| 00:01.56 | WIMPy | You did configure --with-asterisk? |
| 00:02.43 | aliverius | no but it did create chan_lcs.so |
| 00:02.47 | aliverius | no but it did create chan_lcr.so |
| 00:03.05 | WIMPy | Hmm. Didn't think that was an auto detect one. |
| 00:07.34 | aliverius | is testing his package |
| 00:09.51 | *** join/#asterisk Wiretap_Work (~Wiretap@unaffiliated/wiretap) |
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| 00:15.01 | aliverius | the package seems ok but breaks one rule. archlinux does not permit files to be installed in /usr/local |
| 00:15.54 | aliverius | conf files should go to etc |
| 00:16.00 | aliverius | that is fixable |
| 00:16.09 | aliverius | ringtones where? in shared? |
| 00:17.11 | *** join/#asterisk ks3 (~ksandy@cpe-184-57-144-100.cinci.res.rr.com) |
| 00:17.11 | WIMPy | Err, yes. The relevant parts are commented out in th makefile. |
| 00:17.40 | WIMPy | I've got no idea why that's fixed to /usr/local/. |
| 00:18.04 | WIMPy | But you can swap those 4 lines around. |
| 00:20.21 | WIMPy | Look for CONFIGdir. |
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| 00:25.17 | aliverius | i shall look at it tomorow. what matters the most is that i have lcr and chan_lcr. i wish i make another phone ring even if i cannot talk with it |
| 00:25.46 | WIMPy | Err, what? |
| 00:25.57 | WIMPy | also needs some sleep. |
| 00:26.09 | aliverius | i want to see my card doing something |
| 00:26.11 | aliverius | anything |
| 00:26.19 | aliverius | just a ring |
| 00:27.00 | *** join/#asterisk vfabi (~fabi@host-static-188-237-249-35.moldtelecom.md) |
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| 00:27.30 | WIMPy | If you have the driver loaded, it's just one entry in interface.conf and one in routing.conf. |
| 00:29.38 | aliverius | so help me God (you) :p |
| 00:29.42 | aliverius | goodnight |
| 00:29.45 | aliverius | rest well |
| 00:29.55 | aliverius | and thanks for all that help |
| 00:30.14 | WIMPy | Will do later, if you haven't got it running by then :-) |
| 00:45.02 | sawgood | DND from Asterisk is much different than 'pushing' the DND button on most phones (I found out how to deterime who is on DND from the CLI) ... |
| 00:45.03 | sawgood | thanks! |
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| 00:51.09 | sawgood | If one wanted to know what the current setting is for something like "maxsilence" for the voicemail module, but maxsilence is not 'set' under the [general] context ... how would I find out the current setting (which I guess would be the default at this point) |
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| 01:58.10 | fang0654 | Anyway to see which zap channel is ringing when I dial a number? |
| 01:58.36 | fang0654 | err dahdi channel |
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| 02:06.01 | nny | <feature feature.1.name="presence" feature.1.enabled="1"> <--- anyone know what XML tree this should be put under for polycom 3.3.1? |
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| 04:20.02 | deltaray | Can I make a GotoIf go straight to another extension? I tried GotoIf($["${CALLERID(num)}" = "....."],555), but that doesn't work. |
| 04:23.29 | kaldemar | deltaray: ?extension,priority |
| 04:23.50 | deltaray | ok, i see now. thx |
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| 06:01.59 | _zoom_ | hey fellas, am looking for a reliable free stun server? |
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| 06:06.31 | zkn | woopiedoo 1.8.4 segmentation fault again.. |
| 06:07.00 | zkn | once i rename astdb, problem soled |
| 06:07.08 | zkn | solved* |
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| 06:29.22 | zkn | does anyone know how to fix the following: ERROR[1565]: netsock2.c:245 ast_sockaddr_resolve: getaddrinfo("barbara", "(null)", ...): Name or service not known |
| 06:29.49 | zkn | looks like it is not able to resolve my hostname |
| 06:30.08 | zkn | whereas nlsookup in bash works fine |
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| 06:36.12 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 06:36.14 | schmidts | good morning |
| 06:36.28 | zkn | morning |
| 06:36.32 | WIMPy | Morning schmidts |
| 06:37.27 | schmidts | hey wimpy do you want to laugh a little bit ;) |
| 06:37.38 | zkn | ...ok my previous question related to /etc/resolv.conf afterall.. i think i got it right now |
| 06:37.39 | WIMPy | Sure. |
| 06:37.44 | WIMPy | What happened? |
| 06:38.01 | schmidts | just open amazon.de and search for "rabbit" (i search for a teddy from winni puh the rabbit) |
| 06:38.29 | schmidts | btw all others can do this too, but only amazon.at and .de will give this results .co.uk isnt that bad ;) |
| 06:39.38 | schmidts | i mean the first result is cheap but WTF this is amazon not beate uhse ;) |
| 06:39.39 | WIMPy | Might be time to point out that I'm living in germanys traditional sex toy capital :-) |
| 06:40.27 | WIMPy | Actually her first shop was just at the other end of the courtyard I'm living at. |
| 06:40.27 | schmidts | amazon the new online sex shop? get kindle to read porn or what? i am a little bit shocked about this. |
| 06:40.39 | schmidts | :D berlin right? |
| 06:40.46 | ChannelZ | http://www.amazon.com/Cloverdale-Fresh-Whole-Rabbit/dp/B00012182G |
| 06:40.55 | WIMPy | Flensburg |
| 06:41.39 | schmidts | ChannelZ thats bad too but in at and de we got this for the very first result http://www.amazon.de/Orion-557536-Perlen-Jelly-Vibrator-Rotation-Funktionen/dp/B000HKG15G/ref=sr_1_1?ie=UTF8&qid=1305787287&sr=8-1 |
| 06:41.49 | schmidts | Wimpy i see ;) |
| 06:41.51 | WIMPy | The interesting thin is that it also happened the other way round. Last year they tried to sell household stuff in the Beate Uhse Shop, like toasters. |
| 06:42.05 | schmidts | :D |
| 06:42.11 | schmidts | Sex sells! |
| 06:42.43 | WIMPy | Just don't use the toaster for the wrong thing. |
| 06:42.44 | ChannelZ | And amazon sells sex! |
| 06:43.23 | schmidts | ChannelZ for sure why not, but when i search for something normal like rabbit i didnt want to find such stuff or at least 8 hits out of the top 10 |
| 06:43.53 | schmidts | if i search for rabbit sex or something ok |
| 06:43.57 | ChannelZ | Don't search for watersports |
| 06:44.03 | schmidts | ? |
| 06:44.04 | WIMPy | LOL |
| 06:44.15 | ChannelZ | it's a very confusing category |
| 06:44.31 | schmidts | http://www.amazon.de/Micro-String-Tanga-Bikini-wei%C3%9F/dp/B003IMHFWI/ref=sr_1_15?ie=UTF8&qid=1305787458&sr=8-15 |
| 06:44.35 | schmidts | something like this? |
| 06:44.42 | WIMPy | never got teh "sport" bit of that. |
| 06:45.07 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-rqwufaejzctwoyhy) |
| 06:45.17 | schmidts | Wimpy think about lying on the beach and not getting an erection when seeing this, maybe this could be called also "sport" |
| 06:45.31 | ChannelZ | no, more like this http://www.amazon.de/dp/1596547804 |
| 06:45.54 | schmidts | OMFG |
| 06:45.54 | WIMPy | When I go swimming here, most people don't wear that much. |
| 06:46.18 | schmidts | needs a coffee NOW brb |
| 06:47.49 | WIMPy | What were the lyrics? "Happy hours, golden showers, on a cruise to freak you out"? |
| 06:47.55 | WIMPy | Seems to work for schmidts :-) |
| 06:51.00 | *** join/#asterisk bjornts (~BTS@it010226.klientdrift.uib.no) |
| 06:51.15 | WIMPy | In the time when I worked at a BU related company, I had to share my office with the guy who edited the hardcore videos. |
| 06:51.46 | WIMPy | The others found the moise of some test systems I had under my desk just as distubing as that. |
| 06:51.51 | WIMPy | LOL |
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| 07:03.38 | schmidts | Wimpy i was once at a customer installing some isdn/Voip atas and they had their pbx in the S&M storage room, that was funny working between some mask, handcuffs and such stuff :D |
| 07:05.34 | WIMPy | Well, that's what lots of ppl use their telefone for :-) |
| 07:08.36 | schmidts | hmm good idea, people allways cry about the cisco SPA phone handset is to heavy ;) |
| 07:08.52 | WIMPy | Err, no, the other way. |
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| 07:09.43 | schmidts | we should stop at this point, i didnt want to think about |
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| 07:10.32 | WIMPy | So what do you use Asterisk for then? |
| 07:14.03 | kaldemar | http://www.voip-info.org/wiki/view/Asterisk+Telemarketer+Torture |
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| 07:29.19 | schmidts | another question, what kind of database do you prefer to use with asterisk? |
| 07:29.49 | schmidts | i want to build a database cluster with fallback but i am not sure which method to use |
| 07:30.19 | schmidts | now we use mysql and we didnt want to change the whole configs only to switch to postgres or something |
| 07:30.49 | schmidts | mysql cluster is very nice but this thing is damn slow on Select statements, like 3 to 4 times slower than just a normal MyIsam DB and thats bad |
| 07:31.12 | schmidts | and it gets even slower if you access the same tables from different places at the same time |
| 07:32.25 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:d8ab:f308:5bf7:7598) |
| 07:35.16 | zkn | this is smth i would like to have too... |
| 07:35.38 | zkn | concept wise |
| 07:51.04 | *** join/#asterisk Emrah (~Me@unaffiliated/emrah) |
| 07:51.09 | Emrah | Hey all |
| 07:52.38 | Emrah | I am having an issue while trying to perform an attended transfer. Receiving call on SIP channel through SIP phone, and transfering to IAX extension (attended). The called IAX hears MOH and the transferee can hear IAX extension fine. Here is the Asterisk CLI output: handle_request_invite: Unable to create/find SIP channel for this INVITE |
