff. We need to evaluate to maybe switch in a 2~3 years. |
| 16:49.57 | hmmhesays | yeah they'll charge me a bunch of money and send me on my way |
| 16:51.42 | b11d | come over here to Hibbing |
| 16:51.45 | b11d | we'll knock it out with some booze |
| 16:51.54 | *** join/#asterisk krondorl (n=krondorl@acid.auricnet.ca) |
| 16:52.06 | b11d | I can set you up with a Northern Ontario remedy for a North Dakota Sore Throat |
| 16:55.11 | hmmhesays | oh yeah? |
| 16:55.11 | hmmhesays | haha |
| 16:55.17 | hmmhesays | no booze |
| 16:55.23 | hmmhesays | no good for this |
| 16:56.20 | krondorl | I know this is a newbie question but then again I am kinda new at this.. In the SIP.CONF file do you have to have context=default in the [general] section? if not there is it assumed as default? |
| 16:56.21 | hmmhesays | look at the notes in sip.conf |
| 16:56.21 | b11d | actually the worst sore throat I've ever had was fixed in a jiff with some brandy in some hot lemon tea.. |
| 16:56.21 | b11d | it was gone instantaneously |
| 16:56.22 | b11d | and I couldnt swallow before that |
| 16:56.29 | krondorl | the are no longer there.. I didn't set this one up and the guy that did got rid of all the notes. |
| 16:57.13 | krondorl | also the asteriskTFOT.pdf doesn't help explain it. |
| 16:58.13 | zazzizza | krondorl: put context=<something> |
| 16:58.44 | zazzizza | and get the sources, there you have the sample *.confs |
| 16:58.47 | mercestes | On the function "ChanIsAvail()" I get an exit non-0 when trying ot do a ChanIsAvail on a phone in a conference. Does ChanIsAvail support detection of a phone active in a conference?? |
| 16:59.31 | krondorl | zazzizza, :) I understand how to enter the line, I just want to know if it has to be there or not and if not does it default to default. |
| 17:00.07 | zazzizza | it has to be there |
| 17:00.13 | zazzizza | otherwise it's =default |
| 17:01.00 | zazzizza | it has to be there if you want the context to be something else (the "default" context, not context "default) |
| 17:01.00 | ManxPower | mercestes: the exit code means nothing. |
| 17:01.42 | mercestes | ManxPower: Ok, it's not actually executing any of hte code I've written either..:( |
| 17:02.01 | krondorl | zazzizza, Thanks, that's what I thought but wasn't sure.. |
| 17:02.09 | zazzizza | krondorl: ;-) |
| 17:02.52 | mercestes | It does work correctly if a phone is available I would like to mention. I'm running Asterisk 1.2.13 with Polycom phones (Firmware 1.6.6) |
| 17:03.26 | ManxPower | mercestes: look at the variables set by ChanIsAvail |
| 17:04.07 | ManxPower | Specifically ${AVAILSTATUS} |
| 17:04.22 | krondorl | Anyone know good IAX phones other than GNET? |
| 17:05.17 | saftsack | does someone of you has a fritzbox? |
| 17:05.31 | b11d | I cant get "The Teacher" by Jethro Tull out of my mind!!!!! |
| 17:06.22 | mercestes | ManxPower: My first line of code is ChanIsAvail(sip/device) My second line is NoOp($arg1} has a status of ${AvailStatus}) for debugging purposes. |
| 17:06.40 | mercestes | ManxPower: It is exitting non-zero before it hits that NoOp. |
| 17:06.47 | mercestes | but it works fine if that phone is not in a conference. |
| 17:07.46 | ManxPower | mercestes: chanisavail should not care about what the phone is connected to, just that it is in use. |
| 17:08.06 | ManxPower | mercestes: put that part of the dialplan on pastebin.ca |
| 17:10.23 | *** join/#asterisk topping (n=topping@207.47.6.182.static.nextweb.net) |
| 17:10.25 | mercestes | ManxPower: Sure thing...just a sec. |
| 17:12.35 | *** join/#asterisk syzygyBSD (n=chatzill@poplar.matraex.com) |
| 17:13.26 | syzygyBSD | for sip.conf, if I want to bind to multiple IPs do I have multiple bindaddr lines or seperate the ips by a comma on one line? |
| 17:14.08 | ManxPower | syzygyBSD: not having a bindaddr will bind to all IPS on the system |
| 17:14.18 | ManxPower | let the standard routing stuff pick which source IP |
| 17:14.21 | syzygyBSD | well, what if I dont' want to bind to all of them |
| 17:14.40 | syzygyBSD | i could also enter 0.0.0.0 if I wanted to bind to all of them |
| 17:17.01 | *** join/#asterisk Ebola (n=Ebola@host86-136-130-111.range86-136.btcentralplus.com) |
| 17:17.39 | *** join/#asterisk hohum (n=dcorbe@jomama.interceltelecoms.net) |
| 17:18.57 | b11d | MY ONE WAY VOICE PROBLEM IS FIXED!!!!! |
| 17:19.00 | b11d | HAHAHAHAAHAHAHAH JOY!!! |
| 17:19.06 | ManxPower | b11d: what was the cause? |
| 17:19.11 | b11d | ip route 0.0.0.0 0.0.0.0 10.0.5.254 |
| 17:19.36 | b11d | that was it |
| 17:19.38 | b11d | fucking routing issue |
| 17:19.44 | ManxPower | Ah. |
| 17:21.03 | syzygyBSD | lol.. I have to deal with a routing setup soon too |
| 17:22.19 | b11d | oh yeah |
| 17:22.26 | b11d | i never thought of it.. |
| 17:22.32 | b11d | when I saw SIP working, i ruled it out.. |
| 17:22.36 | ManxPower | obviously neither did anyone else |
| 17:22.36 | b11d | but yeah.. it was a routing issue |
| 17:23.00 | *** join/#asterisk DasTech (n=DasTech@ppp-71-128-113-209.dsl.irvnca.pacbell.net) |
| 17:23.04 | DasTech | morning |
| 17:23.12 | b11d | morning |
| 17:23.32 | b11d | well, yeah.. you guys did ask if it was a routing issue, but when we saw that ping was working, and SIP was working.. who would have |
| 17:23.33 | b11d | thought. |
| 17:23.35 | syzygyBSD | mine is a bit of a pain in the ass though, multiple network interfaces each with a connection to the internet, also acting as our office router |
| 17:23.54 | DasTech | any major issues found in 1.4-beta 3 |
| 17:23.54 | *** join/#asterisk hoobastooba (n=ckwall@63.149.122.93) |
| 17:24.11 | DasTech | or is there going to be a rc1 soon |
| 17:24.19 | syzygyBSD | 3 subnets, 2 gateways... bah, later |
| 17:24.28 | DasTech | I have 1.4-beta 3 and thus far had no issues |
| 17:24.57 | Qwell[] | DasTech: no rc |
| 17:25.03 | Qwell[] | DasTech: probably 1 more beta, then release |
| 17:25.05 | b11d | Qwell, you had it all along man |
| 17:25.05 | mercestes | ManxPower: Sorry, on a tech call...let me resolve htis and i'll post my code for you. |
| 17:25.12 | DasTech | ok |
| 17:25.12 | Qwell[] | b11d: wrong port? |
| 17:25.14 | *** join/#asterisk pmnke (n=perlmonk@hubert.perlmonkee.com) |
| 17:25.23 | b11d | routing issue.. so . kind of. |
| 17:25.35 | b11d | it just couldnt reach the phone.. |
| 17:26.07 | DasTech | qwell and major bugs in beta3 that have been documented ? |
| 17:26.16 | Qwell[] | DasTech: bugs.digium.com |
| 17:26.21 | pmnke | Here in oregon, we have to use 10 digit dialing for local numbers (no leading 1 as in for long distance numbers) - so I add some leading x's to my [trunklocal] exten line, but now it catches long distance numbers and they thusly are not dialed correctly... does anyone have any idea on how to make this not happen? |
| 17:26.22 | ManxPower | Ah. I NEVER use Linux as a router. |
| 17:32.23 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 17:32.23 | *** topic/#asterisk is Asterisk: The Open Source PBX -=- http://www.asterisk.org -=- New Asterisk 1.0, 1.2, 1.2-netsec, 1.4beta, Asterisk-addons 1.2, 1.4beta, Zaptel 1.2, 1.4beta, and Libpri 1.2 releases now available! (Oct. 18, 2006) -=- Join #freepbx for freepbx/trixbox support. -=- Join #asterisk-gui to learn about the new Asterisk GUI framework |
| 17:32.36 | hmmhesays | ~hmmhesays |
| 17:32.38 | jbot | well, hmmhesays is not really here... |
| 17:32.48 | ManxPower | that picture does not match the model number |
| 17:32.54 | hmmhesays | someone needs to put something better in there |
| 17:33.43 | *** part/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 17:33.45 | b11d | yeah |
| 17:33.47 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 17:33.51 | b11d | damn marketing depts.. |
| 17:33.54 | b11d | i hate them |
| 17:34.15 | hoobastooba | is there no way to limit inbound and outbound calls individually? instead of limiting all calls? |
| 17:34.42 | *** join/#asterisk fx0 (n=fx0@cypher.punk.net) |
| 17:34.47 | roxy_ | ManxPower: I got the difference between FXS/FXO. So this connect to 2FXO and 2FXS: http://www.thevoipconnection.com/store/customer/product.php?productid=16185 |
| 17:34.49 | jeremy_g | why do i keep getting these messages EBUG[30557] chan_sip.c: = No match Their Call ID: 25877@192.168.0.84 Their Tag 3620 Our tag: as65a3af4a |
| 17:34.49 | jeremy_g | Nov 3 18:02:36 DEBUG[30557] chan_sip.c: = No match Their Call ID: 37b436035e10bb77494ed0824179fb68@192.168.0.2 Their Tag as24c9f703 Our tag: as486a |
| 17:34.59 | *** join/#asterisk dostidilse (i=Shashu@203.200.75.4) |
| 17:35.30 | b11d | you'd want the tdm22b then i think roxy_.. |
| 17:35.33 | ManxPower | jeremy_g: because you are running in debug mode. |
| 17:35.33 | b11d | 2fxo and 2fxs |
| 17:35.47 | dostidilse | can anyone help me out in finding a good training manuall for newb on astersik |
| 17:35.50 | *** join/#asterisk SwK_ (n=Silik0nJ@208-44-30-242.dia.static.qwest.net) |
| 17:36.06 | b11d | dostidilse.. voip-info.org and sitting right here |
| 17:36.07 | b11d | best manual |
| 17:36.15 | hoobastooba | dostidilse: you want how to build it? or use it? |
| 17:36.16 | b11d | if you're new.. hang out |
| 17:36.27 | roxy_ | dostidilse: I am reading the O'Reilly, it is good. |
| 17:36.34 | hoobastooba | yes |
| 17:36.47 | roxy_ | and a safari book |
| 17:36.56 | hoobastooba | that is a great start. |
| 17:37.03 | dostidilse | hmm |
| 17:37.06 | dostidilse | thanks for the input |
| 17:37.09 | hoobastooba | next what you need is to try it all out hard knocks |
| 17:37.23 | b11d | yeah.. dont expect to make it work right the first time either.. |
| 17:37.31 | b11d | you're going to have to fux with it for awhile.. |
| 17:37.32 | b11d | learn it.. |
| 17:37.35 | hoobastooba | you will find that each implementation has its own issues that are not documented anywhere. |
| 17:37.57 | dostidilse | i will try .. else i will use this support channel |
| 17:38.03 | hmmhesays | so true |
| 17:38.07 | b11d | you will learn an immense amount of info sitting in here |
| 17:38.33 | hmmhesays | tinkering |
| 17:38.37 | hoobastooba | if you read the books, and the wiki... you will be able to formulate the questions you need to ask here. |
| 17:38.37 | b11d | also by listening to others.. |
| 17:38.43 | jeremy_g | b11d:hows it going dude |
| 17:39.18 | b11d | its going great man.. you? |
| 17:40.27 | *** join/#asterisk IgorG (n=FeedomPa@host-195-162-53-193.pppoe.omsknet.ru) |
| 17:42.06 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 17:42.06 | *** topic/#asterisk is Asterisk: The Open Source PBX -=- http://www.asterisk.org -=- New Asterisk 1.0, 1.2, 1.2-netsec, 1.4beta, Asterisk-addons 1.2, 1.4beta, Zaptel 1.2, 1.4beta, and Libpri 1.2 releases now available! (Oct. 18, 2006) -=- Join #freepbx for freepbx/trixbox support. -=- Join #asterisk-gui to learn about the new Asterisk GUI framework |
| 17:42.06 | roxy_ | I am gonna try to reformulate. If I get the TDM400P, I don't get anything but can add 4 module to it. The price of it depends if it is FXS or FXO and how many I need . Right ? |
| 17:42.06 | b11d | right. |
| 17:42.06 | hmmhesays | great movie |
| 17:42.06 | monsted | right |
| 17:42.07 | hmmhesays | that deserved a sequel |
| 17:42.19 | b11d | yeah.. i love that scene where he is mooning the movie theatre line, and then they just stop. |
| 17:42.21 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 17:42.21 | b11d | that cracked me up |
| 17:42.46 | monsted | the card is $80, an FXS module $70 and an FXO module $80 (or somewhere along those lines) |
| 17:42.49 | *** join/#asterisk _-Jon-_ (n=jon@CPE000d8861e8f7-CM00080d290642.cpe.net.cable.rogers.com) |
| 17:42.54 | krondorl | Anyone know a good brand of IAX phones other than GNET?? |
| 17:42.57 | roxy_ | If I buy a TMD400P with 1FXS and 1FXO, can I add to that same card later ? (I don't want to spend much, just playing around atm.) |
| 17:43.01 | _-Jon-_ | Hey everyone |
| 17:43.05 | b11d | yes you can |
| 17:43.09 | b11d | they are moduler |
| 17:43.10 | b11d | doh |
| 17:43.12 | b11d | modular |
| 17:43.27 | roxy_ | b11d, monsted: thanks |
| 17:43.33 | mercestes | ManxPower: http://pastebin.ca/236178 |
| 17:43.41 | _-Jon-_ | I'm wondering if this is possible: To add a bit of text in front of the incoming callers name. eg: Line 1: <cid name> |
| 17:43.41 | b11d | glad to provide advice when I can |
| 17:43.54 | b11d | modify the source? |
| 17:44.23 | _-Jon-_ | b11d, oh okay, so no other way? |
| 17:44.54 | ManxPower | mercestes: which version of Asterisk? |
| 17:44.58 | mercestes | 1.2.13 |
| 17:44.59 | b11d | not that im aware of.. |
| 17:45.03 | b11d | i could be wrong _-Jon-_ |
| 17:45.54 | hmmhesays | Set(CALLERID(name)="WhatUP ${CALLERIDNAME}") |
| 17:45.57 | ManxPower | mercestes: it looks good to me, no idea why it is not working |
| 17:45.58 | _-Jon-_ | Wait, what about: SetCallerID("Line1: {CALLERIDNAME}" <{CALLERIDNUM}>)? |
| 17:46.06 | mercestes | ManxPower: Ok thanks..I'll tinker with it some more. |
| 17:46.12 | ManxPower | DON'T USE QUOTES IN CALLERID |
| 17:46.26 | _-Jon-_ | hmmhesays, you think that'll work? |
| 17:46.46 | hmmhesays | without the quotes apparently |
| 17:46.47 | b11d | that might work |
| 17:47.04 | *** join/#asterisk GaVak (n=denniso@adsl-074-228-124-003.sip.sav.bellsouth.net) |
| 17:47.06 | hmmhesays | ManxPower: I thought you had to use quotes around a string |
| 17:47.19 | _-Jon-_ | me too |
| 17:47.43 | hoobastooba | exten => s,4,Set(CALLERID(name)=${ARG1}) |
| 17:47.48 | hoobastooba | no quotes |
| 17:47.56 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 17:48.05 | *** join/#asterisk pmnke (n=perlmonk@hubert.perlmonkee.com) |
| 17:48.23 | hoobastooba | _-Jon-_: does your phone not put the Line 1 on the phone already? |
| 17:48.35 | hmmhesays | Set(ARG1="I am a String") |
| 17:48.44 | *** join/#asterisk Ebola (n=Ebola@host86-136-130-111.range86-136.btcentralplus.com) |
| 17:48.46 | Qwell[] | quotes are bad |
| 17:48.46 | pmnke | So... if I have an extension pattern such as: "_011." - this matches on the FIRST digit pressed after 011, and attempts to dial out. |
| 17:48.58 | pmnke | is there a way to tell asterisk to wait a little longer so people can dial? |
| 17:49.03 | b11d | wait |
| 17:49.11 | Qwell[] | pmnke: increase your digittimeout, or whatever |
| 17:50.18 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 17:50.18 | *** topic/#asterisk is Asterisk: The Open Source PBX -=- http://www.asterisk.org -=- New Asterisk 1.0, 1.2, 1.2-netsec, 1.4beta, Asterisk-addons 1.2, 1.4beta, Zaptel 1.2, 1.4beta, and Libpri 1.2 releases now available! (Oct. 18, 2006) -=- Join #freepbx for freepbx/trixbox support. -=- Join #asterisk-gui to learn about the new Asterisk GUI framework |
| 17:50.34 | hmmhesays | the Set example on the wiki uses quotes for a string no quotes for an int |
| 17:50.39 | Qwell[] | it's wrong |
| 17:50.52 | Qwell[] | it's all just strings, unless it's in $[] |
| 17:50.59 | hmmhesays | so it would be Set(ARG1=I am a string) |
| 17:51.02 | Qwell[] | correct |
| 17:51.05 | hmmhesays | ok |
| 17:51.10 | hmmhesays | someone should change that |
| 17:52.00 | _-Jon-_ | Is there any way to get the calleridname to show in the console when I place a call? Just for testing purposes |
| 17:52.19 | hmmhesays | NoOP(${CALLERIDNAME}) |
| 17:52.20 | Qwell[] | NoOp(${CALLERID(name)}) |
| 17:52.23 | hoobastooba | if you are setting caller id it will show in the CLI |
| 17:52.24 | _-Jon-_ | Ah, thanks :) |
| 17:52.27 | Qwell[] | hmmhesays: You lose :p |
| 17:52.29 | hmmhesays | yeah |
| 17:52.31 | hmmhesays | you're right |
| 17:52.37 | hmmhesays | thats deprecated |
| 17:52.39 | Qwell[] | in 1.4 anyhow |
| 17:52.43 | Qwell[] | in 1.4 I think it's finally gone |
| 17:52.52 | hmmhesays | I'm sick give me a break |
| 17:52.55 | pmnke | b11d: wait happens after the call is answered... I don't want it to answer yet, as that confuses the user. |
| 17:53.12 | b11d | oh |
| 17:53.16 | b11d | yeah.. sorry |
| 17:53.23 | hoobastooba | pmnke: what? |
| 17:53.26 | pmnke | Qwell[]: I tried altering my digittimeout - it doesn't help. |
| 17:53.47 | pmnke | hoobastooba: I want to enable international dialing, I am using a pattern of "_011." |
| 17:54.00 | pmnke | This, however, matches on the FIRST digit pressed after the last 1 |
| 17:54.05 | pmnke | and dials out. |
| 17:54.14 | pmnke | I need to make Asterisk wait a reasonable number of seconds |
| 17:54.21 | pmnke | for the full string. |
| 17:55.31 | hmmhesays | _011. shouldn't match that |
| 17:55.49 | hoobastooba | exten => _011!,n,Dial(Zap/g1/${EXTEN:0}) |
| 17:55.55 | hoobastooba | that is how i do it |
| 17:55.55 | Qwell[] | eh? |
| 17:56.03 | Qwell[] | hoobastooba: Maybe you can explain exactly what ! does |
| 17:56.20 | hmmhesays | that makes it louder, with more emphasis |
| 17:56.32 | _-Jon-_ | thanks for your help guys. works pefectly :) |
| 17:56.33 | pmnke | yeah, "!" isn't mentioned in any of the docs I've read. |
| 17:56.36 | Qwell[] | I'd ~lart you, if jbot were alive |
| 17:56.55 | hmmhesays | ~lart |
| 17:57.03 | pmnke | regardless, I tried "!" with the same result. |
| 17:57.12 | hmmhesays | jbot is alive |
| 17:57.14 | pmnke | I am trying to dial "011-254-..." |
| 17:57.22 | Qwell[] | pmnke: One of your timeouts are too low |
| 17:57.24 | pmnke | and it keeps dialing out at 0112 |
| 17:57.27 | Qwell[] | Should be digittimeout I thought |
| 17:57.31 | hmmhesays | ~lart asterisk Qwell[] |
| 17:57.37 | Qwell[] | ~kill hmmhesays |
| 17:57.40 | jbot | ACTION shoots a charged fluxneutron gun at hmmhesays |
| 17:57.49 | hmmhesays | lol |
| 17:57.57 | hmmhesays | bah I hate sore throats |
| 17:58.09 | b11d | im telling you.. brandy, tea, lemon. |
| 17:58.09 | Qwell[] | That's what you get :p |
| 17:58.10 | b11d | it works |
| 17:58.20 | Qwell[] | b11d: tea / lemon optional? |
| 17:58.21 | hoobastooba | Qwell[]: i am not sure what ! does... got it from an example :-D |
| 17:58.23 | b11d | use real lemons into a cup of tea, add a shot of brandy. |
| 17:58.25 | b11d | it really works |
| 17:59.01 | b11d | I had a sore throat so bad one day I couldnt swallow.. went to the culinary dept here on campus and they whipped one up for me. |
| 17:59.07 | b11d | the sore throat was gone on the FIRST sip |
| 17:59.38 | pmnke | Qwell[] doesn't a digit timeout only effect an answered call also? |
| 17:59.45 | Qwell[] | pmnke: don't think so |
| 18:00.44 | b11d | no way.. Microsoft is parterning with Novell to support SUSE Linux?? |
| 18:00.46 | *** part/#asterisk andresmujica (n=andresmu@201.245.231.252) |
| 18:00.51 | b11d | I knew MS would have to bite it sooner or later |
| 18:01.01 | *** join/#asterisk barttg (n=barttg@pool-70-20-23-23.bstnma.fios.verizon.net) |
| 18:01.10 | pmnke | exten => _011.,1,Set(TIMEOUT(digit)=5) |
| 18:01.11 | pmnke | exten => _011.,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) |
| 18:01.21 | pmnke | still dials out on 0112 |
| 18:01.29 | hoobastooba | b11d: where did you read about MS and Novel on suse? |
| 18:01.36 | hoobastooba | is there a press release? |
| 18:02.08 | b11d | yes |
| 18:02.10 | b11d | hang on |
| 18:02.20 | b11d | http://www.marketwatch.com/news/story/Story.aspx?guid=%7BEEF34C41%2D480F%2D4ABC%2D9D0F%2DE5BC53E5C552%7D&siteid= |
| 18:02.24 | b11d | they are "expected to announce" |
| 18:02.24 | b11d | oh |
| 18:02.40 | barttg | hi all. does anyone know a good gui with real-time support? |
| 18:02.51 | b11d | :| |
| 18:02.52 | hoobastooba | so suse is soon to require a license is what that means. |
| 18:02.59 | b11d | it already requires a license |
| 18:03.04 | b11d | at least, SLES did |
| 18:03.10 | b11d | Novell's commerical SUSE |
| 18:03.13 | hoobastooba | oh |
| 18:03.20 | b11d | the State of MN licences it from them already.. we have been for like 3 years. |
| 18:03.23 | b11d | its bullshit |
| 18:03.25 | b11d | total bullshit |
| 18:03.55 | *** join/#asterisk ToTo (n=ToTo@host97-157-dynamic.2-87-r.retail.telecomitalia.it) |
| 18:04.37 | b11d | I should clean my office |
| 18:04.42 | b11d | it really got bad over the last week |
| 18:05.09 | pmnke | http://kepler.net/~perlmonkee/stuff/int.txt |
| 18:05.23 | pmnke | that is a few lines from the asterisk console when I try to dial. |
| 18:05.28 | pmnke | it is not respecting any of the timeouts. |
| 18:05.28 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 18:05.40 | pmnke | it says its setting them, but those lines all print out at the same time. |
| 18:05.55 | b11d | what exactly is the exten => line |
| 18:06.41 | pmnke | reload that text file to see. |
| 18:07.11 | pmnke | (or scroll up to just before the SuSE conversation) |
| 18:07.24 | b11d | hmm |
| 18:07.29 | b11d | i dont know :( |
| 18:07.31 | Qwell[] | yeah, show us the exten line |
| 18:07.35 | Qwell[] | the full line |
| 18:07.41 | hoobastooba | yeah show us |
| 18:07.41 | b11d | its at that URL qwell |
| 18:07.43 | *** join/#asterisk W9SH (n=W9SH@adsl-068-209-117-205.sip.asm.bellsouth.net) |
| 18:07.43 | Qwell[] | oh, you did |
| 18:07.44 | b11d | yeah show us |
| 18:07.45 | b11d | show us |
| 18:07.51 | b11d | http://kepler.net/~perlmonkee/stuff/int.txt |
| 18:07.52 | b11d | its there |
| 18:08.04 | b11d | (fyi, pastebin may be easier for you pmnke) |
| 18:08.43 | pmnke | pfft, why use that when my irc client is on the same machine hosting that text file. |
| 18:09.03 | Corydon-w | pmnke: um, how do you pronounce your nick? |
| 18:09.09 | pmnke | I don't |
| 18:10.21 | *** part/#asterisk hoobastooba (n=ckwall@63.149.122.93) |
| 18:13.45 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 18:14.07 | *** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
| 18:14.16 | FuriousGeorge | good afternoon all |
| 18:14.26 | b11d | afternoon |
| 18:14.41 | b11d | its pem-key |
| 18:14.49 | b11d | pmn-ke |
| 18:14.56 | pmnke | any ideas anyone? |
| 18:15.03 | *** join/#asterisk oreonixon (n=oreonixo@63.149.122.93) |
| 18:15.06 | b11d | yeah.. a sanding machine based around the concept of an electric razor |
| 18:15.11 | b11d | i dont think its been done |
| 18:15.18 | pmnke | any ideas related to my problem? |
| 18:15.20 | b11d | oh |
| 18:15.22 | b11d | :) |
| 18:15.25 | b11d | hrm.. |
| 18:15.51 | b11d | the problem is you want someone to be able to dial 1 or 111 right |
| 18:15.56 | hmmhesays | what you say we go picking wild flowers, got a spot way back in the woods |
| 18:16.00 | b11d | without it just jumping to 1 before they can enter the other two 1's |
| 18:16.07 | b11d | im down for that hmm |
| 18:16.10 | hmmhesays | sneak away for a couple of hours, spend a little time, picking wildflowers |
| 18:16.10 | pmnke | b11d: no. |
| 18:16.17 | b11d | oh |
| 18:16.22 | pmnke | read the text file =( |
| 18:16.25 | b11d | hmm.. want to eat mushrooms and ride my magic carpet? |
| 18:16.27 | b11d | ok |
| 18:16.31 | pmnke | <PROTECTED> |
| 18:16.51 | pmnke | it sees that I've matched with "011" and then I hit 2 (the firt digit in the country code I want to call) |
| 18:17.07 | pmnke | and says "hey! that matches 1 or more characters!" (which is what "." says to do) |
| 18:17.10 | pmnke | and dials out. |
| 18:17.21 | pmnke | so it ends up dialing 0112 |
| 18:17.50 | b11d | and when you remove that last . ? |
| 18:18.04 | pmnke | t-then it would only match on 011 |
| 18:18.11 | pmnke | and dial out right then and there. |
| 18:18.26 | b11d | argh.. i see what you mean |
| 18:19.19 | *** join/#asterisk jm|home (n=jamiem@dilbert.jamiem.com) |
| 18:19.25 | jm|home | hello |
| 18:20.01 | b11d | hi |
| 18:20.34 | jm|home | what do I use in extensions.conf to check for off hook? |
| 18:20.47 | b11d | read this pmnke: http://www.voip-info.org/wiki/index.php?page=Asterisk+Extension+Matching |
| 18:20.47 | Qwell[] | jm|home: You don't. However... |
| 18:20.51 | jm|home | :( |
| 18:21.00 | jm|home | however .... |
| 18:21.02 | Qwell[] | I assume you're meaning on a zap phone, you want it to automatically do something when you pick up the phone? |
| 18:21.08 | jm|home | no |
| 18:21.12 | Qwell[] | explain |
| 18:21.16 | jm|home | it is a zap phone, yes |
| 18:21.21 | b11d | yes.. explain |
| 18:21.31 | jm|home | there's an analogue phone on the same line (an extension in the lounge) |