| 07:53.00 | Emrah | I can't find anything relevant with our Google friend |
| 07:53.41 | Emrah | Any idea? |
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| 08:18.33 | ChannelZ | dunno off hand but I wonder if it's a directmedia issue, if it's trying to do a reinvite in spite of the differing channel types |
| 08:20.22 | ChannelZ | but alas I shal stop speculating and go off to bed |
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| 09:23.43 | moodyy | is there a way to terminate a channel that is chanspying a call? |
| 09:24.29 | moodyy | or if the spyed channels are terminated the channel that was spying is terminated also |
| 09:24.57 | *** join/#asterisk timahvo1 (~rogue@41.223.57.73) |
| 09:25.42 | zkn | hangup request <tab> does it show up that way? |
| 09:25.57 | moodyy | no |
| 09:27.00 | moodyy | i originate a call to a extension and execute the chanspy, but whem the spyed call end the spying channel continues connected |
| 09:27.13 | zkn | channel request hangup <tab> ? |
| 09:27.57 | zkn | what deos "core show channels verbose" give you ? |
| 09:28.25 | moodyy | this as to be done in the dialplan |
| 09:36.28 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 09:44.09 | kaldemar | moodyy: use option E for app ChanSpy. |
| 09:44.58 | kaldemar | core show application ChanSpy will show you that among the rest of the options. |
| 09:46.37 | puzzled | hi |
| 09:47.13 | wdoekes2 | does anyone hazard a guess as to why my app_fax loads spandsp from /usr/lib instead of the one in /usr/local/lib? ldd shows that the one in local should get precedence, but lsof shows the one in /usr/lib as loaded |
| 09:50.22 | puzzled | wdoekes2: no idea if that's possible but maybe it linked against the spandsp lib in /usr/lib when you built it so it wants to use that one? |
| 09:51.33 | wdoekes2 | nah.. on my test machine it safely switches between the two (where /local is preferred) |
| 09:52.30 | wdoekes2 | the only thing I can thinks of is that LD_LIBRARY_PATH is less complete in the live version (asterisk is started from cron) |
| 09:53.11 | puzzled | iirc you can preload a lib so asterisk will use that one. have you tried that? |
| 09:53.21 | WIMPy | did you ldconfig? |
| 09:53.53 | wdoekes2 | yes I did |
| 09:54.48 | wdoekes2 | haven't tried preloading.. but I cannot to much on this live box to disturb it |
| 09:55.34 | puzzled | LD_PRELOAD=/usr/local/lib/libspandsp.so asterisk |
| 09:56.57 | puzzled | coppice: what's the preferred version of spandsp as a prereq for asterisk 1.8? |
| 10:01.15 | WiretapSeven | I do wish asterisk wouldn't forget its SIP/TCP peers on reload |
| 10:02.50 | wdoekes2 | WiretapSeven: tried realtime? |
| 10:03.20 | WiretapSeven | wdoekes2, eh, why would I? |
| 10:03.30 | WiretapSeven | my SIP/UDP peers work find |
| 10:03.31 | WiretapSeven | fine* |
| 10:03.40 | WiretapSeven | my SIP/TCP peers however get forgotten |
| 10:04.00 | wdoekes2 | ok, never mind then |
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| 10:22.34 | moodyy | the chanSpy E option is aveilable in asterisk 1.6 ? i dont see this option qhen i run core show application CahnSpy |
| 10:22.43 | moodyy | Sorry ChanSPY |
| 10:23.49 | *** part/#asterisk rajmohan (~rajmohan@122.164.182.89) |
| 10:24.02 | wdoekes2 | I don't see it in the source of 1.6.2.x either |
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| 10:24.47 | moodyy | is this only available in asterisk 1.8? |
| 10:28.07 | kaldemar | yes, it is a 1.8 option. |
| 10:35.50 | aliverius | is it normal to have hisax and misdn modules loaded at the same time? |
| 10:37.42 | aliverius | how can i troubleshoot if misdn works? for example see that i have an active d channel? |
| 10:38.29 | aliverius | also a question about cabling. does connecting with an ethernet cable to the NT work? |
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| 10:38.56 | aliverius | WIMPy: goodmorning. take a look above when you can, please |
| 10:40.15 | aliverius | i dont know even if my connection works |
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| 10:49.01 | aliverius | http://pastebin.com/nDueFgq4 this means the side of the computer is fine right? |
| 10:49.03 | coppice | puzzled: the latest is the preferred version :-) |
| 10:49.26 | puzzled | coppice: thanks |
| 10:52.46 | coppice | puzzled: Use 0.0.6pre18, and tell me if you have any problems. I see various things that suggest spandsp may not be working as well as the Digium FAX module with 1.8.x, but nobody seems to follow up to resolve this. spandsp should work better than the Digium FAX module. it does with older versions of asterisk |
| 10:53.21 | puzzled | coppice: ok. will let you know if I see something odd |
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| 11:27.11 | *** join/#asterisk GreatSUN (~greatsun@178-190-215-168.adsl.highway.telekom.at) |
| 11:27.15 | GreatSUN | hi all |
| 11:28.34 | GreatSUN | can someone help me with asterisk 1.8 in combination with iax2, dahdi and hylafax, please? |
| 11:28.54 | GreatSUN | incoming faxes are working out without problems |
| 11:29.30 | GreatSUN | with outgoing I get all circuits busy, but dialing seams to show up correctly in debug |
| 11:30.00 | GreatSUN | Executing [s@macro-dialout-trunk:20] Dial("IAX2/300-1458", "DAHDI/g0/0800664648,300,") in new stack |
| 11:31.34 | GreatSUN | when dialing with my sip-phone I get the fax on the other side without problems |
| 11:40.15 | *** join/#asterisk guilhermebr (~Guilherme@200.103.96.98) |
| 11:44.47 | GreatSUN | are you all having lunch? |
| 11:44.55 | GreatSUN | noone in here alive? |
| 11:46.06 | wdoekes2 | you forgot to leave after complaining about a too slow response ;) |
| 11:46.44 | GreatSUN | lol |
| 11:47.05 | GreatSUN | wdoekes2, I will not leave until my provider cuts internet line again |
| 11:47.17 | GreatSUN | which happens every 8 hours |
| 11:47.19 | GreatSUN | :-D |
| 11:47.38 | GreatSUN | wdoekes2, but I guess you can't help me with my problem, can you? |
| 11:48.01 | wdoekes2 | I don't know hylafax nor any other channel driver than sip |
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| 11:58.02 | *** join/#asterisk marlowe (~marlowe@static-72-66-8-138.washdc.fios.verizon.net) |
| 12:01.15 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 12:03.34 | GreatSUN | I will set an idle point... |
| 12:03.35 | GreatSUN | . |
| 12:04.02 | GreatSUN | as noone seams to communicate in here atm |
| 12:05.09 | *** join/#asterisk dwayne (~dwayne@75-150-14-49-Atlanta.hfc.comcastbusiness.net) |
| 12:06.40 | *** join/#asterisk marlowe (~marlowe@static-72-66-8-138.washdc.fios.verizon.net) |
| 12:07.10 | *** join/#asterisk fhmiv (~fhmiv@c-67-173-205-151.hsd1.ga.comcast.net) |
| 12:08.24 | kaldemar | GreatSUN: i'd start by checking that g0 (group=0 in chan_dahdi.conf) is configured with correct channels. |
| 12:09.17 | GreatSUN | kaldemar, even though call through sip-phone works? |
| 12:10.00 | kaldemar | GreatSUN: SIP->DAHDI? |
| 12:12.36 | *** join/#asterisk marlowe (~marlowe@static-72-66-8-138.washdc.fios.verizon.net) |
| 12:12.38 | GreatSUN | kaldemar, exactly |
| 12:12.43 | GreatSUN | SIP->DAHDI works |
| 12:12.50 | GreatSUN | IAX2->DAHDI doesn't |
| 12:12.57 | GreatSUN | DAHDI->SIP works |
| 12:13.07 | GreatSUN | DAHDI->IAX2 works |
| 12:14.45 | kaldemar | GreatSUN: what does a working call through DAHDI look like? |
| 12:14.58 | kaldemar | i.e. SIP->DAHDI |
| 12:15.30 | GreatSUN | you mean that: |
| 12:15.31 | GreatSUN | Executing [s@macro-dialout-trunk:20] Dial("SIP/100-000000d8", "DAHDI/g0/4569,300,") in new stack |
| 12:17.01 | kaldemar | is 0800664648 a valid number? |
| 12:17.44 | kaldemar | what kind of a DAHDI interface is it? analog? PRI? BRI? |
| 12:20.30 | GreatSUN | kaldemar, as I called exactly the same number with my sip-phone through exactly the same dahdi channel, the number as well as the dahdi channel are working properly |
| 12:20.44 | *** join/#asterisk zorp75ck (~zorp75ck@146.186.115.103) |
| 12:21.26 | *** join/#asterisk fish-bulb (~qcstewart@nat/digium/x-wtfhyysbzmqqbzxc) |
| 12:21.33 | GreatSUN | and the dahdi interface: quadBRI PCI ISDN Card 1 Span 2 [TE] (cardID 4) Layer 1 ACTIVATED (F7) |
| 12:23.38 | *** join/#asterisk glam (~glam@113.97.3.196) |
| 12:25.26 | kaldemar | do you get a cause code in the protocol debug? iirc, pri intense debug span <span> |
| 12:33.32 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 12:33.43 | GreatSUN | kaldemar: http://nopaste.info/f94ef23fcc.html |
| 12:42.33 | GreatSUN | kaldemar: any idea? |
| 12:45.05 | m4xx | if i have Set(var=123) ... and i try to reference ${var:3:3} will that error or return an empty value? |
| 12:45.14 | schmidts | GreatSUN i am not sure but i see this: 1 < Ext: 1 Cause: Info. element nonexist or not implemented (99), class = Protocol Error (e.g. unknown message) (6) ] |
| 12:46.08 | schmidts | http://networking.ringofsaturn.com/Routers/isdncausecodes.php -> cause 99 |
| 12:46.16 | *** join/#asterisk rajmohan (~rajmohan@122.164.109.66) |
| 12:47.17 | *** join/#asterisk MarKsaitis (~MarKsaiti@195.59.185.18) |
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| 12:50.38 | *** mode/#asterisk [+o malcolmd] by ChanServ |
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| 12:59.55 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:01.18 | *** part/#asterisk andresmujica (~andresmuj@ubuntu/member/andresmujica) |
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| 13:25.30 | Khratos | Good morning |
| 13:26.00 | leifmadsen | o/ |