| 18:21.39 | Qwell[] | ugh |
| 18:21.44 | jm|home | yes, I know ... |
| 18:21.54 | jm|home | but I can't afford a FXS<-->FXO just yet |
| 18:22.02 | jm|home | I have two other IP phone (office, kitchen) |
| 18:22.12 | jm|home | if the analogue phone picks up the IP phones keep ringing |
| 18:22.23 | jm|home | I was trying to do a nast-E-hack |
| 18:22.42 | jm|home | where I would ring for a few seconds, check for offhook, goto ring for a few seconds again, check for offhook .... |
| 18:22.48 | jm|home | iyswim? |
| 18:24.04 | pmnke | b11d: that link explains to me that things should work the way I think they will. |
| 18:24.08 | pmnke | the fact of the matter is, they are *not* working. |
| 18:24.19 | pmnke | asterisk is *not* waiting. |
| 18:24.43 | *** join/#asterisk xezz (n=xez@serial.trust-it.gr) |
| 18:25.53 | pmnke | oh, this is interesting. from the voip-info page on DigitTimeout |
| 18:26.04 | pmnke | "Note that if the user has typed a sequence of digits that make up a valid extension number, it will be interpreted immediately, without waiting for the timeout." |
| 18:26.44 | xezz | hello |
| 18:26.50 | xezz | anyone have seen that before : |
| 18:27.07 | *** join/#asterisk slayer192 (n=slayer19@66.138.39.225) |
| 18:27.20 | xezz | Nov 3 19:07:44 DEBUG[3646] manager.c: Manager received command 'QueueStatus' |
| 18:27.21 | xezz | Nov 3 19:07:44 DEBUG[3646] manager.c: Manager received command 'Status' |
| 18:27.21 | xezz | Nov 3 19:07:44 DEBUG[3646] manager.c: Manager received command 'ZapShowChannels |
| 18:27.28 | hmmhesays | nothing like the vitamin painkiller combo |
| 18:27.48 | hmmhesays | 50 mg zinc 250mg mixed vitamins 200mg vitamin C and 400mg ibuprofen |
| 18:28.40 | GaVak | Ok, I've been juggling around with * behind NAT, Polycom 501 behind NAT. I've gotten it to work on my DSL in the office, but I can't get a remote users to send/receive audio. I've tried port forwarding and even DMZ'ing his phone on his router. |
| 18:29.10 | GaVak | Is it possible that since it is a Vonage network adapter, RTP will not function the right way on his network? |
| 18:29.31 | FuriousGeorge | does anyone feel look looking at a signalling dump and telling me where my rtp is going, cuz i cant figure it out :) |
| 18:29.44 | hmmhesays | rtp debug |
| 18:29.48 | hmmhesays | that'll tell you in a hurry |
| 18:29.53 | FuriousGeorge | hmmhesays: its for SER |
| 18:29.59 | FuriousGeorge | i was trying to sneak one in there |
| 18:30.23 | hmmhesays | well, if you aren't using rtpproxy then rtp is going between your endpoints |
| 18:30.32 | hmmhesays | unless one is behind nat, then one rtp stream is going nowhere |
| 18:30.32 | FuriousGeorge | hmmhesays: but i am using rtpproxy |
| 18:30.52 | FuriousGeorge | i even made sure to specify what ports it would use in the source |
| 18:32.13 | *** join/#asterisk ManxPower (n=manxpowe@189.sub-75-203-37.myvzw.com) |
| 18:33.27 | pmnke | How do you people make international calls in your dial plan? |
| 18:33.38 | *** join/#asterisk _alex_mx_ (n=alex@200.78.229.18) |
| 18:34.03 | hmmhesays | pmnke: you are asking an extremely vague question |
| 18:34.10 | hmmhesays | pastebin the relevant portions of your dialplan |
| 18:34.19 | fx0 | some of them prepend 011 before dialing, heh. |
| 18:34.37 | *** join/#asterisk Sasch (n=Admin@host102-30-static.107-82-b.business.telecomitalia.it) |
| 18:36.16 | *** join/#asterisk EyeCue (n=eyecue@unaffiliated/eyecue) |
| 18:36.43 | ManxPower | for international dialing I use exten => _011XXXX.,1,Dial(.... |
| 18:36.50 | ManxPower | well more accuratly |
| 18:37.12 | ManxPower | exten => _9011XXXX.,1,Dial(Zap/g1/${EXTEN:1}) |
| 18:37.44 | hmmhesays | compiling openser |
| 18:37.44 | hmmhesays | fun |
| 18:37.56 | hmmhesays | I wish asterisk could pass through t.38 damnit |
| 18:38.49 | shellshark | it can |
| 18:38.59 | shellshark | there is a chan_t38 in beta iirc |
| 18:39.31 | file | shellshark: no there isn't... |
| 18:39.50 | *** join/#asterisk deb_user (n=none@70-59-108-105.albq.qwest.net) |
| 18:39.57 | Sasch | when i compile asterisk return an error |
| 18:40.02 | Sasch | chan_phone.c:41:29: error: linux/compiler.h: No such file or directory |
| 18:40.03 | Sasch | make[1]: *** [chan_phone.o] Error 1 |
| 18:40.03 | Sasch | make[1]: Leaving directory `/usr/src/asterisk-1.2/channels' |
| 18:40.03 | Sasch | make: *** [subdirs] Error 1 |
| 18:40.03 | shellshark | hmm i seen something... |
| 18:40.05 | Sasch | why ?? |
| 18:40.07 | *** join/#asterisk Ebola (n=Ebola@host86-136-130-111.range86-136.btcentralplus.com) |
| 18:40.12 | shellshark | Sasch: pastebin please |
| 18:40.12 | file | shellshark: what distro are you using? |
| 18:40.13 | deb_user | does anybody have any idea why softphone voip connections on ubuntu are so lossy? |
| 18:40.18 | file | GAH |
| 18:40.22 | file | Sasch: what distro are you using? |
| 18:40.23 | Sasch | <shellshark> ok excusme |
| 18:40.24 | shellshark | file: does it matter? :) |
| 18:40.32 | Sasch | ubuntu server 6.10 |
| 18:40.34 | deb_user | I have a dual boot machine, and when I use x-lite on xp its soooo much better compared to ubuntu |
| 18:40.35 | hmmhesays | file file file |
| 18:40.51 | hmmhesays | how goes it? |
| 18:41.13 | file | not bad, yourself? |
| 18:41.18 | Sasch | <file> can help me ?? |
| 18:41.37 | file | Sasch: go into chan_phone.c and remove the include line for it... |
| 18:41.49 | hmmhesays | file: other than lying in bed for the last 3 days sick |
| 18:41.52 | hmmhesays | i'm good |
| 18:42.23 | hmmhesays | trying to get this ser install to work for me |
| 18:42.53 | hmmhesays | *openser |
| 18:43.43 | jeremy_g | love u guys |
| 18:43.45 | jeremy_g | i gotta go |
| 18:43.49 | jeremy_g | its weekend |
| 18:44.07 | *** join/#asterisk brif8 (n=brif8@67.78.24.178) |
| 18:44.24 | _alex_mx_ | anyone running * newer than 1.2.9 and has active SIP calls that can help me out a sec? |
| 18:44.31 | brif8 | file: this is Barry from bug # 0008280 |
| 18:44.38 | deb_user | has anybody found ekiga even usable on ubuntu? |
| 18:44.50 | hmmhesays | I used it on fc5 |
| 18:45.36 | brif8 | file: I tried your host = ip_address not dynamic and insecure = very, and still when I enable the Caller ID on the CG-410 is hangs ups and reports failed to authenticate user |
| 18:46.02 | ManxPower | brif8: does your callerid stuff have quotes in it? |
| 18:46.08 | deb_user | does anybody out there use a softphone on ubuntu? |
| 18:46.12 | file | do a sip debug and pastebin it |
| 18:46.58 | brif8 | ManxPower: no it's just a straight number |
| 18:47.13 | ManxPower | paste the callerid= line from sip.conf |
| 18:47.16 | brif8 | file: will do you want the debug from the call being recieved |
| 18:47.40 | file | ManxPower: he is receiving a SIP INVITE from a remote device (FXO) and the INVITE contains callerid number in the From user field, so it is not getting matched against a user |
| 18:47.44 | ManxPower | you know that many phones use the callerid as the auth info, right? |
| 18:47.52 | file | so authentication fails |
| 18:47.57 | file | brif8: sure |
| 18:48.00 | ManxPower | I never run into the issue since I never use the callerid info from the phone, I always override it with callerid= |
| 18:48.28 | deb_user | I have yet to find a satisfied softphone user on ubuntu |
| 18:49.29 | pmnke | I think people lost interest in my problem because it is too complicated. |
| 18:49.37 | cpm | softphones kinda suck. |
| 18:49.53 | cpm | ubuntu notwithstanding |
| 18:49.54 | deb_user | cpm: I've had pretty good success with them on windows |
| 18:50.05 | deb_user | cpm: x-lite has great quality on windows |
| 18:50.12 | cpm | yeah, there are some pretty decent commercial solutions |
| 18:50.15 | deb_user | cpm: and kiax isn't bad either |
| 18:50.17 | ManxPower | All SoftPHones suck! (c) 2006, ManxPower |
| 18:50.32 | deb_user | well...what would the asterisk ubergeeks recommend? |
| 18:50.34 | deb_user | an ata? |
| 18:50.40 | cpm | kiax sucks, I use it, but I have no illusions about it. |
| 18:50.47 | ManxPower | deb_user: SIPura |
| 18:50.53 | *** join/#asterisk Geliman (n=scorpio@unaffiliated/drkshdw) |
| 18:50.59 | ManxPower | SPA2100 is a good ATA + Router + NAT |
| 18:51.08 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 18:51.16 | cpm | I have very good luck with iaxy2s, but they are spendy, and tend to just drop dead |
| 18:51.18 | deb_user | ManxPower: I don't need a router or NAT |
| 18:51.28 | monsted | i just want some cheap SIP FXO and FXS devices |
| 18:51.28 | deb_user | just at ATA |
| 18:51.31 | ManxPower | deb_user: then get the SPA 2000 |
| 18:51.40 | deb_user | ManxPower: you have experience with it? |
| 18:51.58 | ManxPower | deb_user: I own an SPA2000, and SPA2100 and an SPA3000 |
| 18:52.07 | _alex_mx_ | anyone know when in 1.2 series a SIP channel "name" changed from SIP/whatever-xxxx to SIP/whatever-xxxxxxxx |
| 18:52.12 | deb_user | how's the quality? |
| 18:52.13 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:52.18 | ManxPower | I don't use them much at the moment because my perm internet connection has 900ms latency and 3000ms of jitter |
| 18:52.47 | ManxPower | _alex_mx_: it would be in UPGRADE.txt |
| 18:52.56 | monsted | ManxPower: ouch, you'd be better off doing VOIP over carrier pigeon |
| 18:52.57 | ManxPower | deb_user: it's good quality |
| 18:53.03 | ManxPower | monsted: Correct. |
| 18:53.23 | file | I don't believe the channel name generation change is in UPGRADE.txt |
| 18:53.39 | _alex_mx_ | don't see it |
| 18:53.54 | brif8 | file: give me a sec it would appear setting the host = ip now the Cg-410 won't register |
| 18:54.00 | file | _alex_mx_: why do you ask? |
| 18:54.07 | file | brif8: if you explicitly put the IP, you can't register |
| 18:54.17 | file | brif8: but if you put it to dynamic you can still use insecure=very |
| 18:54.49 | ManxPower | Sorry, UPGRADE.txt mentions the IAX2 channel names changing |
| 18:54.54 | _alex_mx_ | file, because we upgraded to branch and one of our apps that parses show channles concise quit working since now channel name has 8 digits at the end instead of 4 |
| 18:55.07 | ManxPower | brif8: if you set host= then devices can't register |
| 18:55.20 | file | using the channel name for things is dangerous |
| 18:55.21 | ManxPower | that's the way it works. |
| 18:55.36 | ManxPower | host= is supposed to specify the IP/hostname of the device. |
| 18:55.39 | _alex_mx_ | file, was just wondering where the change was documented |
| 18:55.41 | brif8 | If I have host = dynamic it was working , now that I have host = 10.10.10.39 it won't register |
| 18:55.55 | krondorl | Does anyone know of any IAX phones other then GNET? |
| 18:56.02 | file | I doubt it was, but the commit list would have a record |
| 18:56.10 | ManxPower | brif8: it is no longer PERMITED to register to Asterisk. |
| 18:56.42 | brif8 | ManxPower: That would appear the case, the WAN line is flashing which the manual says means it hasn't / can't register |
| 18:56.50 | brif8 | ok got it registered again |
| 18:57.03 | ManxPower | host=dynamic means "far end will register to tell us it's ip address". host=anythingelse means "host will not register because we know its IP" |
| 18:57.29 | ManxPower | you do realize that ALL registration does is notify the server of the client's IP, righjt? |
| 18:57.43 | ManxPower | if you want to permit/deny by IP then use permit= and deny= |
| 18:58.36 | *** join/#asterisk sexyken (n=sexyken@c-24-23-203-168.hsd1.ca.comcast.net) |
| 19:00.03 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 19:00.03 | *** topic/#asterisk is Asterisk: The Open Source PBX -=- http://www.asterisk.org -=- New Asterisk 1.0, 1.2, 1.2-netsec, 1.4beta, Asterisk-addons 1.2, 1.4beta, Zaptel 1.2, 1.4beta, and Libpri 1.2 releases now available! (Oct. 18, 2006) -=- Join #freepbx for freepbx/trixbox support. -=- Join #asterisk-gui to learn about the new Asterisk GUI framework |
| 19:04.28 | _alex_mx_ | file, how else can you start/stop monitor through the manager if you don't use the full channel name? |
| 19:06.21 | brif8 | file: http://pastebin.ca/236289 has both Caller ID Disabled and the call being received and then Caller ID enabled and the failed authentication |
| 19:06.39 | file | I meant parsing out the details to gather stuff, as channel naming conventions differ from channel driver to channel driver |
| 19:07.26 | file | brif8: do you have a user entry in sip.conf called 3000 |
| 19:07.35 | *** join/#asterisk xezz (n=xez@serial.trust-it.gr) |
| 19:07.40 | file | er wait, yes you do |
| 19:07.51 | file | pastebin: |
| 19:07.55 | file | sip show peer 3030 |
| 19:09.31 | brif8 | file: yes |
| 19:09.59 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.71.53) |
| 19:10.51 | pmnke | well... I guess the temporary work around is to disable "early dial" on my phones. |
| 19:11.00 | pmnke | but users wont be happy about that. |
| 19:11.00 | brif8 | file: http://pastebin.ca/236297 has sip show peer 3000 and [3000] from sip.conf |
| 19:11.12 | Zodiacal | anyone know of a good apc battery backup monitor for linux? |
| 19:11.40 | file | brif8: no no, it's matching peer 3030 |
| 19:11.42 | file | not 3000 |
| 19:12.06 | *** join/#asterisk SwK_ (n=Silik0nJ@208.44.30.243) |
| 19:12.13 | iCEBrkr | dddd |
| 19:12.15 | iCEBrkr | errr |
| 19:12.47 | brif8 | file: no 3030 is another unused port on the CG-410 the CG-410 has 4 ports but only one line connected 3000,3010,3020 and 3030 but only 3000 is active |
| 19:13.14 | file | I'm telling you what sip debug told me |
| 19:13.28 | brif8 | ok hey I'll send you 3030 |
| 19:13.33 | file | it is matching the peer named 3030 |
| 19:13.39 | file | so make sure insecure=very is there |
| 19:14.18 | brif8 | http://pastebin.ca/236304 for 3030 |
| 19:14.34 | file | it's not set to insecure=very |
| 19:15.01 | brif8 | ok adding insecure=yes and sendrpid and trustrpid to 3030 |
| 19:16.18 | brif8 | I bow to file: how did you see 3030 |
| 19:16.26 | file | it said it in the debug |
| 19:16.35 | file | Found peer '3030' |
| 19:17.14 | b11d | how can I submit something to be included on the wiki? |
| 19:17.20 | b11d | I'd like to post something about getting my vg-224 working |
| 19:17.24 | brif8 | ok the call now doesn't drop but I still don't see the caller ID on the IP phone ? |
| 19:17.38 | *** join/#asterisk Geliman (n=scorpio@unaffiliated/drkshdw) |
| 19:17.46 | ManxPower | b11d: you create an account on the wiki, then you can post/edit |
| 19:17.53 | b11d | oh ok.. |
| 19:18.05 | b11d | oh yeah "register" |
| 19:18.46 | b11d | bah. the registration fails because I use greylisting.. |
| 19:18.48 | b11d | :| |
| 19:19.26 | b11d | there we go |
| 19:19.35 | brif8 | file: ok http://pastebin.ca/236314 has callerID enabled as can be seen in the NoOp line, but the IP Phones still show 3000 as calling ??? |
| 19:19.53 | file | 3000 is coming in as the callerid name |
| 19:20.13 | file | so your phones might only be showing you the name |
| 19:20.14 | brif8 | why is it not the 3523029577 which is making the call |
| 19:20.31 | brif8 | how do I address that ? |
| 19:20.31 | file | I don't know, I don't control your SIP device |
| 19:20.48 | brif8 | IP phone is a snom 200 |
| 19:21.37 | *** join/#asterisk mitcheloc (n=mitchelo@titaniumsoft.net) |
| 19:21.48 | Sasch | this Nov 3 20:21:30 WARNING[3569]: chan_zap.c:921 zt_open: Unable to specify channel 1: No such device or address |
| 19:21.51 | Sasch | why ?? |
| 19:22.28 | brif8 | anyway that the 3000 and 3523029577 can be swtiched around ? |
| 19:22.39 | brif8 | by asterisk when it receives the call |
| 19:22.54 | yardB | i just set uo my freeworld account .. is it possible to call someone overseas by dial the 6 digit number only? |
| 19:23.13 | file | sure, dialplan logic |
| 19:23.31 | file | Set(CALLERID(name)=${CALLERID(num)}) |
| 19:24.36 | *** join/#asterisk xlogik (n=Miranda@pawn.twbg.com) |
| 19:25.33 | pifiu | i have a question, every third call or so, it seems i get this error about receiving a mini frame before teh first full frame, and the CLI gets spammed with it. When that happens it seems the caller cannot be heard, but he can hear the person who picks up the phone |
| 19:25.36 | pifiu | what causes this? |
| 19:25.47 | *** join/#asterisk linlin (n=linlin@71.194.70.13) |
| 19:25.47 | pifiu | i am using this on a 3 way IAX server setup |
| 19:26.13 | *** join/#asterisk sb_mx (n=sb_mx@200.78.229.18) |
| 19:26.24 | pifiu | however this one thing is happening between really 2, since there is a colocated server and then a server at each site. the colo just sends out the calls to the appropiate location |
| 19:26.28 | pifiu | when it sends it out thats when it happens |
| 19:27.16 | brif8 | file: I bow once again to you |
| 19:27.37 | brif8 | thank you |
| 19:27.53 | *** join/#asterisk blueneon (n=blueneon@dsl-146-30-49.telkomadsl.co.za) |
| 19:28.20 | blueneon | how would i block an extension from being able to make calls? |
| 19:28.40 | *** join/#asterisk Defraz (n=t0tal@fw.centrisys.com) |
| 19:28.44 | blueneon | i mean, to allow it to dial anything 3 digits or less, but 4+ blocked |
| 19:30.05 | sb_mx | blueneon, i would make custom context for that extension and only include exten => XXX,1,blablabla under its dialing rules |
| 19:31.33 | sb_mx | however, that would let him only dial anything with 3 digits not less |
| 19:32.10 | blueneon | hmm |
| 19:33.55 | sb_mx | not sure if this would work but you could add exten => XXXX.,1 and after that add exten => X. |
| 19:34.12 | *** part/#asterisk DasTech (n=DasTech@ppp-71-128-113-209.dsl.irvnca.pacbell.net) |
| 19:34.31 | blueneon | why not exten => _XXX./4,1,Hangup() |
| 19:34.42 | blueneon | where 4 is the extension i dont want calling more than 3 digits |
| 19:35.03 | *** join/#asterisk MIXX941 (n=mark@unaffiliated/mixx941) |
| 19:35.24 | sb_mx | if im not mistaken _XXX. means 3 or more |
| 19:36.25 | blueneon | yes |
| 19:36.28 | blueneon | thats what i wanted |
| 19:36.29 | *** join/#asterisk linlin (n=linlin@71.194.70.13) |
| 19:36.35 | blueneon | 3 or more must be blocked |
| 19:36.36 | blueneon | ;) |
| 19:37.01 | sb_mx | ohh thought you said "allow it to dial anything 3 digits or less" :P |
| 19:38.46 | *** join/#asterisk reza_ (i=reza@abort.boom.net) |
| 19:39.04 | reza_ | how do i enable the jitter buffer in 1.4-beta2? |
| 19:40.38 | b11d | why not go to beta3? |
| 19:41.52 | *** join/#asterisk anthonyl (i=anthony@nat/digium/x-b8f7478335efedbe) |
| 19:43.17 | reza_ | b11d - i downlaoded and built beta2 just yesterday |
| 19:43.26 | reza_ | what's the most stable version anyhow? |
| 19:44.06 | b11d | 1.2.13 is the most stable |
| 19:44.12 | b11d | but 1.4-beta3 is out.. |
| 19:44.22 | b11d | 1.4 aparently IS suffering from issues still.. so be warned. |
| 19:44.46 | file | can't make it better without people telling us where the issues are :D |
| 19:44.51 | b11d | exactly right |
| 19:44.57 | b11d | so report your bugs |
| 19:45.02 | pifiu | b11d |
| 19:45.04 | pifiu | help me! lol |
| 19:45.09 | b11d | whats the prob? |
| 19:45.12 | anthonyl | i would really just recomend useing 1.4svn |
| 19:45.18 | anthonyl | using* |
| 19:45.19 | *** join/#asterisk syzygyBSD (n=chatzill@poplar.matraex.com) |
| 19:45.44 | *** join/#asterisk knobenheimer (n=knobenhe@c-71-205-184-144.hsd1.mi.comcast.net) |
| 19:45.55 | b11d | pifiu.. can you show me what those errors are? |
| 19:45.58 | b11d | can you pastebin any? |
| 19:47.39 | pifiu | sure |
| 19:47.40 | pifiu | lol |
| 19:47.44 | pifiu | its the same thing over and over |
| 19:47.44 | reza_ | can i use 1.2.13 with the latest libs/zaptel shit or do i have to downgrade all that? |
| 19:47.45 | pifiu | one second |
| 19:49.17 | b11d | ok |
| 19:49.47 | grEvenX | hm |
| 19:50.02 | grEvenX | do you have to call NoCDR() at a specific time to avoid warning? |
| 19:50.14 | b11d | reza_.. i think you'd have to downgrade |
| 19:50.29 | b11d | and get all the libs and shit for 1.2.13 specifically |
| 19:50.36 | b11d | but I could be wrong.. i havent done it |
| 19:50.41 | *** join/#asterisk Dibblah (n=Dibblah@80-192-39-135.stb.ubr02.dund.blueyonder.co.uk) |
| 19:51.19 | *** part/#asterisk brif8 (n=brif8@67.78.24.178) |
| 19:51.57 | pifiu | ok b11d sorry i was gone for a second |
| 19:52.04 | pifiu | did you seew hat my problem is? |
| 19:52.12 | pifiu | what i had written? |
| 19:52.23 | b11d | yeah but i'd like to see an example of this error you're seeing |
| 19:52.27 | pifiu | ok |
| 19:52.42 | yardB | question to anyone who can help! |
| 19:52.47 | *** join/#asterisk Un1x (i=Un1x@CPE001731208485-CM00080d850684.cpe.net.cable.rogers.com) |
| 19:52.54 | Dibblah | This is a stupid question. But VOIP gateway boxes (HT611 specifically) are meant to just work aren't they? |
| 19:52.56 | b11d | ask |
| 19:53.22 | pifiu | here you go |
| 19:53.22 | pifiu | http://pastebin.ca/236386 |
| 19:53.52 | yardB | i am trying to call someone in jamaica ..is it sufficient to dial the 6digit number only or do i have to use a prefix .. i am calling from the usa' |
| 19:54.15 | b11d | hrm.. what * version pifiu? |
| 19:54.21 | Dibblah | yardB: Depends on your VOIP provider. |
| 19:54.34 | yardB | freedailup |
| 19:54.35 | *** join/#asterisk mv00 (n=darealg@80-235-135-138.cable.ubr07.newt.blueyonder.co.uk) |
| 19:54.46 | pifiu | 1.2.13 the latest stable |
| 19:54.48 | mercestes | I am trying to use ChanIsAvail() to detect teh status of a phone. When I place a phone into a conference the ChanIsAvail() seems to crash out. Is there a known issue with ChanIsAvail() and detecting a phone in a conference?? |
| 19:55.03 | pifiu | i am using IAX completely all the way |
| 19:55.17 | b11d | what kind of latency are you seeing between those peers? |
| 19:55.19 | pifiu | could zapata or something witht he timing be affecting it? |
| 19:55.21 | b11d | any packet loss? |
| 19:55.22 | pifiu | nothing |
| 19:55.26 | b11d | hrm |
| 19:55.26 | pifiu | like 20-50ms |
| 19:55.29 | b11d | oh |
| 19:55.35 | pifiu | nothing huge |
| 19:55.41 | pifiu | even if it was 100 it still isnt bad |
| 19:55.44 | *** join/#asterisk UlluKaPatha (n=adfa@adsl-66-139-19-181.dsl.hstntx.swbell.net) |
| 19:55.46 | *** join/#asterisk hmmhesays (n=hmmhesay@24-117-135-28.cpe.cableone.net) |
| 19:55.48 | yardB | Dibblah: i am attempting to dial someone with a freeworld number |
| 19:55.49 | UlluKaPatha | hi guys |
| 19:55.50 | pifiu | one is a colocated server, the other 2 are dsl connections |
| 19:55.56 | pifiu | but its not packet loss, its always like every 3rd call |
| 19:55.57 | UlluKaPatha | I need help with e&m |
| 19:55.59 | hmmhesays | ok what exactly is record-route in ser |
| 19:56.12 | UlluKaPatha | I am using em_w |
| 19:56.12 | reza_ | anyone use voxbone? |
| 19:56.35 | UlluKaPatha | the call is chopped off within the first dial tone, I get an error message "call cannot be completed as dialed |
| 19:56.51 | UlluKaPatha | and the phone shows that there is a voicemail from an unkown caller |
| 19:57.06 | UlluKaPatha | when I dial into my pbx from my cell phone |
| 19:57.23 | UlluKaPatha | guys |
| 19:57.34 | *** part/#asterisk krondorl (n=krondorl@acid.auricnet.ca) |
| 19:57.36 | UlluKaPatha | I really appreciate help on this matter |
| 19:58.17 | UlluKaPatha | I am using TE110p |
| 19:58.49 | UlluKaPatha | helllllllllllo |
| 19:58.57 | UlluKaPatha | can anyone help me out here? |
| 20:00.25 | pifiu | b11d any idea? |
| 20:00.39 | Corydon-w | UlluKaPatha: add a 'w' before the first digit |
| 20:01.03 | b11d | im looking |
| 20:01.08 | b11d | this is causing audio issues then? |
| 20:01.16 | b11d | from what im reading, it can be ignored.. |
| 20:01.24 | b11d | but.. if its causing an issue,thats different |
| 20:02.34 | hmmhesays | openser day openser day |
| 20:02.49 | UlluKaPatha | codydon |
| 20:02.50 | *** join/#asterisk jart (n=user@ool-44c04b3a.dyn.optonline.net) |
| 20:02.53 | jart | so pigs |
| 20:02.59 | UlluKaPatha | w before the first digi? |
| 20:03.03 | jart | what sort of scalability improvements were made in 1.4 |
| 20:03.06 | UlluKaPatha | I am sorry, I do not know what that means |
| 20:03.12 | b11d | oh you know we memcached the grid overlays.. |