| 13:32.32 | *** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn) |
| 13:34.30 | *** join/#asterisk sbszulu (~dundubala@41.16.75.48) |
| 13:39.51 | *** join/#asterisk eharris (~eharris@99-179-7-82.lightspeed.austtx.sbcglobal.net) |
| 13:47.12 | zkn | umm...what could be the reason why I have needed to rotate astdb already two times this weeks to get Asterisk 1.8.4 to start up again ? |
| 13:47.50 | zkn | it's like the db gets corrupt for some reason |
| 13:49.56 | *** join/#asterisk kaushal (~kaushal@115.246.249.22) |
| 13:49.58 | kaushal | Hi |
| 13:50.26 | kaushal | Can someone please guide me the memory requirement of Asterisk Server ? |
| 13:51.08 | kaushal | is there a benchmark sort of done on Asterisk 1.8.4 Open Source PBX ? |
| 13:51.36 | kaushal | I have 2 Gigs of DDR3 RAM with 500 GB of Hard Drive Space |
| 13:51.43 | serafie | kaushal: depends on how you use it. I've got Asterisk running on one of these: http://www.tonidoplug.com |
| 13:52.35 | kaushal | but the developer who is porting his application on the Asterisk box demands for additional memory ? |
| 13:52.57 | kaushal | How do i question him ? |
| 13:53.09 | *** join/#asterisk Gugge (~gugge@vlan2.dlxhosting.dk) |
| 13:54.09 | kaushal | Is there a minimum requirement Specification for running Asterisk Box |
| 13:54.51 | kaushal | Basically we will be dialing out 320 Calls per min using Asterisk |
| 13:54.53 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:54.54 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:55.19 | leifmadsen | kaushal: the minimum is whatever asterisk starts with -- the actual requirements are totally dependent on what you're doing |
| 13:55.29 | jaytee | I want to use an Asterisk queue to "throttle" calls to an IVR on another system. The IVR uses Dialogic's HMP for SIP and can register to Asterisk. If I set a call-limit=24 for the 24 licensed ports on the IVR will a single persistent queue member be able to answer more than one call at a time? |
| 13:55.33 | leifmadsen | the "what" is extremely subjective |
| 13:55.57 | leifmadsen | jaytee: just use ringinuse=yes |
| 13:56.05 | kaushal | leifmadsen: ok |
| 13:56.24 | leifmadsen | jaytee: then I might just use a Local channel to call the other PBX, and use GROUP() and GROUP_COUNT() to track the simultaneous call thing |
| 13:56.26 | jaytee | leif, that's what I thought after reading that part of this incredibly rich and detailed book. |
| 13:56.30 | kaushal | leifmadsen: Any benchmark being done for Memory Usage ? |
| 13:56.45 | jaytee | and was just going to add that as part of my question. it looks like that's all I really need. |
| 13:56.51 | leifmadsen | ExecIf($[${GROUP_COUNT(ivr_limit)} > 25]?Congestion()) |
| 13:57.01 | leifmadsen | jaytee: ya, forget call-limit I think |
| 13:57.39 | jaytee | on another system I setup I had 4 licensed ports and had the IVR registering as 4 persistent queue members but setting it up for 24 of them would be a bit cumbersome |
| 13:57.43 | leifmadsen | kaushal: memory usage is really not an issue with asterisk -- if it loads, it's pretty much stable other than what gets written to memory for channel variabels and such (minimal data really overall), unless you have a memory leak. Asterisk is much more CPU intensive. |
| 13:58.11 | leifmadsen | jaytee: ya I think a single queue member (Local channel) with GROUP() and GROUP_COUNT() would work, then just use ringinuse=yes |
| 13:58.48 | kaushal | leifmadsen: ok |
| 13:58.56 | kaushal | leifmadsen: understood now |
| 13:59.01 | jaytee | leif, thanks! |
| 13:59.19 | kaushal | is there a wiki for it ? |
| 13:59.38 | serafie | kaushal: http://wiki.asterisk.org |
| 14:00.08 | serafie | more specifically, https://wiki.asterisk.org/wiki/display/AST/Home |
| 14:00.11 | leifmadsen | or http://ofps.oreilly.com (non-wiki) |
| 14:02.05 | kaushal | leifmadsen: I see you everywhere :) |
| 14:02.19 | kaushal | I feel great conversing with you |
| 14:03.27 | kaushal | so minimum can be as little as 512 MB of RAM ? |
| 14:03.46 | kaushal | if its fully loaded |
| 14:04.03 | leifmadsen | kaushal: sure |
| 14:04.17 | *** join/#asterisk dfamorato (~dfamorato@173-9-190-185-miami.txt.hfc.comcastbusiness.net) |
| 14:04.18 | leifmadsen | is The Great Gatsby |
| 14:04.34 | kaushal | The reason behind so many questions is to basically understand every bit of it |
| 14:06.13 | leifmadsen | pretty sure that is unattainable :) |
| 14:09.20 | *** join/#asterisk Sertys (~sertys@89.252.247.42) |
| 14:10.20 | *** join/#asterisk felimwhiteley (~quassel@109.255.104.145) |
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| 14:16.20 | *** join/#asterisk buzzyd (~buzzyd@server.asyouneed.net) |
| 14:16.45 | buzzyd | Hi all |
| 14:17.20 | jaytee | hi |
| 14:17.20 | buzzyd | I'm trying to use my reload script on asterisk 1.8 and I'm getting utils.c:1180 ast_careful_fwrite: fwrite() returned error: Broken pipe |
| 14:17.45 | buzzyd | It's the script that was on the wiki for accessing the manager via telnet |
| 14:17.45 | Nugget | telnet is eeeeeeevil! |
| 14:17.49 | buzzyd | :) |
| 14:18.10 | buzzyd | is there a better way I can call reload via php now? |
| 14:19.34 | GreatSUN | hey guys |
| 14:19.38 | Khratos | buzzyd: you could program a little script that sends request via HTTP, it's way more effective that using general sockets in php (as you have to read ALL asterisk response before the script ends, or you will have 'broken pipe') |
| 14:19.51 | GreatSUN | looks like we have knowledge people in here again :o) |
| 14:20.13 | GreatSUN | can somone of you try to help me with my IAX2->DAHDI problem? |
| 14:20.25 | buzzyd | any docs on this? |
| 14:20.31 | *** join/#asterisk davlefou (~david@41.225.228.149) |
| 14:20.40 | GreatSUN | calls incoming from DAHDI and directed to IAX2 -> hylafax are working properly |
| 14:21.08 | GreatSUN | but the other way round results in a "all circuits are busy" message |
| 14:21.15 | Khratos | buzzyd: yes, the asterisk source contains very good documentation on the subject (only what has to do with Asterisk), about the script using http with php, you can see very good info at php.net |
| 14:21.16 | GreatSUN | see here: http://nopaste.info/f94ef23fcc.html |
| 14:21.26 | Khratos | its just a matter of combining them |
| 14:21.29 | GreatSUN | (with pri intense debug) |
| 14:22.04 | Khratos | Ah!, and read the HTTP rfc, at least the initial parts to understand how to create an HTTP 'session' |
| 14:23.45 | buzzyd | Khratos: having a look for it now cheers, |
| 14:24.04 | Khratos | buzzyd: Basically, 1. you have to configure Asterisk to use its http server (see documentation, its very good), 2. Craft http headers with php, and write them to Asterisk using sockets (its quite simple, there are examples on php.net) |
| 14:24.44 | Khratos | 3. Asterisk will responde with a 'session_id' header that you will have to pass to it on every subsequent request (that is, after sending the 'login' action, with username and secret parameters -GET request-) |
| 14:25.21 | Khratos | you can save it using $_SESSION in php |
| 14:25.57 | Khratos | After that, you will feel in heaven managing Asterisk with HTTP protocol instead of generick sockets only |
| 14:27.34 | Khratos | A simple request might look like GET http://[your-asterisk-box]:[your-port]/[security-token]/rawman?action=status |
| 14:28.51 | Khratos | buzzyd: Asterisk will respond to that request with a 'plain text' (in that case, cause of the 'rawman' -see documentation- part), which you can filter with regexp (preg_match) in php |
| 14:29.52 | Khratos | You then can craft a nice array and conver it to JSON if you like, and have a powerfull php/javascript application that can do almost everything with Asterisk |
| 14:30.08 | Khratos | convert** (see: json_encode(), php) |
| 14:30.17 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 14:31.30 | buzzyd | Khratos: all sound very cool thank you... |
| 14:31.57 | Khratos | You are welcome |
| 14:33.19 | pabelanger | buzzyd: you usually see that message when an script loses it's connection to the AMI and Asterisk is trying to send something to it. |
| 14:34.09 | Khratos | yes, in that case, Asterisk will ignore changes (if you have told it to do something) after that |
| 14:34.49 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 14:35.18 | buzzyd | @pabelanger: Thank you but what would cause it to loose it's connection as script works fine on 1.4 |
| 14:35.59 | leifmadsen | lose* |
| 14:36.24 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:36.38 | Khratos | I faced that issue once, script was ignoring Asterisk response (at is had finish it execution) and Asterisk saw a 'broken pipe' |
| 14:37.12 | pabelanger | Is your AMI connection persistent? or do you Login / Logoff each time the script it run? Are you waiting for Asterisk to ACK the Logoff? |
| 14:37.38 | pabelanger | My guess, you send a command to asterisk then close the socket |
| 14:37.45 | pabelanger | when Asterisk is trying to send a reply back |
| 14:37.54 | pabelanger | then you get that error |
| 14:39.17 | Khratos | Yes, and things get worst if you try to continiously read from a socket in PHP. The script will hang there until you see a green cat |
| 14:40.03 | buzzyd | Login/Logoff its the following script http://pastebin.com/download.php?i=YegCjz30 |
| 14:41.30 | pabelanger | buzzyd: yes, so you are not waiting for Response: Success. Which asterisk is trying to send to the closed socket |
| 14:42.47 | WIMPy | aliverius: Hi. No, both hosax and misdn are not a good thing, but your output of misdn_info looks fine. |
| 14:42.59 | WIMPy | And yes, a standard patch cable is ok. |