| 20:03.18 | b11d | did a bunch of tuning on the j-bar code.. |
| 20:03.20 | b11d | you know.. |
| 20:03.27 | b11d | its totally scalable now |
| 20:03.40 | UlluKaPatha | well, I am dialign into the Asterisk |
| 20:03.45 | Corydon-w | UlluKaPatha: Dial(Zap/g0/w${EXTEN:1}) |
| 20:03.48 | jart | grid overlays? |
| 20:03.53 | jart | j-bar? |
| 20:03.58 | b11d | stfu |
| 20:04.04 | UlluKaPatha | when I dialoutside, i get ringtones, but the phone I am calling never rings |
| 20:04.08 | b11d | :) |
| 20:04.10 | b11d | its a joke eh |
| 20:04.30 | b11d | pifiu.. thats a strange problem lad. |
| 20:04.33 | jart | ah, i guess i'm not enough of a newb to nod along |
| 20:04.41 | b11d | yeah, i guess not.. |
| 20:04.45 | *** join/#asterisk haggai (n=halls@credativ.bcnadsl.com) |
| 20:04.50 | UlluKaPatha | thanks, let me try that Corydon but will that fix the outgoing issue too? |
| 20:04.57 | yardB | one more time to anyone: to dial a freedialup number in another country ..is it sufficient to use a prefix? i am also a freeworluser |
| 20:05.04 | jart | i take asterisk too seriously because i get paid a lot of money to screw with it 16 hours a day |
| 20:05.08 | Corydon-w | UlluKaPatha: that's for the outgoing issue |
| 20:05.14 | jart | it destroys my sense of humor |
| 20:20.21 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 20:20.21 | *** topic/#asterisk is Asterisk: The Open Source PBX -=- http://www.asterisk.org -=- New Asterisk 1.0, 1.2, 1.2-netsec, 1.4beta, Asterisk-addons 1.2, 1.4beta, Zaptel 1.2, 1.4beta, and Libpri 1.2 releases now available! (Oct. 18, 2006) -=- Join #freepbx for freepbx/trixbox support. -=- Join #asterisk-gui to learn about the new Asterisk GUI framework |
| 20:20.30 | mfroes | do anyone has worked with sipsak ???? |
| 20:20.33 | *** join/#asterisk hoobastooba (n=ckwall@63.149.122.93) |
| 20:21.10 | hoobastooba | i added call-limit=1 to my sip.conf entries, then reloaded... it worked. Now i have removed it and have reloaded multiple times and it still is limiting my calls. |
| 20:21.16 | hoobastooba | how can i get this back to how I want it. |
| 20:21.23 | Corydon-w | call-limit=0 |
| 20:21.33 | hoobastooba | will try |
| 20:21.59 | hoobastooba | i cant just remove the line and reload? |
| 20:22.27 | UlluKaPatha | thanks corydon |
| 20:22.32 | UlluKaPatha | its not working |
| 20:22.39 | UlluKaPatha | I guess I am gonna go back to PRI singalling |
| 20:22.44 | b11d | how can there be this many people idling in here right now? |
| 20:22.48 | b11d | fucking idlers |
| 20:22.48 | b11d | :) |
| 20:22.55 | mercestes | .... |
| 20:22.58 | b11d | there we go |
| 20:23.00 | mercestes | I'm not idling...I believe I asked a question. |
| 20:23.17 | b11d | where? |
| 20:24.20 | UlluKaPatha | corydon: talking of calleridname, if I go on SIP trunks, will that support calleridnames? |
| 20:24.21 | mercestes | earlier. |
| 20:24.38 | mercestes | I'm trying to use ChanIsAvail() and when I try to ChanIsAvail() on a phone in a conference call it immediately crashes. |
| 20:24.51 | mercestes | My only other code at this point just NoOp's what hte status is but it doesn't provide that output. |
| 20:25.11 | Corydon-w | UlluKaPatha: yes, but if you've got a T1, I'd recommend PRI |
| 20:25.46 | mercestes | I'm running Asterisk 1.2.13. |
| 20:26.08 | b11d | asterisk crashes? |
| 20:26.09 | pifiu | b11d what do you mean check my routing config? |
| 20:26.12 | b11d | does it core dump? |
| 20:26.19 | b11d | check your routing config on your asterisk boxes |
| 20:26.19 | mercestes | no, but the code exits non-zero |
| 20:26.26 | b11d | submit a bug report :) |
| 20:27.07 | b11d | pifiu.. set up static routes between your asterisk boxes? |
| 20:27.48 | *** join/#asterisk alexns (n=alexns@static-71-240-121-39.pitt.east.verizon.net) |
| 20:28.00 | UlluKaPatha | ok thanks |
| 20:28.18 | pifiu | no |
| 20:28.25 | alexns | upgrade from ast 1.2 to 1.4 zaptel doesn't load in asterisk; different config file ?? |
| 20:28.41 | pifiu | b11d i have the peer as dynamic, and the host as static for each entry |
| 20:28.51 | b11d | im not talking about anythign to do with asterisk |
| 20:28.56 | b11d | im talking about your OS routing table |
| 20:29.27 | *** join/#asterisk oreonixon (n=oreonixo@63.149.122.93) |
| 20:29.29 | *** join/#asterisk umay (n=chris@71-208-192-243.hlrn.qwest.net) |
| 20:29.44 | pifiu | oh |
| 20:29.47 | pifiu | no i havent touched that |
| 20:29.57 | pifiu | i dont even know what to cehck or modify |
| 20:30.25 | b11d | you know how to set a default gateway and stuff right? |
| 20:30.32 | b11d | check that stuff out.. just make sure its correct. |
| 20:30.54 | *** join/#asterisk Un1x (i=Un1x@CPE001731208485-CM00080d850684.cpe.net.cable.rogers.com) |
| 20:31.42 | b11d | sup Un1x |
| 20:31.46 | b11d | what kind of processor should you get? |
| 20:31.49 | b11d | :P |
| 20:33.23 | alexns | upgrade from ast 1.2 to 1.4 zaptel doesn't load in asterisk; different config file ?? |
| 20:33.57 | b11d | there must be.. everyone seems to be asking that |
| 20:34.10 | b11d | read the updating & changelog files |
| 20:34.24 | file | did you upgrade zaptel to 1.4? |
| 20:34.31 | alexns | hehe just looking for the quick answer |
| 20:34.33 | alexns | yep |
| 20:34.38 | file | rerun configure? |
| 20:34.44 | file | confirm it picked up zaptel? |
| 20:34.52 | file | used make menuselect to confirm dependency was met? |
| 20:36.00 | *** join/#asterisk zotz (n=zotz@24.244.133.107) |
| 20:36.28 | alexns | yes show zaptel is there |
| 20:36.42 | alexns | zaptel driver is functioning |
| 20:37.04 | file | okay, so what does "doesn't load in asterisk" mean |
| 20:39.02 | alexns | zaptel works but asterisk is not loading the module, i am rebuiliding asterisk again after doing menuselect |
| 20:39.52 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 20:39.59 | alexns | hmm it is working now, strange |
| 20:40.03 | alexns | thanks file |
| 20:40.04 | b11d | whats the best way to buy a house.. |
| 20:40.13 | b11d | mortage broker, or bank, or person to person? |
| 20:40.21 | *** join/#asterisk kiddy (n=kiddy@59.93.7.19) |
| 20:40.28 | file | b11d: rob a bank, pay cash, move the house to Antarctica |
| 20:40.33 | b11d | done and done |
| 20:40.38 | Un1x | lol |
| 20:40.44 | b11d | i know zizzizum will help me out.. he lives in Argentina |
| 20:40.51 | b11d | thats like.. hours from the antarctic |
| 20:41.03 | kiddy | is there is any way to hear emails through our asterisk extensions ? |
| 20:41.12 | b11d | yeah |
| 20:41.15 | b11d | use festival |
| 20:41.23 | b11d | convert text to awesomely synthezied speech |
| 20:41.53 | b11d | or, pay people in india $0.10 an hour to just log in, and read your email back to you.. |
| 20:41.57 | b11d | that'd be awesome :) |
| 20:41.59 | kiddy | b11d : can you please give me the URL ,and is it need a specific hardware ? |
| 20:42.12 | b11d | no its not specific hardware.. hang on |
| 20:42.20 | kiddy | ok |
| 20:42.31 | b11d | start here: |
| 20:42.32 | b11d | http://www.voip-info.org/wiki-Asterisk+Festival+installation |
| 20:42.44 | b11d | the voice is generates isnt that hot though, in all seriousness. |
| 20:42.45 | b11d | but it would work |
| 20:42.55 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 20:42.57 | PakiPenguin | hello |
| 20:43.03 | b11d | you'll probably have to hack out some script to get your email and pass it to festival though |
| 20:43.06 | b11d | hey Paki |
| 20:43.15 | PakiPenguin | hey b11d |
| 20:43.22 | b11d | whats the real deal with Musharraf? Is his new book on the level? |
| 20:43.38 | b11d | he certainly made me think highly of pakistan, but I wondered if it was a snow job.. |
| 20:43.43 | b11d | if you dont mind the question, that is. |
| 20:43.58 | kiddy | b11d : but I read somewhere that it need sound cards , ok let me start it |
| 20:44.06 | b11d | umm.. yeah you should have a sound card. |
| 20:44.17 | b11d | i guess they are so common I dont htink of them as "special hardware" |
| 20:45.00 | b11d | I actually had festival reading back weather reports to people... so it can work, and does work. |
| 20:46.21 | b11d | no comment then Paki? |
| 20:48.51 | kiddy | hmm there is no Multimedia audio controller on my server |
| 20:49.15 | b11d | you dont have onboard sound? |
| 20:49.21 | b11d | and no sound cards? |
| 20:50.37 | kiddy | no when I run lspci it doesn't show the sound card |
| 20:50.48 | kiddy | and I think it doesn't have onboard |
| 20:51.16 | b11d | why not take a look? |
| 20:51.24 | b11d | maybe onboard sound is disabled in the bios |
| 20:52.52 | PakiPenguin | haha |
| 20:53.04 | PakiPenguin | nope its a nice book , did you read it? |
| 20:53.09 | b11d | yes |
| 20:54.24 | b11d | but his portrayal of his good deeds is true then? |
| 20:54.41 | b11d | thats good.. because as I was reading it i was thinking "if this is all true, then more countries need leadership like this" |
| 20:56.03 | PakiPenguin | yeah :) its true |
| 20:56.06 | PakiPenguin | ost of it |
| 20:56.16 | PakiPenguin | b11d, he's a good leader generally |
| 20:56.43 | b11d | thats cool.. im glad to hear that.. |
| 20:57.01 | b11d | Pakistan certainly seems like a hell of a country.. im glad I read that book, it totally shattered my misconceptions about it. |
| 20:57.17 | PakiPenguin | haha :) |
| 20:57.28 | PakiPenguin | people have a lot of misconceptions |
| 20:57.40 | b11d | yeah you got that right.. i've been telling people at work all about it.. |
| 20:57.43 | justinu|laptop | most the media's fault |
| 20:57.47 | b11d | it sure is |
| 20:57.48 | justinu|laptop | mostly |
| 20:58.05 | b11d | are things as tense with India as the media plays? |
| 20:58.11 | b11d | what do the average people think? |
| 20:58.12 | justinu|laptop | brad pitt is making a movie about daniel perl |
| 20:58.20 | b11d | yeah thats probably not good :) |
| 20:58.29 | PakiPenguin | b11d, on the govt. level , i am not sure |
| 20:58.45 | PakiPenguin | but people dont have that much rage , what ever media shows |
| 20:59.00 | b11d | thats good to hear.. |
| 20:59.22 | b11d | I think his ideas about education are excellent.. I wish we'd do that stuff here in USA or in Canada |
| 21:00.07 | justinu|laptop | there was something on an independent TV channel about hollywood's 100 year long stereotyping of muslims |
| 21:00.24 | b11d | oh yeah? that'd be interesting to watch.. |
| 21:00.38 | b11d | ahh.. down with ALL stereotypes.. they are NEVER right. |
| 21:01.49 | justinu|laptop | if you have directv, check out link TV... chan 375 i think |
| 21:01.59 | b11d | I got rid of TV like 4 years ago |
| 21:02.01 | b11d | :/ |
| 21:02.17 | b11d | in fact, outside of work,i enjoy almost NO technology.. |
| 21:02.23 | justinu|laptop | fair enough... it's not all bad, just 95% of it |
| 21:02.23 | b11d | no phone, no cable, no internet, no computers.. |
| 21:02.27 | b11d | agreed |
| 21:03.07 | justinu|laptop | i wouldn't watch TV either, except for my dual tuner directv tivo |
| 21:03.12 | PakiPenguin | :) |
| 21:03.26 | PakiPenguin | no tv for me either :) hehe |
| 21:03.26 | b11d | :) |
| 21:03.37 | b11d | i DO download family guy and south park.. so I do watch those.. |
| 21:03.38 | b11d | but thats it. |
| 21:03.42 | *** join/#asterisk sb_mx (n=sb_mx@200.78.229.18) |
| 21:03.50 | justinu|laptop | there's a documentary on HBO about diebold electronic voting machines: "Hacking democracry" |
| 21:03.52 | b11d | otherwise, its discussion around some beers with some friends, or into the books |
| 21:03.59 | justinu|laptop | diebold tried to get an injunction to keep it off the air |
| 21:03.59 | b11d | you should watch Votergate |
| 21:04.02 | b11d | that is an eye opener! |
| 21:04.12 | b11d | hacking democracy eh.. im down with that.. ive got to see that |
| 21:04.24 | b11d | watch "Votergate" though.. its on google video |
| 21:04.26 | b11d | fucking crazy.. |
| 21:04.29 | justinu|laptop | cool |
| 21:04.32 | b11d | e-voting machines are NOT ok. |
| 21:04.37 | b11d | not yet anyway |
| 21:04.54 | justinu|laptop | not until the hardware and source is open |