| 14:44.04 | Khratos | I love php when dealing with web applications, but when dealing with sockets and stuff, it just hit a wall |
| 14:45.08 | pabelanger | Khratos: python + StarPy = Awesome :) |
| 14:46.08 | buzzyd | Something like http://pastebin.com/download.php?i=wCJjZ0tU |
| 14:46.18 | Khratos | I have heard a lot about python, and my searches on the topic made me conclude that it's second almost to none |
| 14:46.30 | *** join/#asterisk sigmounte (~sigmounte@sd-13198.dedibox.fr) |
| 14:46.40 | sigmounte | hi ! |
| 14:47.29 | sigmounte | i can see in my log that asterisk is playing MOH on my queue , but i can't hear anything ? any idea where i can start searching ( for precision , moh work on normal calls ) |
| 14:48.32 | sigmounte | ( Open Source Asterisk , everything freshly compiled from source with dahdi , meet me working ) |
| 14:48.32 | *** join/#asterisk zkn (~zkn@82.131.33.251.cable.starman.ee) |
| 14:51.07 | *** join/#asterisk davlefou (~david@41.225.135.188) |
| 14:58.27 | *** part/#asterisk ketema (~ketema@kjhmacpro.ketema.net) |
| 14:59.01 | *** join/#asterisk sigmounte (~sigmounte@sd-13198.dedibox.fr) |
| 15:04.14 | *** join/#asterisk wonderworld (~ww@port-92-201-149-166.dynamic.qsc.de) |
| 15:06.28 | *** join/#asterisk drift- (ae3001a4@gateway/web/freenode/ip.174.48.1.164) |
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| 15:14.02 | *** join/#asterisk cusco (~tralala@a89-152-96-250.cpe.netcabo.pt) |
| 15:14.04 | cusco | hi folks |
| 15:14.11 | cusco | I've got a little question |
| 15:14.41 | pabelanger | ~ask |
| 15:14.42 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 15:15.33 | cusco | yes |
| 15:15.34 | cusco | wait |
| 15:15.42 | cusco | I've got: -- Executing [sendFax@FAX:1] SendFAX("DAHDI/125-1", "/var/log/asterisk/fax/outbound/1305817896.documento1.tiff") in new stack |
| 15:15.52 | cusco | and then: [May 19 16:12:05] WARNING[3098]: app_fax.c:817 transmit: Transmission error |
| 15:15.55 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 15:16.26 | cusco | and it goes to the h extension and I NoOp some fax variables where status is SUCCESS and error is empty: -- Executing [h@FAX:4] NoOp("DAHDI/125-1", "Fax sent to 213825351 with 1 pages, 14400 bitrate, status: SUCCESS error: ) |
| 15:16.30 | Khratos | that infobot is a relief |
| 15:16.54 | cusco | ~infobot |
| 15:16.54 | infobot | from memory, infobot is in need of training, can someone train me? |
| 15:17.00 | cusco | hah |
| 15:17.13 | cusco | ~blootbot |
| 15:17.14 | infobot | well, blootbot is an IRC bot written in perl descended from infobot. Hosted on SF. This project is now merged back into the main infobot project. See [infobot] |
| 15:17.15 | WIMPy | hands infobot an infinite loop |
| 15:17.56 | cusco | ok well... I would like to know why the transmission error... |
| 15:20.26 | *** join/#asterisk errr (~errr@fedora/errr) |
| 15:22.27 | errr | In my dial plan I have a system command being executed but its not working correctly.. I have turned up the verbose and it is getting to that part of the dial plan and moving on, but the command does not appear to be working.. is there further debugging I can do from in asterisk to find out whats going on, or do I need to just debug in the script to see what up? |
| 15:22.59 | errr | If I run the command from the cli that it shows to have run from the dial plan it works.. just not from with in asterisk.. |
| 15:23.19 | aliverius | WIMPy: still, the qt interface doesnt show a green light, like if the NT and the card cannot see eachother |
| 15:23.52 | ChannelZ | Are you running it from the CLI as the same user your asterisk runs under? Perhaps it has no access. |
| 15:24.06 | errr | ChannelZ: ah good point, Ill try as the asterisk user |
| 15:24.10 | WIMPy | aliverius: which qt interface? do you have LCR running already? |
| 15:24.41 | errr | ChannelZ: yes that does work |
| 15:25.03 | WIMPy | aliverius: But no link can be normal, due to powersaving. |
| 15:25.13 | aliverius | WIMPy: there is a gui tool in misdnuser git |
| 15:25.19 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 15:25.34 | WIMPy | What's it called? |
| 15:25.39 | aliverius | WIMPy: it has indicators for D and 2 B channels. |
| 15:25.42 | aliverius | one moment |
| 15:25.53 | *** join/#asterisk Jcook_5xData (~Jcook_5xD@173.162.32.1) |
| 15:25.55 | ChannelZ | hmm |
| 15:26.15 | aliverius | WIMPy: qmisdnwatch |
| 15:26.46 | WIMPy | Ja, found that. |
| 15:27.09 | aliverius | WIMPy: do you think i should bypass this and just go test lcr? |
| 15:27.18 | WIMPy | yes |
| 15:27.26 | aliverius | ok |
| 15:27.44 | Jcook_5xData | need some help I randumly receive : chan_dahdi.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 then my pri reset happen about 3 times a day. running asterisknow, rhino t1 card with EC |
| 15:28.03 | WIMPy | ah, it's qt4 only. I'm still on 3.5 here. |
| 15:28.23 | aliverius | qt3? omg! |
| 15:28.34 | WIMPy | Jcook_5xData: Either you have a noisy line or IRQ processing troubles. |
| 15:29.06 | Jcook_5xData | how can I tell if IRQ problem |
| 15:29.50 | Jcook_5xData | is it as simple as setting bois to not pnp |
| 15:29.51 | Khratos | Jcook_5xData: maybe dahdi_tool could tell if there are IRQ signal missed |
| 15:32.31 | Jcook_5xData | stupid ? if I use dahdi_tool on a live system will the user have problems |
| 15:33.56 | buzzyd | Khratos: I can't find the http docs in the source code do you know where they are? |
| 15:35.33 | *** join/#asterisk chandoo (~chandoo@ool-4a5a2824.dyn.optonline.net) |
| 15:36.33 | *** join/#asterisk fireman_biff (~biff@65.48.133.102) |
| 15:37.41 | buzzyd | Atlernatively happy to pay someone to write a reload script that works in Asterisk 1.8 and can be used in php ;) |
| 15:38.23 | WIMPy | you need to make them |
| 15:38.32 | WIMPy | There's a not at the and of make. |
| 15:38.46 | *** join/#asterisk shaggy2 (~craig@unaffiliated/shaggy2) |
| 15:39.44 | *** join/#asterisk fullstop (~fullstop@static-173-210-91-4.saucontech.com) |
| 15:40.25 | shaggy2 | hello I keep getting Segmentation fault; asterisk ended with exit status 139; asterisk exited on signal 11; Automatically restarting asterisk |
| 15:40.35 | fireman_biff | Hi, I'm using asterisk 1.6.2.13 and the voicemail indicators no longer work, ie message light, different dial tone, message count on the phone display. If I disable then re-enable each extension one-by-one it works but I lose existing voicemail. Is there anything else I can try? |
| 15:41.13 | WIMPy | ~collectdebug |
| 15:41.13 | infobot | hmm... collectdebug is a method of collecting logs allowing others help troubleshoot an issue. Refer to https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 15:41.25 | WIMPy | shaggy2: ^^ |
| 15:41.37 | WIMPy | fireman_biff: That does not sound like an Asterisk question. |
| 15:42.22 | fireman_biff | WIMPy: what kind of question does it sound like? you think its the phones? |
| 15:42.24 | *** join/#asterisk moy (~moy@CPE002719f00364-CM0026f3a1701d.cpe.net.cable.rogers.com) |
| 15:42.56 | WIMPy | No, it's sounds like some GUI f***up. |
| 15:42.58 | shaggy2 | ok note my error please I can not use asterisk so that reference to me is pointless |
| 15:43.34 | WIMPy | shaggy2: No, it's the way to find out, why. |
| 15:43.55 | shaggy2 | ok to use that I need to get into the asterisk CLI |
| 15:43.59 | shaggy2 | correct? |
| 15:44.05 | fireman_biff | how would a graphical interface break voicemail thats been working fine? |
| 15:44.29 | WIMPy | shaggy2: No, just the shell. |
| 15:44.31 | *** join/#asterisk timahvo1 (~rogue@41.223.57.72) |
| 15:45.13 | WIMPy | fireman_biff: No idea, but how is "disabling" and "re-enabling" mailboxes going to fix a bug? |
| 15:45.43 | *** part/#asterisk rajmohan (~rajmohan@122.164.12.3) |
| 15:48.48 | Khratos | sorry buzzyd , I was AFK. If you dowload asterisk source, inside the 'doc' directory (I think, or inside a subdirectory); do 'make pdf' |
| 15:49.53 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 15:50.03 | *** join/#asterisk m_tadeu (~quassel@89.181.11.84) |
| 15:50.03 | WIMPy | aliverius: Just tried qmisdnwatch on my netbook. Have you tried device actions -> connect layer 1? |
| 15:50.19 | aliverius | yes |
| 15:50.30 | aliverius | doesnt give me a green light |
| 15:50.32 | WIMPy | That didn't work? |
| 15:51.10 | WIMPy | Hmm. that might not be good. |
| 15:51.52 | aliverius | possibly... |
| 15:52.03 | aliverius | NT problem? |
| 15:52.28 | WIMPy | Do you have anything else to connect to the NT? |
| 15:54.06 | aliverius | no isdn device |
| 15:54.20 | Jcook_5xData | if I run dahdi_tool on a live system with call going on will it cause a problem? |
| 15:54.24 | aliverius | pots and usb work |
| 15:54.40 | aliverius | i mean it is alive |
| 15:55.03 | WIMPy | aliverius: USB? What kind of device is that? |
| 15:57.04 | aliverius | it is a device that connects to a U-line and provides 2 S buses 2 pots one srial and one usb to work as a modem |
| 15:57.41 | WIMPy | Interesting thni. Haven't heard of that combination before. |
| 15:58.05 | WIMPy | It it really an NT or is it an NGN IAD? |
| 15:58.27 | aliverius | http://www.intracom-telecom.com/helpdesks/netmod/downl/netmod_usb_manual.zip |
| 15:58.31 | aliverius | what is the second? |
| 15:58.43 | aliverius | googles |
| 15:58.58 | WIMPy | Something that acually works via IP. |
| 15:59.08 | *** join/#asterisk Defraz (~Defraz@tim.spudnik.com) |
| 16:00.10 | aliverius | no no, it is an NT with extra fucntionality |
| 16:00.21 | aliverius | there is a simple NT version |
| 16:00.29 | aliverius | one that provides 2 pstn lines |
| 16:00.35 | aliverius | and one that has it all |