| 21:04.57 | b11d | I love how we care about our elections here, but those voting machines are whats running elections in Afghanistan and Iraq. |
| 21:05.00 | b11d | we dont care about them.. |
| 21:05.07 | b11d | you got that right justinu|laptop |
| 21:05.31 | justinu|laptop | it's our democracy (supposedly), not a trade secret of diebold, inc. |
| 21:05.50 | b11d | thats right. I wish people would stop thinking politics is for politicians.. its FOR THE PEOPLE. |
| 21:06.08 | b11d | But no.. people think their involvement with politics needs to end at office bullshit and reality TV. |
| 21:06.37 | b11d | but we can thank out lawyer-politicians for that.. |
| 21:06.44 | b11d | in 1964, the Defense Authorization Act was 1 page.. |
| 21:06.48 | b11d | in 1977, it was 75 pages.. |
| 21:06.51 | b11d | today.. its 988 pages. |
| 21:06.53 | b11d | WTF is with that |
| 21:07.11 | *** join/#asterisk hohum (n=dcorbe@jomama.interceltelecoms.net) |
| 21:07.14 | *** join/#asterisk santiago (n=santiago@debian/developer/santiago) |
| 21:07.52 | *** part/#asterisk kiddy (n=kiddy@59.93.7.19) |
| 21:08.59 | b11d | if I keep talking like that, i'll get shipped off to gitmo.. the one destination in Cuba you CAN get to from the USA. |
| 21:09.11 | *** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
| 21:09.50 | PakiPenguin | lol |
| 21:09.54 | LuxuriousGeorge | :) |
| 21:10.02 | PakiPenguin | have you seen the road to gt. bay ? |
| 21:10.06 | b11d | no |
| 21:10.14 | Corydon-w | What, no Gorgeous George? |
| 21:10.29 | PakiPenguin | watch that |
| 21:11.01 | b11d | im downloading it now |
| 21:11.16 | b11d | http://video.google.com/videoplay?docid=-599098805530677622&q=road+to+guantanamo&hl=en |
| 21:11.16 | b11d | right? |
| 21:11.56 | PakiPenguin | yup! |
| 21:12.03 | florz | I'd say, that's _on_ Cuba, only, though, not _in_ Cuba =:-) |
| 21:12.28 | b11d | true |
| 21:12.49 | b11d | bastard. |
| 21:13.10 | b11d | :) |
| 21:13.10 | florz | :-) |
| 21:13.36 | b11d | as a dual citizen, of canada and the usa, I cannot legally travel to Cuba from Canada.. |
| 21:13.42 | b11d | which is bullshit.. |
| 21:13.49 | b11d | americans will get arrested for travelling to cuba through another country.. |
| 21:13.56 | b11d | what a retarted embargo |
| 21:14.21 | b11d | 700mb.. thank god im on an OC-3. |
| 21:15.04 | PakiPenguin | hehe |
| 21:15.14 | b11d | i love keepvid.com |
| 21:15.19 | PakiPenguin | b11d, i would like to steal that |
| 21:15.24 | b11d | keepvid.com |
| 21:15.26 | b11d | just download it |
| 21:15.32 | PakiPenguin | no oc-3 :) |
| 21:15.34 | b11d | oh |
| 21:15.35 | b11d | yeah.. |
| 21:15.36 | PakiPenguin | i have it on dvd |
| 21:15.37 | PakiPenguin | hehe |
| 21:15.39 | b11d | its sweet |
| 21:15.50 | PakiPenguin | how much do u pay for it |
| 21:15.53 | b11d | $0 |
| 21:15.58 | b11d | the State of Minnesota pays for it |
| 21:16.08 | b11d | actually pays for the four of them I have in the back |
| 21:16.25 | b11d | but we're a state hub.. all the northern minnesota campuses go through us. |
| 21:16.33 | PakiPenguin | haha nice nice |
| 21:16.37 | b11d | same with all the local state agencies.. |
| 21:16.39 | PakiPenguin | univeristy bw :p |
| 21:16.42 | b11d | :) |
| 21:16.52 | b11d | AND ITS ALL MINE.. MUAHAHAHAHA |
| 21:17.00 | b11d | I throttle the rest of the campus down to like 10k/sec |
| 21:17.03 | PakiPenguin | i can figure! |
| 21:17.08 | b11d | meanwhile I pull at like 3MB/sec :) |
| 21:17.09 | PakiPenguin | lol nooo!! thats mean |
| 21:17.11 | b11d | ok.. i dont do that.. but I could |
| 21:17.11 | b11d | :) |
| 21:17.31 | b11d | im not that big of a dick.. |
| 21:17.43 | b11d | same with censorship.. I get a request to block a website every week.. |
| 21:17.46 | b11d | but I wont do it |
| 21:17.48 | PakiPenguin | and i am happy with my 384k connection :p |
| 21:18.04 | b11d | you should be.. i dont know how oldschool you are, but remember the old days? |
| 21:18.09 | b11d | I rememebr 1200 baud. |
| 21:18.22 | PakiPenguin | lol i do |
| 21:19.48 | b11d | so what do you have going on this weekend PakiPenguin? |
| 21:20.05 | PakiPenguin | just came back from work |
| 21:20.19 | PakiPenguin | got work again in the morning :( |
| 21:20.25 | b11d | :( |
| 21:20.42 | PakiPenguin | how about you? |
| 21:20.58 | b11d | i declared that i am NOT going to work this weekend.. which is unusual for me.. |
| 21:21.05 | b11d | and im going to get buzzed with some friends tonight.. |
| 21:21.08 | b11d | and relax tomorrow |
| 21:21.25 | b11d | maybe finish off Bob Woodwards new book "State of Denial" |
| 21:21.39 | b11d | good read on what a fucking prick Donald Rumsfeld is.. |
| 21:21.54 | [shodan] | anyone knows how to use callerid with a SPA3k2 ? I tried dial(sip/fxo1/*67,60,D(www${EXTEN})) but that doesn't work , when I listen on the line I can hear the SPA dial *67 clearly , but then when ${EXTEN} is dialed it's too short and highly distorted that even the pstn line can't recognize de digits, what's the problem :> |
| 21:21.55 | [shodan] | ? |
| 21:22.14 | b11d | I cant say I know.. |
| 21:23.12 | b11d | so how long have you lived in Pakistan PakiPenguin? |
| 21:23.26 | b11d | ^^^^ anyone still fall for that? |
| 21:23.31 | [shodan] | I mean callerid blocking ! |
| 21:23.56 | b11d | I still dont know |
| 21:24.00 | b11d | im sorry to say |
| 21:24.06 | PakiPenguin | b11d, holdon on phone |
| 21:24.22 | b11d | sure |
| 21:26.50 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 21:31.15 | *** join/#asterisk Eliran_Itzhak (n=eliran@bzq-82-81-63-217.red.bezeqint.net) |
| 21:33.40 | b11d | this got dead |
| 21:36.04 | b11d | i see the Saddam Hussein case is pretty much over.. the verdict will be out soon. |
| 21:36.08 | b11d | what a showcase trial that has been |
| 21:36.17 | b11d | I wonder if he'll be able to appeal |
| 21:36.53 | b11d | also, I love that it focused on his war-crimes against the Kurds.. which we ENCOURAGED at the time.. |
| 21:37.03 | b11d | because we need that relationship with Turkey, and they hate the Kurds. |
| 21:37.32 | b11d | When people start to complain about human rights atrocities, you have to look at how we reacted when the event took place.. not 20+ years after the fact. |
| 21:37.33 | b11d | end-rant |
| 21:37.40 | *** part/#asterisk amdtech (n=amd011@ss-5-100.shsu.edu) |
| 21:39.36 | *** join/#asterisk alerios (n=alerios@190.24.97.148) |
| 21:40.07 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 21:40.11 | b11d | wb |
| 21:40.30 | PakiPenguin | thanks |
| 21:40.38 | PakiPenguin | accidently closed xchat |
| 21:41.02 | b11d | that'll happen |
| 21:43.13 | b11d | holy shit.. they're making Super Troopers 2 |
| 21:43.18 | b11d | excellent |
| 21:44.36 | jart | oh no |
| 21:45.12 | b11d | oh.. yeah.. sorry it wont appeal to you Mr. Spock |
| 21:45.22 | b11d | probably far to "low brow" |
| 21:51.18 | b11d | this is pretty neat: |
| 21:51.19 | b11d | http://chir.ag/phernalia/preztags/ |
| 21:51.23 | b11d | move that slider around. |
| 21:51.28 | *** join/#asterisk [F] (n=f@pool-72-66-18-227.washdc.fios.verizon.net) |
| 21:51.47 | [F] | question: can anyone help me figure out what specific hardware I need to get an asterisk server properly running? |
| 21:51.54 | b11d | well, lets talk about that |
| 21:52.04 | b11d | what do you want to do with the Asterisk system? |
| 21:52.10 | *** join/#asterisk eurocrash (n=eurocras@69.15.209.41) |
| 21:52.37 | *** join/#asterisk echosyp (n=stfu@ip70-185-147-60.lu.dl.cox.net) |
| 21:52.39 | [F] | well, to be quite honest with you I don't know. |
| 21:52.46 | b11d | well good.. be honest! |
| 21:52.59 | b11d | How many telephones do you need to support?> |
| 21:53.00 | [F] | for whatever reason |
| 21:53.03 | [F] | one. |
| 21:53.03 | [F] | for me. |
| 21:53.05 | [F] | and thats it. |
| 21:53.07 | b11d | oh ok.. this'll be easy then :) |
| 21:53.18 | [F] | for whatever reason I'm just very, very interested in the whole telephone system. |
| 21:53.25 | b11d | do you want to push all of your calls across the internet (pure VoIP) or do you want to use an existing line? |
| 21:53.31 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 21:53.33 | echosyp | existing |
| 21:53.40 | b11d | ? |
| 21:53.46 | [F] | which do you recommend? is there benefit of one over the other? |
| 21:53.47 | echosyp | we are in cahoots |
| 21:53.47 | yardB | b11d: i am listening .. i am in the same boat i have a system .. i have working as a PBX land line only but not sure whatesle asterisk can do |
| 21:53.50 | b11d | oh ok |
| 21:53.56 | eurocrash | hey... gentoo + te110p = howto? ... ztcfg -v does't like showing my shiney new TE110P card |
| 21:54.09 | b11d | one at a time here eh |
| 21:54.10 | b11d | :) |
| 21:54.24 | yardB | i would like to push mine over IP |
| 21:54.30 | [F] | b11d: seems like you're the lecturer now. ;p |
| 21:54.36 | [F] | sorry ;p |
| 21:54.39 | b11d | :) |
| 21:54.45 | yardB | any material i could read? |
| 21:54.49 | b11d | voip-info.org |
| 21:54.50 | b11d | read all |
| 21:54.51 | [TK]D-Fender | yardB : .... |
| 21:54.52 | [TK]D-Fender | ~book |
| 21:54.57 | jbot | rumour has it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 21:54.57 | [F] | yardB: wikipedia has a crap load. |
| 21:55.11 | yardB | i have that asterisk book which heps but i need more info on VOIP |
| 21:55.16 | *** join/#asterisk iulius (n=iulius@mail1.technologieshq.com) |
| 21:55.22 | b11d | do you want to go to a soft phone, or voip phone, or do you want to use your traditional pots phone? |
| 21:55.22 | [F] | jbot: !!! downloadable PDF, thanks. |
| 21:55.43 | [F] | i want to use a traditional phone. |
| 21:55.49 | [F] | well look |
| 21:55.52 | [F] | before I tell you that |
| 21:56.01 | [F] | I want to cut down on expenses |
| 21:56.10 | [F] | so if I can avoid it, I'd like to not get hardware I could do without. |
| 21:56.23 | b11d | you'll need a fxo/fxs card.. I recommend a Digium TDM400P with one fxs and one fxo port.. |
| 21:56.25 | [E] | i'll buy the analog to digital converter |
| 21:56.26 | b11d | that should get you going.. |
| 21:56.27 | [E] | i don't care |
| 21:56.43 | b11d | thats all you'll need, and a (hopefully) dedicated PC to run it on |
| 21:56.59 | [F] | b11d: could I get that at a BestBuy or is it something i'd probably have to get online? |
| 21:57.03 | b11d | you *might* run into issues with your traditional line.. |
| 21:57.07 | b11d | you've got to order it online.. |
| 21:57.13 | b11d | it'd be sweet if they were that common though :) |
| 21:57.13 | [F] | alright. |
| 21:57.18 | b11d | they arent pricey. |
| 21:57.23 | b11d | shouldnt be a big deal |
| 21:57.24 | [E] | $100 |
| 21:57.26 | [E] | ish |
| 21:57.27 | [F] | cool. |
| 21:57.31 | PakiPenguin | back :) |
| 21:57.33 | b11d | web |
| 21:57.34 | b11d | wb |
| 21:57.35 | b11d | ;) |
| 21:57.41 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 21:57.45 | PakiPenguin | hehe |
| 21:57.51 | [TK]D-Fender | I highly recommend against PCI based FXS. Use ATA's for that if needed. |
| 21:58.00 | b11d | echo? |
| 21:58.08 | [E] | yea? |
| 21:58.10 | [F] | [TK]D-Fender: ATA? |
| 21:58.14 | b11d | Analog Telephone Adapter |
| 21:58.15 | [E] | oh |
| 21:58.25 | [TK]D-Fender | b11d : PCI FXS is more clostly, places a higher load on your server, and less functional/flexible. |
| 21:58.31 | b11d | ahh.. cool.. |
| 21:58.48 | hads | But if you want to use an analog fax then bridging on the card FXO -> FXS is the most reliable. |
| 21:58.58 | [TK]D-Fender | [F] : I suggest the Linksys SPA-2002 if you want to use analog phones with * |
| 21:59.03 | b11d | due to codec issues? |
| 21:59.19 | [F] | [TK]D-Fender: that looks really sexy |
| 21:59.20 | [E] | so, i want to use my home phone with the server and setup voicemail, and use my pocket pc as a wifi phone |
| 21:59.34 | [TK]D-Fender | hads : No, if you want reliable faxing you'll leave it on a dedicated line that doesn't even APPROACH *. |
| 21:59.58 | [F] | do you guys recommend the asterisk distro? |
| 22:00.03 | [F] | http://www.voip-info.org/wiki/view/Asterisk+Install+CDROM |
| 22:00.12 | [E] | f |
| 22:00.14 | hads | Well for the few installations that I have, and analog FAX bridged on a TDM card works reliably. |
| 22:00.15 | [TK]D-Fender | [F] : Use whatever you feel most comfortable administering. |
| 22:00.18 | [E] | my server is crashing |
| 22:00.20 | [E] | :( |
| 22:00.22 | b11d | :( |
| 22:00.26 | [F] | [E]: thats because its Ubuntu for God's sake. |
| 22:00.31 | kFuQ | rofl |
| 22:00.32 | GaVak | I'm a little confused about the switch => command. How can I GoTo it if it doesn't have an exension or a priority? |
| 22:00.32 | [E] | yeah |
| 22:00.33 | [E] | heh |
| 22:00.39 | [D] | heh |
| 22:00.44 | [E] | heh |
| 22:00.55 | [TK]D-Fender | [F] : Though we might suggest you stick with one of the more popular and standardized ones like RHEL, CentOS, Debian, Slackware, FC, etc |
| 22:00.58 | [F] | so the asterisk install CD is just a stripped down distro with asterisk installed? |
| 22:01.07 | [D] | pretty much |
| 22:01.07 | [TK]D-Fender | [F] : At best |
| 22:01.12 | [F] | [TK]D-Fender: I've run slackware for about 5/6 years and just recently switched to ArchLinux. |
| 22:01.14 | monsted | has anyone used Audiocodes MP112 FXO gateways with asterisk? |
| 22:01.14 | [TK]D-Fender | [F] : I would personally suggest centOS. |
| 22:01.16 | [E] | this box sucks |
| 22:01.23 | justinu|laptop | gentoo all the way :P |
| 22:01.23 | [D] | I've been running Asterisk on FreeBSD for awhile.. rock solid. |
| 22:01.28 | [D] | fuck all linux distros |
| 22:01.28 | [D] | :) |
| 22:01.48 | [F] | i heard there were some complications with BSD |
| 22:01.49 | [E] | wooo, started this time |
| 22:01.51 | [TK]D-Fender | [D] : You can, but its not to say that it is a bump-free road, and lets not forget Zaptel.... |
| 22:01.53 | [F] | but maybe i'm just imaginging it |
| 22:01.59 | [D] | my zap channels work fine |
| 22:02.06 | [D] | im telling you, its solid |
| 22:02.09 | hads | And for those who don't want a dedicated fax line that is a good solution. |
| 22:02.12 | [D] | i cant speak to future versions though |
| 22:02.21 | [TK]D-Fender | [D] : Its just a question of "made for and works easily with" |
| 22:02.26 | [D] | yep |
| 22:02.28 | [D] | you're right |
| 22:02.31 | kFuQ | good luck with * & Fbsd here too |
| 22:02.33 | GaVak | Slack 11 with the huge26.s kernel had no problems w/ zap |
| 22:02.36 | [TK]D-Fender | hads : Yeah, if thats all you've got... |
| 22:02.50 | b11d | I dont know what the fuss is about.. FreeBSD and Asterisk and Zaptel are all working GREAT for me. |
| 22:02.51 | [E] | really |
| 22:02.53 | [E] | slack huh |
| 22:03.07 | hads | Have you tried an analog fax bridged on a TDM card? |
| 22:03.15 | b11d | i have not |
| 22:03.17 | [E] | b11d |
| 22:03.28 | [E] | do you know of a softphone for pocket pc that works well with asterisk |
| 22:03.30 | PakiPenguin | uff |
| 22:03.30 | [E] | i type slow |
| 22:03.31 | b11d | no |
| 22:03.34 | PakiPenguin | that is confusing |
| 22:03.35 | b11d | i know of no good softphones |
| 22:03.35 | hohum | what to I set type= if I only want to receive calls from a peer and not send |
| 22:03.39 | PakiPenguin | d,e,f :) |
| 22:03.40 | PakiPenguin | haha |
| 22:03.43 | b11d | :) |
| 22:03.58 | [F] | def is one of my favorite words. :) |
| 22:04.04 | [TK]D-Fender | Slackware = 100% complaint free success in my exerience. |
| 22:04.06 | b11d | what about the def tones? |
| 22:04.09 | *** join/#asterisk haikumore (n=haikumor@87.218.172.73) |
| 22:04.21 | [F] | b11d: the deftones, the def tones, the deft ones... |
| 22:04.21 | hohum | what to I set type= if I only want to receive calls from a peer and not send |
| 22:04.31 | [F] | b11d: one of my favorite bands. |
| 22:04.32 | b11d | :) |
| 22:04.34 | b11d | nice.. |
| 22:04.36 | b11d | good band indeed |
| 22:04.46 | kFuQ | hohum: don't give them any outgoing dialplan |
| 22:04.49 | GaVak | hohum: user? |
| 22:05.01 | b11d | I saw them twice in T.O. |
| 22:05.04 | b11d | back in the mid 1990's |
| 22:05.06 | [F] | okay. So lets say I have the Linksys SPA-2002 or the PCI card you mentioned, I have asterisk installed on a computer, and I've got an analog phone. Is there anything i'm missing? |
| 22:05.19 | [F] | b11d: I saw them in DC. It was fantastic. Have you got their new (leaked) CD? |
| 22:05.24 | [TK]D-Fender | [F] : That pretty much covers it. |
| 22:05.27 | b11d | no, not yet.. is it great? |
| 22:05.28 | [F] | or, i think it might have been released yesterday or today. |
| 22:05.30 | [F] | oh man it rocks. |
| 22:05.34 | b11d | really? ohh man |
| 22:05.36 | b11d | i need it |
| 22:05.38 | [F] | WAAAY better than there previou stuff |
| 22:05.41 | [F] | more White Pony'ish |
| 22:05.46 | [TK]D-Fender | [F] : how many lines do you have? |
| 22:05.52 | [F] | [TK]D-Fender: phone lines? one. |
| 22:05.55 | b11d | hrm.. thats cool.. im looking forward to it! |
| 22:05.57 | [F] | my internet is fiber optics. |
| 22:05.59 | hohum | gevak: sure its user and not peer? |
| 22:06.05 | knobenheimer | better than minerva? that album was terrible. |
| 22:06.12 | GaVak | one end is user, the other end is peer |
| 22:06.12 | [F] | knobenheimer: thats the album i'm talking about |
| 22:06.17 | [F] | thats the one this is way better than. |
| 22:06.24 | knobenheimer | good |
| 22:06.27 | knobenheimer | what's it called? |
| 22:06.29 | [TK]D-Fender | [F] : Then I might suggest you look at the SPA-3102. that'll give you 1 FXS & 1 FXO in a signle ATA. then means you don't even need to plug anything extra into your server |
| 22:06.35 | [F] | [TK]D-Fender: is having one line a problem? |
| 22:06.38 | [F] | knobenheimer: saturday night wrist |
| 22:06.50 | [F] | i don't want my parents and sister to not be able to use the phone. |
| 22:07.21 | [TK]D-Fender | [F] : The SPA-3102 is probably a better bet for you, and cheaper too. |
| 22:07.32 | [F] | [TK]D-Fender: the 2002 is cheaper I think. |
| 22:07.47 | [F] | but as long as its under $100 its fine. |
| 22:07.54 | [TK]D-Fender | [F] : I'm suggesting the SPA-3102 for FXO purposes so you don't need the TDM card. |
| 22:08.31 | [TK]D-Fender | [F] : So 1 x SPA-3102 for 1 FXS, 1 FXO, and you can add more SPA-2002's as needed for each pair of phones you'd like to convert |
| 22:08.44 | [F] | sorry what is FXS and FXO? |
| 22:08.59 | b11d | FXS = connects to phones |
| 22:09.04 | b11d | FXO = connects to the PSTN |
| 22:09.07 | [F] | and are you saying I need both the 3102 and 2002? |
| 22:09.34 | [E] | how are we gonna connect our servers? |
| 22:09.37 | [TK]D-Fender | [F] : the 3102 with let you take in your PSTN line, *AND* also let you plug in 1 phone as well. |
| 22:09.39 | b11d | ethernet |
| 22:09.45 | [F] | [TK]D-Fender: ahh |
| 22:09.53 | [F] | so if i only have one phone |
| 22:09.57 | [F] | ALL i need is the 3102 |
| 22:09.57 | [TK]D-Fender | [F] : so you'd only need 1 3102 |
| 22:09.58 | [F] | and thats it |
| 22:09.59 | [F] | and a computer |
| 22:10.05 | [F] | sweet. |
| 22:10.08 | [E] | do it |
| 22:10.09 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.193) |
| 22:10.24 | [F] | [TK]D-Fender: after that is it just a matter of configuring my asterisk server? |
| 22:10.27 | [TK]D-Fender | [F] : and if you want more independant analog phones going into your system then I'd suggest you use SPA-2002's from that point (1 ports on each) |
| 22:10.28 | *** join/#asterisk vader-- (n=johndoe@204.183.88.101) |
| 22:10.30 | [F] | or is more preperation necessary? |
| 22:10.41 | [TK]D-Fender | [F] : exactly. You never even have to shut it down for this, |
| 22:10.42 | [F] | [TK]D-Fender: why do you recommend 2002 from that point on? |
| 22:11.00 | PakiPenguin | [F], if you need more extensions in rooms |
| 22:11.04 | [TK]D-Fender | [F] : well... the 3102's point is that it can take in 1 land-line. you only HAVE 1 :) |
| 22:11.20 | [F] | i see. the 2002 does nothing with land-lines? |
| 22:11.21 | [TK]D-Fender | [F] : so beyond the first what you might want would be extra PHONE ports. |
| 22:11.37 | [TK]D-Fender | [F] : Correct. the 2002 only lets you use more PHONES as SIP phones. |
| 22:11.46 | [F] | okay everything makes perfect sense. |
| 22:11.57 | [TK]D-Fender | [F] : Takes a bit of gettin used to. |
| 22:12.10 | [E] | so |
| 22:12.18 | [F] | now I need to sell a bob dylan ticket, a G. Love and Special Sauce ticket, get the cash to get the SPA-3102. |
| 22:12.19 | [E] | my question is, how are we going to link our servers |
| 22:12.20 | hohum | question |
| 22:12.22 | [F] | and then i'll be in business. |
| 22:12.27 | b11d | Bob Dylan is from this town.. Hibbing.. and he fucking hate us here. |
| 22:12.33 | b11d | I know people who used to boo him off stage.. fucking old timers |
| 22:12.38 | [F] | b11d: how come? :( :( |
| 22:12.41 | kFuQ | stubhub? |
| 22:12.46 | [E] | he sucks |
| 22:12.48 | [E] | heh |
| 22:12.52 | [F] | ... |
| 22:12.55 | b11d | because everyone in this town profits off of his name, and dont ask him for permission. |
| 22:12.58 | hohum | if I have a peer with a phone number (like a DID) and I set it type=user and stick it in a specific context, I ought to be able to put an entry in that context in my dialplan to like for instance play an IVR back, right? |
| 22:13.04 | b11d | plus, hibbing people hated him when he started.. |
| 22:13.08 | b11d | i mean, really hated him.. |
| 22:13.13 | b11d | he was the "rich jewish kid" |
| 22:13.16 | b11d | stupid bigots.. |
| 22:13.30 | [F] | their loss I suppose. |
| 22:13.36 | b11d | yeah.. |
| 22:13.49 | [F] | okay, so now suppose i have my asterisk server running fine. |
| 22:13.54 | b11d | everyone thinks they are so great here in Hibbing.. "ohh.. Bob Dylan.. Bob Dylan.. blah blah blah" |
| 22:13.55 | b11d | I hate it |
| 22:14.01 | [F] | and i want to make a phone call |
| 22:14.04 | *** join/#asterisk souphead (n=soup@dns2.cascom.ca) |
| 22:14.11 | [F] | i pick up my analog phone, dial a number, and call whomever I want? |
| 22:14.24 | [F] | and it'll call regardless if I use VoIP or land-line? |
| 22:14.26 | [F] | oh, another thing. |
| 22:14.29 | [F] | when I use my land-line. |
| 22:14.35 | [F] | through asterisk. |
| 22:14.45 | [F] | my family, who aren't going to be connected to asterisk but still use the phone |
| 22:14.50 | [F] | will there be any conflict? |
| 22:15.16 | [TK]D-Fender | [F] : Well you'll be getting dial-tone from your ATA and * will process all calls. Your server will determine where the call being placed is to go over the internet, or out your landline |
| 22:15.26 | [TK]D-Fender | [F] : You control all of that |
| 22:15.29 | *** join/#asterisk aao_pwner (n=aao_pwne@c-24-21-91-140.hsd1.wa.comcast.net) |
| 22:15.45 | hoobastooba | ahhh what does this mean? |
| 22:15.46 | hoobastooba | Nov 3 15:15:18 WARNING[19301]: pbx.c:794 pbx_find_extension: Maximum PBX stack exceeded |
| 22:15.57 | [F] | [TK]D-Fender: yeah, but my question is, if other people in my house need to use the landline, is it wise for me to also use the landline? |
| 22:15.59 | hoobastooba | saw it when i did a reaload |
| 22:16.01 | [TK]D-Fender | [F] : Ok, how many phones are you looking to VoIP-enable? |
| 22:16.02 | hoobastooba | relaod |
| 22:16.04 | hoobastooba | reload |
| 22:16.10 | [E] | 1 |
| 22:16.11 | [F] | [TK]D-Fender: just one for me. |
| 22:16.55 | [TK]D-Fender | [F] : AH, so you want to basically leave them COMPLETELY alone and jsut let YOUR phone work kinda-hybrid like? |
| 22:17.11 | [F] | yes. that is exactly it. |
| 22:17.21 | [F] | i just want my phone to do what it needs to do. |
| 22:17.22 | [TK]D-Fender | [F] : Ah, then you ONLY need an SPA-3102. |
| 22:17.25 | [F] | and i will do what it takes to do it. |
| 22:17.30 | [F] | as long as they are left alone, and i get what i want. |
| 22:17.36 | [F] | alright, cool. |
| 22:17.52 | b11d | "Teacher" by Jethro Tull.. still stuck in my head.. |
| 22:17.54 | hohum | if I have a peer with a phone number (like a DID) and I set it type=user and stick it in a specific context, I ought to be able to put an entry in that context in my dialplan to like for instance play an IVR back, right? |
| 22:17.57 | b11d | please..please get out of my head |