| 16:00.49 | WIMPy | Ok. I've seen NTs with POTS, but not with "modem". |
| 16:00.58 | aliverius | it is standard with greek isdn |
| 16:01.58 | *** join/#asterisk gmaruzz (~gmaruzz@static-217-133-80-112.clienti.tiscali.it) |
| 16:02.38 | *** join/#asterisk cyford (Technologi@adsl-074-188-021-230.sip.asm.bellsouth.net) |
| 16:03.13 | aliverius | nobody buys isdn "modems" here |
| 16:03.14 | *** join/#asterisk Hail2theKingBaby (~Hail2theK@194-176-87-181.onyx.net) |
| 16:03.31 | aliverius | how should i call it |
| 16:03.32 | fullstop | Good Afternoon! |
| 16:03.42 | aliverius | hi fullstop |
| 16:03.45 | WIMPy | Terminal Adaptor |
| 16:04.09 | aliverius | thanks for the hint ;) |
| 16:04.20 | fullstop | aliverius: Good evening, if you are in Greece. |
| 16:04.35 | aliverius | :D |
| 16:04.37 | aliverius | ty |
| 16:07.05 | Hail2theKingBaby | HI there. I've busy reading Asterisk - The Future of Telephony... is this the best place to start with Asterisk? |
| 16:07.34 | Hail2theKingBaby | I'm only up to chapter 3 but already it's given me a great insight into what Asterisk is and isn't |
| 16:08.14 | aliverius | is wondering what to do to troubleshoot his connection |
| 16:08.21 | buzzyd | khratos: no joy, I've got the latest source and can't find any relevant info in doc folder and make pdf does nothing. Arg :) |
| 16:08.21 | WIMPy | It's certainli a good start. |
| 16:08.30 | kaldemar | Hail2theKingBaby: you're on the right track |
| 16:08.38 | kaldemar | ~book |
| 16:08.38 | infobot | For more information about the Asterisk book, see ~thebook |
| 16:08.46 | kaldemar | ~thebook |
| 16:08.46 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 16:09.00 | kaldemar | there's a newer version too |
| 16:09.13 | *** join/#asterisk cyborg-one (1000@85-238-116-105.broadband.tenet.odessa.ua) |
| 16:09.16 | aliverius | ~solvemyproblem |
| 16:09.21 | errr | lol |
| 16:09.27 | errr | nice try aliverius |
| 16:09.37 | aliverius | this bot sucks |
| 16:09.51 | aliverius | my setup sucks too |
| 16:10.02 | aliverius | i probably suck as well |
| 16:10.06 | *** join/#asterisk sereal (~jjrh@2001:4830:16ca:1:21a:6bff:fe6a:3cd2) |
| 16:10.09 | Hail2theKingBaby | Ah so is The Definitive Guide a new version of The Future of Telephony or a completely different and newer publication? |
| 16:10.17 | WIMPy | aliverius: Hmm. Nothing else to test is rather unfortunate. |
| 16:10.26 | sereal | Has anyone used the pyst python asterisk manager interface? |
| 16:11.01 | aliverius | WIMPy: maybe i will buy a cheap isdn phone |
| 16:12.09 | WIMPy | aliverius: Nothin you could borrow perhaps? |
| 16:12.21 | WIMPy | Or another card? |
| 16:12.31 | aliverius | only another NT |
| 16:12.37 | aliverius | different revision |
| 16:13.21 | WIMPy | wouldn't expect the NT to fail, but then I wouldn't expect other things to fail, either. |
| 16:13.41 | *** join/#asterisk Russ (foobar@ip70-176-251-1.ph.ph.cox.net) |
| 16:13.45 | WIMPy | But I have to admit that I once had a HFC-S card fail. |
| 16:13.57 | Russ | I get weird timestamps from an IAX service |
| 16:14.22 | fullstop | Hail2theKingBaby: I believe that it is a different publication. |
| 16:15.00 | Hail2theKingBaby | ah ok I'll look it up too cheers |
| 16:15.09 | Russ | 9639, 9639, 9659, 9699, 9719, 9739, 9759, 9759, etc |
| 16:15.41 | Russ | so every 120ms, a timestamp gets "reused" but the packet has new audio data |
| 16:16.03 | fullstop | I know that this is an asterisk channel, but what to people here think of freeswitch? I won't be leaving asterisk, but I've wondered if there is anything that FS does better that Asterisk could learn from? |
| 16:16.34 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 16:16.50 | *** join/#asterisk davlefou (~david@41.225.223.140) |
| 16:18.11 | WIMPy | aliverius: qmisdnwatch seems rather unreliable. |
| 16:18.15 | fullstop | this place is dead without Katty. |
| 16:19.29 | WIMPy | We did have a bit of off-topic this morning :-) But right, where's she gone? |
| 16:19.51 | aliverius | Ext(port 0: hfc-pci.1) TE ptmp use:0 L2 unkn L1 unkn |
| 16:19.51 | aliverius | <PROTECTED> |
| 16:19.56 | fullstop | If I had to guess, I would say that she is off baking. |
| 16:19.58 | carrar | Y*A*W*N |
| 16:19.59 | Jcook_5xData | can someone take a look at this http://pastebin.com/Xajg1sz6 i think WIMPy was right I have a IRQ problem |
| 16:19.59 | aliverius | even when i pick the pots phone |
| 16:20.47 | WIMPy | aliverius: That's ok. You won't see mor unless you use the card. |
| 16:21.15 | fullstop | Is that from a laptop or do you have a server board with the intel integrated graphics? |
| 16:21.33 | WIMPy | Jcook_5xData: Shared IRQs can cause trouble, yes. |
| 16:22.12 | *** join/#asterisk vfabi (~fabi@host-static-109-185-193-114.moldtelecom.md) |
| 16:22.47 | *** join/#asterisk war9407 (war@c-71-62-61-45.hsd1.va.comcast.net) |
| 16:23.50 | WIMPy | Wow. That's a lot of IRQ sharing going on there. and none >15. Looks like there's something fundametally wrong there. |
| 16:23.53 | *** join/#asterisk djb3li3ny (~djbelieny@173.226.191.80) |
| 16:24.10 | djb3li3ny | Good day people from asterisk! |
| 16:24.11 | Jcook_5xData | WIMPy, what can I do? not like the old days of dip switches. and I have no other pci ports |
| 16:24.58 | aliverius | WIMPy: how could i use lcr to make the card do something? something without messing with asterisk yet. maybe make a phone ring |
| 16:25.33 | Jcook_5xData | I can turn off none use port like printer and serial, change the pnp setting to other of what ever it is |
| 16:26.33 | WIMPy | Jcook_5xData: For some reason your system doesn't support high IRQs. That's bad. Could be your BIOS setting or your Kernel. |
| 16:27.27 | Jcook_5xData | running straight asterisknow. I install 1.6 w asterisk gui |
| 16:27.28 | WIMPy | aliverius: Just edit interface.conf for a single card in TE-ptmp mode. |
| 16:28.04 | WIMPy | The 'lcr fork' and 'lcaradmin status'. If you call that line, you should see that then. |
| 16:28.11 | Jcook_5xData | WIMPy, I think you right that may bouncing problem. JOY late night here me |
| 16:29.33 | WIMPy | Jcook_5xDataL Sharing an IRQ with bot SATA and ethernet is surely asking for trouble. |
| 16:29.56 | aliverius | WIMPy: [Ext] |
| 16:29.56 | aliverius | extern |
| 16:29.56 | aliverius | portnum 0 is not enough? |
| 16:30.16 | aliverius | docs are not very thoroough |
| 16:30.59 | WIMPy | Yes, that will do. |
| 16:31.14 | Jcook_5xData | WIMPy, thanks :) |
| 16:33.54 | *** join/#asterisk sgimeno (~chatzilla@163.117.206.10) |
| 16:34.14 | buzzyd | Can any one supply a reload script in php that works in Asterisk 1.8 as I need one and will happily supply beer money... |
| 16:34.52 | Kobaz | if you have autoload=yes |
| 16:35.05 | Kobaz | but you have specific load=> entries |
| 16:35.19 | Kobaz | does that force the load order of those entries, or is it still undefined |
| 16:35.43 | Kobaz | buzzyd: reload script as in.... ? |
| 16:37.07 | aliverius | WIMPy: interesting stuff: lcradmin testcall 0 210652xxxxx 210654xxxxx |
| 16:37.13 | aliverius | then |
| 16:37.22 | aliverius | # lcradmin state |
| 16:37.33 | aliverius | 19.05.11 18:45:36.396 EP(2): INTERFACE (too busy) interface 0 |
| 16:37.33 | aliverius | 19.05.11 18:45:36.396 EP(2): INTERFACE (no free ports found) |
| 16:37.47 | buzzyd | Kobaz: As in this which worked for 1.4 but doesn't in 1.8 http://pastebin.com/download.php?i=unskVB6C |
| 16:38.28 | WIMPy | aliverius: You see the card on top of the stat display? |
| 16:38.44 | aliverius | Ext(port 0: hfc-pci.1) TE ptmp use:0 L2 unkn L1 unkn |
| 16:38.59 | Kobaz | what's the non-download link? |
| 16:38.59 | WIMPy | Ok. |
| 16:39.16 | buzzyd | http://pastebin.com/unskVB6C |
| 16:39.28 | WIMPy | I don't think I ever used 'testcall'. Lets see what that does. |
| 16:39.44 | Kobaz | i wouldn't see why that wouldn't work |
| 16:39.52 | Kobaz | buzzyd: any errors? |
| 16:40.08 | buzzyd | utils.c:1180 ast_careful_fwrite: fwrite() returned error: Broken pipe |
| 16:40.12 | carrar | http://devour.com/video/flood-moat/ |
| 16:40.15 | carrar | insaine |
| 16:40.19 | WIMPy | aliverius: you need to use the name if the interface, i.e. "ext". |
| 16:40.23 | WIMPy | of |
| 16:40.32 | Kobaz | buzzyd: did you try and telnet in and paste in what the script is sending |
| 16:40.42 | Kobaz | buzzyd: telnet localhost 5038 |
| 16:41.42 | buzzyd | yes but didn't work different error just doing it now. |
| 16:41.56 | aliverius | RELEASE |
| 16:41.56 | aliverius | <PROTECTED> |
| 16:42.14 | aliverius | ich spreche deutsh nicht sehr gut |
| 16:42.16 | WIMPy | aliverius: That doesn't look bad. |
| 16:42.30 | aliverius | but i didnt get a ring |
| 16:42.56 | *** join/#asterisk eharris (~eharris@99-179-7-82.lightspeed.austtx.sbcglobal.net) |
| 16:43.04 | WIMPy | Could you see L1 up and L2 up? |
| 16:43.18 | WIMPy | "no user responding" |
| 16:43.50 | WIMPy | Ah, ok, you get that if it can't activate L1. |
| 16:43.55 | WIMPy | Bad. |
| 16:44.33 | WIMPy | Looks like you got some issue on the physical side. |
| 16:46.11 | buzzyd | Actually yes it does work via telnet must have typo'd last time. |
| 16:47.21 | Khratos | buzzyd: http://66.128.60.148/asterisk-doc/ |
| 16:48.02 | *** join/#asterisk war9407 (war@c-71-62-61-45.hsd1.va.comcast.net) |
| 16:48.48 | Khratos | I put that on that server. That comes inside Asterisk src directory, inside doc/ |
| 16:50.21 | WIMPy | aliverius: I see the NetMod has some switches to configure you S0 bus. Have you checked those? |
| 16:50.38 | Khratos | And, of course, the official documentation: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Manager+Interface+%28AMI%29 |
| 16:51.39 | aliverius | WIMPy: thank you for having done such a thorough research for me... yes i ve seen those and i will play with them a bit . i think they are set correctly (default) |
| 16:53.04 | WIMPy | aliverius: You want the short but, I guess. However I would expect a wrong setting to be unreliable, not to not work at all, but you never know. |
| 16:53.33 | WIMPy | If you can terminate the card, you should do that as well. |
| 16:53.35 | aliverius | the short you mean not the extended |
| 16:53.45 | WIMPy | yes |
| 16:53.58 | aliverius | should i terminate the card or not? |
| 16:54.04 | aliverius | lets paly |
| 16:54.07 | aliverius | play |
| 16:54.40 | WIMPy | It's a bus and both ends should be terminates. |
| 16:54.45 | WIMPy | d |
| 16:55.12 | Kobaz | static pthread_t shaun_of_the_dead_thread = AST_PTHREADT_NULL; |
| 16:55.13 | Kobaz | haha |
| 16:55.23 | dan__t | Alright, so I understand the difference between static and dynamic realtimes. |
| 17:04.00 | QuantumSchema | Good afternoon all! |
| 17:04.36 | dan__t | hi. |
| 17:05.01 | buzzyd | Khratos: Thank you I have totally different docs in my asterisk folder som I am just grabbing them from that server. |
| 17:08.59 | *** join/#asterisk timahvo1 (~rogue@41.223.57.74) |
| 17:12.43 | *** join/#asterisk weta (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 17:13.57 | dan__t | So, MixMonitor wasn't working because I enclosed <command> in double quotes. |
| 17:14.14 | dan__t | Dumb idea, I guess, going that route because I gave a variable as an argument. |
| 17:14.26 | dan__t | But, if that's how it wants to work, who am I to argue |
| 17:15.27 | carrar | You are dan_t |
| 17:15.32 | carrar | __ |
| 17:19.08 | dan__t | What? |
| 17:19.20 | carrar | What what |
| 17:19.36 | dan__t | whaaaat |
| 17:22.35 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 17:27.50 | Khratos | buzzyd: they are available in source distributions (1.6, and 1.8) |
| 17:28.04 | Khratos | In fact, 1.8 documentation is huge |
| 17:28.11 | Kobaz | hmm |
| 17:28.15 | Kobaz | 1.8.4 is crashing on startup |
| 17:28.20 | Kobaz | loading cel_odbc |
| 17:35.34 | *** join/#asterisk davlefou (~david@196.203.146.117) |
| 17:40.39 | *** join/#asterisk _pepo_ (c9ea54aa@gateway/web/freenode/ip.201.234.84.170) |
| 17:40.48 | _pepo_ | hi friends |
| 17:42.29 | Kobaz | fixed it |
| 17:48.28 | *** join/#asterisk _pepo_ (c9ea54aa@gateway/web/freenode/ip.201.234.84.170) |
| 17:48.55 | _pepo_ | I am looking for some softphone for SIP that I can change the interface (maybe with develop) with the logos of my job, What project I can use? |
| 17:50.00 | leifmadsen | _pepo_: for a price, you can use zoiper |
| 17:50.13 | leifmadsen | otherwise, look at ekiga or something open source |
| 17:50.22 | *** join/#asterisk GreatSUN (~greatsun@178-190-209-52.adsl.highway.telekom.at) |
| 17:50.24 | *** part/#asterisk fireman_biff (~biff@65.48.133.102) |
| 17:50.28 | GreatSUN | re |
| 17:51.10 | leifmadsen | fd |
| 17:51.13 | leifmadsen | vc |
| 17:55.42 | carrar | _pepo_, I think counter path offers that too |
| 17:56.02 | *** join/#asterisk davlefou (~david@41.225.94.240) |
| 17:56.50 | carrar | yeah they do |
| 17:56.56 | carrar | http://www.counterpath.com/softphone-skin-showcase.html |
| 17:58.11 | *** join/#asterisk chandoo (~chandoo@ool-4a5a2824.dyn.optonline.net) |
| 17:59.56 | sereal | Has anyone used the pyst python asterisk manager interface? |
| 18:04.19 | *** join/#asterisk timahvo1 (~rogue@41.223.57.74) |
| 18:04.59 | *** join/#asterisk chandoo (~chandoo@ool-4a5a2824.dyn.optonline.net) |
| 18:09.40 | *** join/#asterisk timahvo1 (~rogue@41.223.57.74) |
| 18:11.31 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 18:11.48 | bn-7bc | Hmm im stuck with a starnge nat problem, my asterisk box is behind a net, wen I try to cal from an ua inside the same nat to a sip server outside it vorks but when my ua is also outside the nat (3g phone) i get no sound but when i call vm on asterisk i van her everything as usual, , so fhat did i do wring (the router rythat does tha nat is a cisco 892w with ios 15.0)? |
| 18:12.41 | *** join/#asterisk chandoo (~chandoo@ool-4a5a2824.dyn.optonline.net) |
| 18:16.32 | buzzyd | Khrakos, |
| 18:17.01 | buzzyd | When ever I put a url into the browser it just shows page not valid |
| 18:19.31 | Kobaz | what's the way to get the highest quality audio from mp3s into asterisk |
| 18:19.49 | Kobaz | if i use madplay to play the mp3s i get weird audio artifacts |
| 18:19.54 | Kobaz | like a sssssh sound every so often |
| 18:20.15 | buzzyd | Kobaz, I always use SLN on mine |
| 18:20.44 | Kobaz | convert the mp3 to sln? |
| 18:20.51 | Kobaz | all i have is the mp3 |
| 18:22.09 | buzzyd | http://media.io/ might do the trick |
| 18:23.00 | Kobaz | well i can convert the mp3 to wav with vlc and then use asterisk to go from wav to sln |
| 18:23.01 | *** part/#asterisk Hail2theKingBaby (~Hail2theK@194-176-87-181.onyx.net) |
| 18:23.04 | Kobaz | but that site is cool |
| 18:24.10 | *** join/#asterisk chandoo (~chandoo@ool-4a5a2824.dyn.optonline.net) |
| 18:24.11 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 18:29.10 | *** join/#asterisk chandoo (~chandoo@ool-4a5a2824.dyn.optonline.net) |
| 18:31.05 | *** join/#asterisk chandoo (~chandoo@ool-4a5a2824.dyn.optonline.net) |
| 18:34.05 | *** join/#asterisk Ryushin (~Ryushin@2001:1938:2a5:0:21c:25ff:fe97:c868) |
| 18:34.37 | _Corey_ | I used that recently when a customer sent over a bunch of WMA prompts... good site |
| 18:39.32 | *** join/#asterisk zorp75ck (~zorp75ck@146.186.115.103) |
| 18:43.31 | *** join/#asterisk sweeper (~sweeper@softcheese.net) |
| 18:44.29 | sweeper | I want mixmonitor to store the output files with a timestamp as the filename, but MixMonitor(${DATETIME}.wav) isn't working....any ideas? |
| 18:45.00 | sweeper | I could make a little shell script to be run after mixmonitor executes, but that doesn't help with multiple ongoing calls |
| 18:46.16 | _Corey_ | sweeper: I use this inside filenames --- ${STRFTIME(${EPOCH},EDT,%F-%H%M)} |
| 18:48.38 | *** join/#asterisk chandoo (~chandoo@ool-4a5a2824.dyn.optonline.net) |
| 18:49.26 | *** join/#asterisk g0rdorin (~g0rdorin@46-116-24-73.bb.netvision.net.il) |
| 18:50.23 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
| 18:50.54 | sweeper | that worked, awesome |
| 18:52.18 | *** join/#asterisk momelod (~smelo@66.46.12.98) |
| 18:52.30 | momelod | Greetings channel |
| 18:53.40 | momelod | Im using asterisk w/ Cisco handsets. When I dial an extension, it it possible to display the Name of the person im dialing on the phone's screen? Currently if i dial say extension 100, only 100 is shown on my screen, not 100 - operator.. |
| 18:57.29 | leifmadsen | that should happen automatically if you're using something like Asterisk 1.8 |
| 18:57.34 | leifmadsen | assuming the device can support it |
| 18:59.21 | WIMPy | o.O |
| 19:00.21 | WIMPy | digs out a 2nd sip phone. |
| 19:01.21 | leifmadsen | momelod: at least I know if you do a transfer or something, the caller display will update with the person who called you, and then who was transferred to you |
| 19:01.34 | GreatSUN | hey guys |
| 19:02.14 | GreatSUN | I am still having problems sending faxes out though hylafax, iaxmodem (IAX2) -> dahdi |
| 19:02.42 | GreatSUN | if I change the codec for iaxmodem(s), I get difference between no dialtone and busy |
| 19:02.57 | GreatSUN | I also have been running a pri debug |
| 19:03.01 | GreatSUN | see here: http://nopaste.info/f94ef23fcc.html |
| 19:03.36 | GreatSUN | but afaik this can't be a dahdi/pri problem (at least not alone) |
| 19:04.01 | GreatSUN | cause when calling the same number from a sip-phone (SIP->DAHDI) it works |
| 19:04.16 | GreatSUN | receiving faxes also works |
| 19:04.31 | WIMPy | "PRI_EVENT_CONFIG_ERR"? What's that? |
| 19:04.41 | GreatSUN | does anyone have any idea what could be the problem |
| 19:04.57 | *** join/#asterisk chandoo (~chandoo@ool-4a5a2824.dyn.optonline.net) |
| 19:05.07 | GreatSUN | WIMPy: normally I'd say it's a configuration error in PRI configuration |
| 19:05.16 | WIMPy | Line 226: BC Speech. |
| 19:05.21 | WIMPy | The usual problem. |
| 19:05.27 | GreatSUN | but since this doesn't come up when connecting from SIP->DAHDI |
| 19:05.47 | GreatSUN | I could bet it should be an IAX problem, not dahdi |
| 19:05.56 | GreatSUN | but you are for sure the expert |
| 19:05.59 | GreatSUN | ;o) |
| 19:06.16 | WIMPy | Who's an expert? |
| 19:06.21 | GreatSUN | you# |
| 19:06.38 | WIMPy | YOu might be able to change it from iaxmodem or from the dialplan. |
| 19:07.53 | *** join/#asterisk mickecarlsson (~Micke@h246n1c1o1101.bredband.skanova.com) |
| 19:10.15 | *** join/#asterisk chandoo (~chandoo@ool-4a5a2824.dyn.optonline.net) |
| 19:10.40 | *** join/#asterisk timahvo1 (~rogue@41.223.57.73) |
| 19:11.37 | sereal | Should I upgrade a production system from 1.4 to 1.8? |
| 19:11.48 | sereal | Will all my configs work or do I need to make changes? |
| 19:12.43 | buzzyd | any PHP/AMI gurus around? |
| 19:12.56 | dan__t | sigh. |
| 19:13.33 | buzzyd | sereal, my test upgrade didn't so no |
| 19:13.38 | serafie | sereal: some of your configs will work, but we cannot say if all will. You need to replicate your system (with 1.4) on a development machine, upgrade, and test test test. |
| 19:14.03 | sereal | okay, so it's not as simple as moving the extensions.conf and sip.conf over. |
| 19:14.12 | sereal | and iax.conf over |
| 19:14.22 | *** join/#asterisk davlefou (~david@41.225.195.237) |
| 19:14.23 | sereal | What didn't work buzzyd? |
| 19:14.34 | serafie | sereal: it may be, but it depends on what you have configured. |