| 22:17.58 | [E] | im doing something similar, but i don't have a fiber optic line |
| 22:18.04 | [TK]D-Fender | [F] : At which point you can let either * or the SPA determine which calls to dump to the PSTN |
| 22:18.29 | [F] | [TK]D-Fender: i have fiber optics, so is VoIP probably a better idea then landline? |
| 22:18.47 | [TK]D-Fender | [F] : Depends what you want, and what you already have. |
| 22:19.40 | [TK]D-Fender | [F] : For instance, I'm a * consultant, and I get my PSTN connectivity through my day-job as well. I have no analog lines, just dry-line DSL. So for me VoIP is the way to go. Save me money |
| 22:19.52 | [E] | i need to get this box connected to lan |
| 22:19.52 | [E] | brb |
| 22:21.33 | hohum | so annoying |
| 22:22.13 | [TK]D-Fender | hohum : Pretty much |
| 22:23.38 | b11d | welp.. im rolling out for the weekend. Take care lads, and have an enjoyable weekend.. even you jart :) |
| 22:25.53 | hohum | k |
| 22:25.56 | hohum | well it doesn't work |
| 22:26.06 | hohum | Asterisk keeps responding with a 484 Address Incomplete |
| 22:26.13 | mercestes | Is ChanIsAvail totally broken?? |
| 22:26.19 | hohum | its not even doing what the documentation suggests it does if it can't find a peer |
| 22:26.35 | hohum | and that would be to dump it into the context listed in [general] of the sip.conf |
| 22:26.41 | hohum | its just plain not working and I don't know why |
| 22:27.39 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 22:27.40 | *** mode/#asterisk [+o anthm] by ChanServ |
| 22:28.20 | hohum | hmm |
| 22:30.11 | hohum | fucking A |
| 22:30.24 | GaVak | ? |
| 22:30.50 | hohum | this used to work |
| 22:31.20 | *** join/#asterisk Tili (n=tili@202.133.65.50) |
| 22:37.28 | mercestes | Ok, what I would like to do is provide users a recording that says "user is on the phone" when a user is on the phone *instead* of presenting them with call waiting or a busy signal. |
| 22:37.53 | mercestes | What method should I use to track to see if a phone is in use or not? We don't want to turn Call Waiting off (sighs). Is there a way to do this?? |
| 22:39.30 | [TK]D-Fender | mercestes : "show application chanisavail" |
| 22:40.14 | mercestes | I've been doing that but whenever a channel is "in use" it simply terminates with a non-zero code. |
| 22:40.36 | mercestes | I have a ChanIsAvail(sip) and a NoOp(${AVAILSTATUS} but it never gets to the NoOp if the channel is in use. |
| 22:40.40 | mercestes | On Asterisk 1.2.13. |
| 22:41.43 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 22:42.11 | [TK]D-Fender | mercestes : Show me exactly how you are calling it. |
| 22:42.22 | mercestes | 3 lines ok? |
| 22:42.25 | mercestes | Wait, i can type it. |
| 22:42.41 | mercestes | exten => s,1,ChanIsAvail(${Arg1}|s) |
| 22:43.04 | *** join/#asterisk beefus (n=soup@dns2.cascom.ca) |
| 22:43.05 | mercestes | exten => s,n,NoOp(Status for ${Arg1} is ${AVAILSTATUS}) |
| 22:43.06 | [TK]D-Fender | mercestes : Pastbin the whoe deal. |
| 22:43.11 | mercestes | ALright. |
| 22:43.49 | beefus | my asterisk will not start on boot, i get a start/stop loop with error: Automatically restarting Asterisk. Asterisk ended with exit status 1 Asterisk died with code 1 |
| 22:44.11 | beefus | but when I login as root to console and run asterisk -vvvgc it works fine |
| 22:44.39 | [TK]D-Fender | beefus : Let me guess... got zaptel cards? |
| 22:44.55 | mtgh | Can someone help me with some SIP debugging, I have a phone that won't register, and it might be because of NAT but I am not sure http://pastebin.ca/236621 |
| 22:45.58 | mercestes | http://pastebin.ca/236623 |
| 22:47.49 | [TK]D-Fender | mercestes : And could you pastebin the call attempt.... |
| 22:48.40 | beefus | yes |
| 22:48.47 | beefus | I have a zaptel te110p |
| 22:49.14 | [TK]D-Fender | beefus : I'd be betting that zaptel isn't loading before * does. |
| 22:49.18 | mercestes | at the bottom of http://pastebin.ca/236629 |
| 22:49.56 | beefus | ya, I moved zaptel priority up to s02 and asterisk to s97 |
| 22:50.32 | [TK]D-Fender | mercestes : Notice something wrong here? - -- Executing ChanIsAvail("SIP/phone2-bc417db0", "SIP/phone1|sj") in new stack |
| 22:50.39 | [TK]D-Fender | mercestes : Look closely. |
| 22:51.07 | mercestes | I got the same issue without the "sj" I was attempting to bypass it by enabling priority jumping. |
| 22:51.20 | mercestes | it works if the phone is available and not busy. |
| 22:51.42 | [TK]D-Fender | mercestes : Well you're showing me apples& oranges. How about you just applyt he changes without the "j" ok? |
| 22:51.45 | mercestes | if you want I can reinstate the code I posted and submit a call without the "j" flag..or am I missing something else?? |
| 22:51.50 | mercestes | *nods* |
| 22:52.06 | [TK]D-Fender | mercestes : then repaste the deal for me :) |
| 22:54.20 | mercestes | http://pastebin.ca/236636 |
| 22:56.08 | [TK]D-Fender | mercestes : Perhaps you should disable priority jumping globally.... |
| 22:58.05 | *** join/#asterisk bkw__ (n=ASSERTKI@adsl-64-149-40-112.dsl.tul2ok.sbcglobal.net) |
| 22:58.16 | *** join/#asterisk oreonixon (n=oreonixo@63.76.221.162) |
| 23:00.41 | hoobastooba | i have a nuisance issue i need advice for. if two phones which are set up on a queue as ringall, answer a call at the exact same time, one phone continues to display the inbound call and ring while the other person actually takes the call. Its hard to describe... is there a way to correct this? |
| 23:01.10 | mercestes | [TK]D-Fender: same result..:( |
| 23:03.12 | [TK]D-Fender | mercestes : try hard-numbering your priorities, and you're sure you have "priorityjumping=no set global? |
| 23:03.28 | mercestes | [globals] |
| 23:03.34 | mercestes | priorityjumping=no |
| 23:04.14 | *** join/#asterisk alerios (n=alerios@190.24.97.148) |
| 23:04.20 | mercestes | Same result. |
| 23:04.57 | mercestes | I am dealing with "simulated" use here. This activity occurs when I am in a Meetme room, and right now I am putting the call "on hold" by dialing a main menu number. I woulnd't think that would effect ChanIsAvail() tho. |
| 23:05.04 | *** join/#asterisk Eliran_Itzhak (n=eliran@bzq-82-81-217-50.cablep.bezeqint.net) |
| 23:06.44 | [TK]D-Fender | mercestes : hrm. something is not right. ahrd-number them, and add a 103 priority |
| 23:06.51 | [TK]D-Fender | hard* |
| 23:06.52 | mercestes | *nods* |
| 23:08.04 | bkw__ | Dum dum de dum |
| 23:08.13 | mercestes | hard numbered, with priority 103, same result..:( |
| 23:08.38 | mercestes | Should I repastebin my calls to this macro with a successful detection and an unsucessful detection? |
| 23:08.45 | mercestes | sa long as the phone is not in use it behaves predictably. |
| 23:09.50 | [TK]D-Fender | mercestes : Must say I can't see this failing like that... |
| 23:09.58 | mercestes | Me neither. |
| 23:10.08 | [TK]D-Fender | mercestes : I used it much like you did for local channel agents as well. |
| 23:10.35 | mercestes | The only thing I can think is that it's not on an actual call...just in a recorded message, or on hold, or in a conference on the same server.... |
| 23:10.41 | mercestes | but...again, I don't think that should break it. |
| 23:12.39 | mercestes | should I.....post the calling code with the code and examples of a successful code execution v/s unsuccessful maybe?? |
| 23:13.37 | [TK]D-Fender | mercestes : dunno..... |
| 23:14.07 | [TK]D-Fender | mercestes : If you've covered all your priority based potential problems then I don't knwo what to suggest. |
| 23:14.27 | [TK]D-Fender | mercestes : though I'd like to see it. |
| 23:14.59 | mercestes | *nods* Give me a sec. |
| 23:21.06 | Un1x | Meh now i just need to learn how to play with crontab :P |
| 23:21.21 | Un1x | ive never used it before in my life but now i need it to backup my confs every week :) |
| 23:21.23 | Un1x | or every 2 weeks |
| 23:21.34 | mercestes | http://pastebin.ca/236662 |
| 23:21.57 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 23:24.14 | zamba | how do i change the default location for recorded conferences? |
| 23:25.56 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.71.53) |
| 23:26.50 | *** join/#asterisk asdx (n=diego@200.61.236.33) |
| 23:26.55 | asdx | hi |
| 23:27.46 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
| 23:27.52 | *** part/#asterisk hoobastooba (n=ckwall@63.149.122.93) |
| 23:27.59 | anonymouz666 | russellb |
| 23:28.07 | anonymouz666 | the GUI is beautiful |
| 23:28.15 | anonymouz666 | Ast-GUI |
| 23:28.47 | mitcheloc | anonymouz666: link? |
| 23:29.14 | anonymouz666 | http://www.asterisk.org/node/111 |
| 23:29.57 | anonymouz666 | what about? |
| 23:31.25 | [TK]D-Fender | mercestes : ashould be 102 actually |
| 23:31.43 | Un1x | :O this gui is supported by asterisk unlike freepbx |
| 23:31.43 | mercestes | Do you think it will make a difference? |
| 23:31.47 | [TK]D-Fender | mercestes : Although i have to head out for a bit. Keep up on it and let me know. |
| 23:31.55 | mercestes | *nods* Alright. |
| 23:32.02 | [TK]D-Fender | mercestes : 103 = useless. If its going to jump, it'd be to 102 |
| 23:32.09 | asdx | 1~0 m, |
| 23:32.58 | Un1x | fuck that looks nice |
| 23:33.01 | Un1x | im going to install it to |
| 23:33.03 | Un1x | lol |
| 23:33.15 | [TK]D-Fender | Yay * GUI = ABE + Web GUI..... |
| 23:33.32 | asdx | a friend is asking me if he can call from asterisk to a normal telephone (non voip) to any part in the world |
| 23:33.35 | *** join/#asterisk xAD (n=xAD@host18-137-static.107-82-b.business.telecomitalia.it) |
| 23:33.50 | mitcheloc | it's not fully functional yet right? |
| 23:34.00 | Un1x | mitcheloc the GUI? |
| 23:34.07 | mitcheloc | yes |
| 23:34.22 | Un1x | dont know first i heard of it i'm going to try it too :) |
| 23:34.43 | Un1x | looks real nice man |
| 23:36.50 | Un1x | but i still wonder if it wil be hard to install and stuff |
| 23:37.09 | Un1x | probably not :P |
| 23:37.17 | Un1x | anywya i'm off cya guys |
| 23:37.35 | asdx | it's possible to call from asterisk to a normal telephone (non-voip) to any part in the world? |
| 23:38.12 | zamba | asdx: with the right hardware added, sure |
| 23:38.37 | zamba | asdx: but not for free, if that's what you're asking :) |
| 23:39.28 | asdx | zamba: not for free? |
| 23:39.36 | zamba | no, why should it be? |
| 23:40.20 | *** join/#asterisk Blanker (n=piovrd@CPE-203-144-23-68.dsl.OntheNet.net) |
| 23:40.57 | Blanker | can anyone help with a te110p irq sharing issue |
| 23:41.35 | asdx | zamba: what do i need to do that? |
| 23:41.47 | asdx | zamba: and why is not free? |
| 23:45.10 | Un1x | nothing is free is the way the goverment wants it only thing free is |
| 23:45.29 | Un1x | well except for programs like asterisk and such wich is open source |
| 23:46.00 | GaVak | is there an easy way to set up reverse transfer of calls? Call parking isn't very efficent. |
| 23:46.13 | justinu|laptop | Blanker: what's up? |
| 23:47.04 | asdx | what do you need then for calling from asterisk to a normal telephone? |
| 23:47.27 | zamba | when trying to join a conference room i get prompted for my name each time.. why is that and how do i disable the feature? |
| 23:49.52 | Blanker | justinu: i have a card which is missing a lot of irqs and is sharing the same irq as the network card. whenever i disconncet from a remote session it kills asterisk |
| 23:50.29 | asdx | do you need drivers in the kernel for the cards? |
| 23:51.40 | GaVak | Blanker: Did you try putting the card in a different PCI slot? |
| 23:51.49 | justinu|laptop | Blanker: ok, you need to fix it... try moving the card to a different slot and cat /proc/interrupt |
| 23:51.55 | justinu|laptop | interrupts |
| 23:52.20 | Blanker | i cant move the card as im not physically next to the machine |
| 23:52.53 | Blanker | i though that might be the case. is there a way to automatically restart asterisk if it stops |
| 23:53.02 | justinu|laptop | check the safe_asterisk script |
| 23:53.03 | Blanker | until i can get to the box to move the card |
| 23:57.02 | *** join/#asterisk Pumas (i=KAos@148.244.74.235) |