| 19:14.40 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 19:15.00 | sereal | I see. Is there any infomation on what specifically breaks? |
| 19:15.17 | sereal | The dialplan isn't doing anything that fancy. |
| 19:15.48 | GreatSUN | does someone have any idea on a windows client to administrate hylafax queue? |
| 19:16.12 | sereal | the only thing I use is Dial(), goto, and Background(), Playback(), and waitexten |
| 19:16.29 | sereal | regardless I would do this on a testmachine first. |
| 19:16.30 | buzzyd | some of my dialplans and AMI doesn't work the same with my reload script so far. still doing it only started yesterday :) |
| 19:16.46 | sereal | I don't use the AMI for anything |
| 19:17.19 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 19:17.29 | sereal | It might be worth while upgrading to 1.8 while my dialplan is fairly small. |
| 19:17.43 | sereal | Are there any signifant benefits of 1.8 v 1.4? |
| 19:19.36 | serafie | yes, bugfixes! |
| 19:20.25 | sereal | Just a pain in the ass to do a big upgrade, the machine just got into production |
| 19:20.50 | leifmadsen | sereal: then I would highly suggest migrating to a separate box and not just installing 1.8 on your 1.4 machine |
| 19:21.05 | leifmadsen | it's like saying, "I have RedHat 9 and want to use CentOS 5.5" |
| 19:21.10 | leifmadsen | totally different beast :) |
| 19:21.23 | sereal | regardless I wouldn't just upgrade my production machine. |
| 19:21.38 | sereal | it's more like saying should you go from RH7 to RH9 |
| 19:21.40 | leifmadsen | sereal: you'd be surprised how many people would, which is why I say what I did.... |
| 19:22.51 | sereal | yeah it's worth saying for sure. |
| 19:23.38 | WIMPy | is stuck. I'm sure I've ssen something to access the Bearer Capability from the dilplan, but I can't find anything. |
| 19:24.03 | leifmadsen | It's more like trying to upgrade RHEL 4.4 to RHEL 6.1 :) |
| 19:24.26 | leifmadsen | at least based on release dates of Asterisk 1.4.x/1.8.x and RHEL 4.x/6.x :) |
| 19:24.37 | sereal | true enough. |
| 19:24.52 | leifmadsen | that was a fun, yet terribly useful distraction for a couple of mins |
| 19:25.00 | leifmadsen | s/useful/useless/ |
| 19:25.27 | sereal | thats neat. |
| 19:25.34 | leifmadsen | heh |
| 19:27.52 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 19:28.27 | *** join/#asterisk NightMonkey (debian-tor@pdpc/supporter/professional/nightmonkey) |
| 19:31.48 | *** join/#asterisk asilva (~andre@2801:88:1000:2::12) |
| 19:32.05 | asilva | Hello, What could cause Asterisk processe to freeze? |
| 19:32.18 | *** join/#asterisk MarKsaitis (~MarKsaiti@client-86-31-252-98.oxfd.adsl.virginmedia.com) |
| 19:32.36 | De_Mon | i'm trying to setup a sip peer that needs the asterisk server to register "from" a specific ip |
| 19:32.46 | leifmadsen | asilva: What would cause a whale to drown? |
| 19:32.54 | asilva | sometimes mine just stop working, it runs under debian 6 2.6.32 kernel. asterisk 1.6.2.17.2 |
| 19:33.01 | leifmadsen | ~collectdebug |
| 19:33.02 | infobot | i heard collectdebug is a method of collecting logs allowing others help troubleshoot an issue. Refer to https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 19:33.04 | De_Mon | looking at my sip peer account i can't tell what parameter i'm supposed to set the from ip in |
| 19:33.15 | leifmadsen | ~coreshowlocks |
| 19:33.30 | leifmadsen | ~asteriskdeadlocks |
| 19:33.33 | leifmadsen | ~debugdeadlock |
| 19:33.35 | leifmadsen | grrrrrr |
| 19:33.39 | leifmadsen | pabelanger: help me out bro! |
| 19:33.45 | asilva | got it! take it easy ;) |
| 19:33.56 | *** join/#asterisk joshaidan (~brianj@S0106001c1023e838.tb.shawcable.net) |
| 19:34.02 | leifmadsen | I'm trying to figure out the rbot link to the 'backtrace / deadlock' stuff |
| 19:34.07 | mickecarlsson | wonders if leifmadsen has looked at the update on mickecarlssons issue |
| 19:34.07 | asilva | I'll try to collect debug info to see if there is enough information to report a bug! |
| 19:34.15 | pabelanger | ~backtrace |
| 19:34.15 | infobot | backtrace is a debugging tool that is useful when trying to track down asterisk crashes (use -g with asterisk to generate a core, then read doc/backtrace.txt). See also https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace |
| 19:34.22 | leifmadsen | yay :) |
| 19:34.31 | leifmadsen | mickecarlsson: I don't know what issue you speak of |
| 19:34.43 | asilva | But in my case, the process don't crash, it just hangs but nothing really happens! |
| 19:34.49 | mickecarlsson | leifmadsen: https://issues.asterisk.org/view.php?id=18681 |
| 19:34.55 | *** join/#asterisk davlefou (~david@41.227.63.143) |
| 19:35.01 | leifmadsen | asilva: which would be a deadlock as documented on that page |
| 19:35.37 | asilva | good to know, deadlock.. I'll look into to understand more about it, and get back here with more info! Thanks for the help |
| 19:35.49 | itsbroken | Hello, I'm seeing some messages like: WARNING[18252]: translate.c:155 framein: no samples for ulawtolin Googling doesn't provide much info... mostly its russian stuff... anyone have any ideas what this is about? |
| 19:36.15 | itsbroken | oops |
| 19:36.47 | itsbroken | translate.c:155 framein: no samples for ulawtolin |
| 19:38.40 | leifmadsen | mickecarlsson: thanks -- issues marked as related |
| 19:38.59 | mickecarlsson | Thank you. I am a pro im compiling Asterisk now :-) |
| 19:39.06 | leifmadsen | :) |
| 19:39.23 | mickecarlsson | btw, the .patch between releases, how can I use them? |
| 19:39.45 | mickecarlsson | Is that a patch to roll up to next release? |
| 19:40.02 | leifmadsen | mickecarlsson: patch between releases? |
| 19:40.26 | mickecarlsson | Yes, in http://downloads.asterisk.org/pub/telephony/asterisk/ |
| 19:40.41 | mickecarlsson | asterisk-1.8.4-patch.gz for eample |
| 19:40.42 | leifmadsen | yes it would be a patch to get you to the next release |
| 19:40.49 | leifmadsen | which would apply to 1.8.3.3 I believe |
| 19:41.02 | mickecarlsson | OK, that could have saved me some time, go figure |
| 19:41.09 | leifmadsen | I can't remember exactly how ti builds the patches and what it applies to :) |
| 19:41.24 | mickecarlsson | Well, I have to read up on that (later) |
| 19:41.26 | leifmadsen | (it's all automated in a script) |
| 19:42.21 | mickecarlsson | OK, I will crawl back to freepbx-dev and continue over there, but will stay connected here. Thanks again Leif |
| 19:45.06 | leifmadsen | enjoy! |
| 19:50.35 | itsbroken | leifmadsen: any idea of what this translate framein no samples for ulawtolin is about? =) |
| 19:51.04 | itsbroken | it doesn't appear to be effecting calls but I assume it may be impacting call quality? |
| 19:51.18 | itsbroken | (effecting the success of calls) |
| 19:55.17 | leifmadsen | itsbroken: no idea |
| 19:55.20 | *** join/#asterisk timahvo1 (~rogue@41.223.57.76) |
| 19:56.02 | WIMPy | Does SetTransferCapability() not exist any more? |
| 19:57.33 | leifmadsen | sounds like something that would have been moved to a dialplan function, but I've never heard of that application |
| 19:58.07 | *** join/#asterisk war9407 (war@c-71-62-61-45.hsd1.va.comcast.net) |
| 19:59.24 | De_Mon | it looks like my peer acount is going out the wrong ip address, how do I bind a particular peer to use a specific outbound address? |
| 19:59.33 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 20:02.15 | *** join/#asterisk dehuman (~dehuman@adsl-99-29-196-191.dsl.sfldmi.sbcglobal.net) |
| 20:05.13 | *** join/#asterisk cerberus_za (~coert@196-210-190-65.dynamic.isadsl.co.za) |
| 20:05.29 | Khratos | leifmadsen: had you noticed that the personal spaces's entries in the wiki are not available for general reading unless people register to it? |
| 20:06.40 | WIMPy | cant find any replacement. |
| 20:07.22 | *** join/#asterisk timahvo1 (~rogue@41.223.57.77) |
| 20:07.55 | *** join/#asterisk timahvo1 (~rogue@41.223.57.73) |
| 20:09.55 | leifmadsen | Khratos: not really -- I haven't used the wiki a ton and am not entirely sure who it is all setup yet |
| 20:12.11 | sereal | How would I ask the AMI if a certain sip peer is on the phone? |
| 20:14.00 | leifmadsen | manager show command Status? |
| 20:14.03 | leifmadsen | not sure what that returns |
| 20:15.15 | leifmadsen | goes off to make coffee finally and jam a bit |
| 20:15.36 | *** join/#asterisk acidjazz (acidjazz@notchill.com) |
| 20:17.36 | *** join/#asterisk timahvo1 (~rogue@41.223.57.75) |
| 20:27.11 | acidjazz | hey guys .. so i'm writing this mobile app that needs to send texts and voicemails .. but obviousely i cant do it w/in the app. i dont really have the time to setup asterisk boxes .. can anyone recommend a good service provider for this w/ good pricing/api ? |
| 20:33.09 | *** join/#asterisk WiretapWork (~Wiretap@unaffiliated/wiretap) |
| 20:35.40 | paulc | acidjazz: Twilio might be worth a look, or Tropo |
| 20:37.43 | De_Mon | acidjazz we've used telesign, and it's pretty good. |
| 20:41.51 | leifmadsen | paulc: +1 |
| 20:42.09 | paulc | doffs hat to leifmadsen |
| 20:42.12 | paulc | how's yer Thursday? |
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| 20:42.19 | leifmadsen | acidjazz: http://smsified.com/ |
| 20:42.30 | leifmadsen | paulc: not bad -- just got back in from cutting the grass for the first time this year |
| 20:42.36 | leifmadsen | it was HUGE because it's been raining for 5 days straight |
| 20:42.40 | leifmadsen | (since Saturday) |
| 20:43.00 | paulc | leifmadsen: NICE. Love that smell, and love being outside. Great sunny day in Vancouver (for a change!) www.katkam.ca and I'm stuck in the office, procrastinating.. |
| 20:43.20 | leifmadsen | paulc: nice! ya it's sunny here now too and going to be amazing for beach volleyball tonight |
| 20:43.39 | paulc | haha awesome.. (is it home time yet? I'm so ready..) |
| 20:43.46 | leifmadsen | it is -- 4:45pm here :) |
| 20:43.49 | leifmadsen | and I work from home, so... yay! |
| 20:44.01 | leifmadsen | I started at 8:00am or so, soooooo ya |
| 20:44.27 | leifmadsen | I'm thinking of going to the basement to cool off and jam on some ableton live |
| 20:45.06 | leifmadsen | hey all! if you haven't bought either Asterisk: The Definitive Guide or the Asterisk Cookbook, please do so! I have a fence and deck to install in a couple of weeks..... :D |
| 20:45.52 | leifmadsen | ...that was supposed to be a joke people... |
| 20:47.10 | paulc | watches the tumbleweed blow through the room |
| 20:47.14 | paulc | tough crowd! |
| 20:47.19 | *** join/#asterisk luckyaba (~Lucky@ip72-194-218-169.sb.sd.cox.net) |
| 20:47.58 | paulc | I should have worked from home today. Traffic in was brutal - 1:50... it'll be 25 mins going home.. and it feels like it might be a BBQ kinda night I think.. then beers.. and the day off tomorrow.. for some Asterisk playing :) |
| 20:48.20 | leifmadsen | paulc: yay! I'm having ribs tonight :) |
| 20:48.29 | leifmadsen | I might do them on the BBQ. Beer is in the fridge. |
| 20:48.40 | leifmadsen | I'll get the fiancee to drive to volleyball, or the neighbors |
| 20:48.48 | paulc | leifmadsen: Did you get your psychic app finished? I was going to show you screen shots from our one-on-one similar type app front end thing |
| 20:50.21 | leifmadsen | paulc: I ended up mostly backing away from that client because they didn't know what they wanted (after building nearly 3 different things) |
| 20:50.25 | leifmadsen | so I'm not sure how it ended |
| 20:50.26 | *** join/#asterisk Guizmo (~Guizmo@adsl-84-226-190-80.adslplus.ch) |
| 20:50.30 | _Corey_ | hmmm, beer and BBQ sounds pretty good right about now |
| 20:50.34 | acidjazz | leifmadsen: awesome man thanks |
| 20:50.40 | acidjazz | leifmadsen: these guys do voicemail? |
| 20:50.47 | leifmadsen | acidjazz: never used them, so not sure |
| 20:50.51 | leifmadsen | probably not... |
| 20:51.02 | leifmadsen | I bet you could integrate with tropo or something for that |
| 20:51.04 | paulc | leifmadsen: Ah that's too bad.. (I know how that goes too) Ours ended up ok for phase 1 and is running nicely, with all those "phase 2 requirements" on ice for now |
| 20:51.29 | leifmadsen | paulc: ya it happens, but luckily not too often. This was only the 2nd client in 7 years I've tried to back away from. |
| 20:51.35 | leifmadsen | they needed someone full time, and I didn't have the time |
| 20:51.44 | leifmadsen | I was trying to be the hero to help them out mostly, and it didn't work out |
| 20:51.51 | acidjazz | twilio looks like its got what i need |
| 20:52.10 | leifmadsen | anyways, I'm off to the nice cool basement with some coffee and will jam out some beams :) |
| 20:52.12 | leifmadsen | beats too |
| 20:53.07 | paulc | leifmadsen: Enjoy! |
| 20:53.11 | Guizmo | hello, I have an asterisk server, but I want to connect it to 3 ISDN phone line. I know I need to get a pci card to do this (from digium or other), but if I want to remove the NT box is it possible ? or do I need to keep the NT box ? |
| 20:53.15 | paulc | acidjazz: Twilio is cool - I like it a lot :-) |
| 20:53.41 | acidjazz | awesome thanks guys im set |
| 20:53.50 | acidjazz | im gonna look so rad pretending i did all this research myself |
| 20:53.56 | paulc | Guizmo: I think usually you keep the NT for termination, loopback, remote testing etc - it's the telco demarc point |
| 20:54.05 | paulc | acidjazz: such an honourable approach :-p |
| 20:56.50 | WiretapWork | Guizmo, you wantto keep the NTE |
| 20:56.54 | Guizmo | paulc: ah ok thanks it is not a stable NT box, it is why I was hoping to bypass it :) but if it is just for loopback, testing, etc it should be ok |
| 21:04.56 | *** join/#asterisk manji (~manjiki@2a02:580:8000:8601:226:bbff:fe13:1c09) |
| 21:08.05 | *** join/#asterisk WiretapWork (~Wiretap@unaffiliated/wiretap) |
| 21:17.17 | *** join/#asterisk ipc9 (~rob@173-162-245-206-NewEngland.hfc.comcastbusiness.net) |
| 21:18.30 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 21:20.36 | ipc9 | Has anyone in here done any soft phone development? |
| 21:30.39 | *** join/#asterisk zachsis (gatsby@yogsothoth.net) |
| 21:30.48 | WiretapWork | anyone have a recommendation for a softphone for iphones? |
| 21:31.51 | malcolmd | ipc9: the ag-projects people or the pjsip people are probably good people to talk to about soft phones. ag-projects for a complete softphone (using their sipsimplesdk) or pjsip folks (they write a sip ua called pjsip). |
| 21:32.17 | malcolmd | WiretapWork: positive reviews for the counterpath bria for iphone |
| 21:32.34 | _Corey_ | +1 on bria for the iphone |
| 21:32.47 | _Corey_ | very professional |
| 21:32.49 | zachsis | x-lite supposedly has a version for iphone |
| 21:32.55 | zachsis | never used it tho' |
| 21:34.34 | malcolmd | i think x-lite is only for windows and mac. bria's available for iphone, in addition to other platforms |
| 21:34.54 | *** join/#asterisk mykhyggz (~col@evolone.org) |
| 21:35.19 | ipc9 | thanks |
| 21:36.06 | zachsis | anyone ever had trouble with asterisk not noticing hangups on an FXO card ? |
| 21:36.38 | *** join/#asterisk rue_work (~rue_work@24.207.122.10) |
| 21:37.04 | rue_work | can I create a black hole mailbox by turning on delete=yes and not entering an email or having attach=yes? |
| 21:37.58 | paulc | rue_work: I'd half hope it would require an email to be sent before deleting.. but nothing stopping you using .forward or similar to accept the emails then throw them into a black hole |
| 21:38.32 | rue_work | hmmm |
| 21:38.40 | malcolmd | zachsis: a digium card? |
| 21:38.52 | zachsis | no, Sangoma A200 |
| 21:39.31 | zachsis | its really intermittent. usualyl happens when asterisk voicemail answers, then the caller hangs up after leaving a message and no hangup is detected |
| 21:40.49 | zachsis | i recently added 'hanguponpolarityswitch=yes' to my chan_dahdi.conf.. and so far it seems good, but we'll know after this weekend |
| 21:41.01 | zachsis | i'm wondering if anyone knows of any other workarounds |
| 21:44.58 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 21:49.09 | *** join/#asterisk JunK-Y (~junky@pdpc/supporter/active/junk-y) |
| 21:49.37 | WIMPy | mickecarlsson: Are you around? |
| 21:54.47 | rue_work | exten => s,n,SET(BLACKLIST=${ODBC_CHECKBLACK(${CALLERID(number)})}) |
| 21:54.48 | rue_work | exten => s,n,Gotoif($[${BLACKLIST}='1']?blackholevm,s,1) |
| 21:54.59 | rue_work | context of a variable, is it per call? |
| 21:55.20 | WIMPy | yes |
| 21:55.24 | rue_work | cool |
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| 22:00.03 | *** join/#asterisk diatonic (~diatonic_@mail.clearwater-research.com) |
| 22:03.11 | *** join/#asterisk drudge` (~drudge@unaffiliated/drudge/x-837452) |
| 22:03.29 | rue_work | did I write that right for a text value of 1? |
| 22:03.39 | rue_work | '1' "1" or just 1? |
| 22:06.32 | WIMPy | No quotes. |
| 22:06.36 | rue_work | ok |
| 22:17.28 | *** join/#asterisk Marvelous (~CE0@196.205.101.164) |
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| 22:21.09 | Jcook_5xData | Are there any good test I can run to see if there any problem with asterisk? |
| 22:22.51 | pabelanger | Jcook_5xData: bamboo.asterisk.org |
| 22:23.14 | pabelanger | or http://svn.asterisk.org/svn/testsuite/asterisk/trunk/ |
| 22:23.45 | pabelanger | Our you can write a new test if once does not exist and we can merge it into the code |
| 22:25.22 | *** join/#asterisk chandoo (~chandoo@ool-4a5a2824.dyn.optonline.net) |
| 22:26.05 | *** join/#asterisk ujjain (~ujjain@unaffiliated/ujjain) |
| 22:26.06 | ujjain | If I give my parents a PAP adapter that has a FXO and an FTO port, can they still use both VOIP and their landline for incoming/outgoing calls? |
| 22:26.57 | pabelanger | ujjain: sure, why not |
| 22:27.15 | WiretapWork | ujjain, if you set it up that way they sure can |
| 22:28.26 | Jcook_5xData | pabelanger, thanks I will check it out |
| 22:28.56 | ujjain | ah ok :) thanks!! |
| 22:31.13 | *** join/#asterisk mykhyggz (~col@evolone.org) |
| 22:39.38 | *** join/#asterisk timahvo1 (~rogue@41.223.57.76) |
| 22:44.35 | *** join/#asterisk mykhyggz (~col@evolone.org) |
| 22:51.23 | *** join/#asterisk cyphorious (~cyphoriou@chello062178189196.2.15.tuwien.teleweb.at) |
| 22:52.32 | WiretapWork | hmm... bria is nonfree |
| 22:52.35 | *** join/#asterisk cyphorious (~cyphoriou@chello062178189196.2.15.tuwien.teleweb.at) |
| 22:53.13 | *** join/#asterisk fhmiv (~fhmiv@adsl-065-005-242-247.sip.ags.bellsouth.net) |
| 22:53.25 | paulc | how non free? |
| 22:53.35 | paulc | like 4.99? or like XXXXXX pricier? |
| 22:55.41 | WiretapWork | paulc, like $10.99 |
| 22:56.23 | paulc | Yeah, that's steep. I have to admit I don't really use VoIP on my iPhone.. but I've got Fring installed.. and Truphone.. maybe one other one as well.. just don't actually "do" that. |
| 22:56.34 | paulc | Bria comes highly recommended.. but at that price point? yeah, that stings.. |
| 23:03.14 | ChannelZ | Do you need 500 copies of it or something? |
| 23:05.26 | *** join/#asterisk mykhyggz (~col@evolone.org) |
| 23:13.48 | carrar | yes |
| 23:14.17 | carrar | bria hoarding |
| 23:20.04 | rue_work | heh, better than soemthing hordign your D channels |
| 23:21.26 | *** part/#asterisk JunK-Y (~junky@pdpc/supporter/active/junk-y) |