| 00:04.54 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 00:07.53 | dieno | anyone fimilia with web Click to Call |
| 00:18.58 | *** join/#asterisk russellb (n=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 00:18.58 | *** mode/#asterisk [+o russellb] by ChanServ |
| 00:19.42 | NovceGuru | anybody successful getting something besides the "offically" supported ata's to work with voicepulse's "non open" accounts? |
| 00:19.59 | NovceGuru | I assume I'd just have to tftp or whatever the pap2 does to get its config from a pc, then read that config into my phone |
| 00:26.11 | *** join/#asterisk hsv-al (n=sdfsdf@user-24-214-126-81.knology.net) |
| 00:26.29 | NovceGuru | or if someone wants to share a pap2/ spa20xx mac/serial number so I can signup with voicepulse :) |
| 00:26.56 | russellb | 42 |
| 00:28.14 | Juggie | 13 |
| 00:28.21 | NovceGuru | 4b |
| 00:28.24 | Juggie | 8f |
| 00:28.33 | hsv-al | so,,,, |
| 00:28.36 | hsv-al | if voicepulse is so "2004" |
| 00:28.43 | hsv-al | whats a good iax2 termination service provider |
| 00:28.45 | hsv-al | and sip, etc. |
| 00:30.19 | *** join/#asterisk zuez (n=sf@catalyst.httpd.org) |
| 00:30.54 | NovceGuru | I'm using them for a hosted voip solution...if I can get signed up |
| 00:31.25 | NovceGuru | Can I use a device other than the Sipura SPA? |
| 00:31.26 | NovceGuru | The Sipura SPA-2000 is the most compatible with all of our advanced features. The Sipura SPA-2000 and SPA-3000 are currently the only devices you can use as your primary line. However, you may use any SIP device as an additional line on your account using the Open Access plan. |
| 00:32.20 | NovceGuru | and apparently the pap2 |
| 00:32.46 | NovceGuru | https://orders.voicepulse.com/SelectDevicePage.aspx :( |
| 00:32.58 | NovceGuru | fail |
| 00:33.30 | NovceGuru | https://orders.voicepulse.com/SelectPlanPage.aspx |
| 00:33.48 | iratik | i use voicepulse... i hate them |
| 00:34.02 | hsv-al | NICEEEEEEEE |
| 00:34.03 | hsv-al | ON CNN |
| 00:34.06 | hsv-al | FAREED ZAKARIA: GPS |
| 00:34.11 | hsv-al | SHOW OF THE DECADE |
| 00:34.23 | hsv-al | ack, wrong chan hahaha |
| 00:34.26 | NovceGuru | always find somebody who hates "*" service |
| 00:38.08 | *** join/#asterisk exothermc (n=miles@74.85.89.146) |
| 00:38.50 | exothermc | who makes a good call center package, for inbound and outbound callcenter handling calls for multiple companies? |
| 00:38.51 | NovceGuru | ah hell they say they can't port my number |
| 00:39.53 | NovceGuru | teliax can... |
| 00:39.53 | NovceGuru | hm |
| 00:42.46 | NovceGuru | I hate this vitelity vs teliax vs vocalocity vs * vs * vs * blah blah |
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| 00:55.20 | ZX81 | can anyone recommend an outlook/ie/thunderbird/firefox click to dial/popping app (not snapanumber)? |
| 00:57.06 | NovceGuru | http://www.voip-info.org/wiki-Asterisk+TAPI |
| 00:57.07 | NovceGuru | free? |
| 01:00.38 | ZX81 | yeah I guess, read that page |
| 01:01.59 | ZX81 | star outlook dialer looked ok, but they've gone :) |
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| 01:35.38 | exothermc | Is there a way to use TAPI like above but to send the call to an agent instead of a device? |
| 01:43.36 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
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| 01:53.10 | exothermc | Ok I guess the question I needed to ask is can you send a call to an agent, instead of device. |
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| 02:15.04 | zuez | Heh, is there a provider that's pretty good with letting me commit to a bulk amount of minutes so I pay really cheap long distance rates? |
| 02:22.20 | *** join/#asterisk shtoom (n=shtoom@121.246.167.147) |
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| 02:24.15 | *** mode/#asterisk [+o russellb] by ChanServ |
| 02:25.50 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
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| 02:34.17 | x86 | i just came up with the best slogan evar for a tampon company.... |
| 02:34.23 | x86 | "we're not #1, but we're up there" |
| 02:35.13 | [TK]D-Fender | ... |
| 02:35.36 | [TK]D-Fender | x86: Visa beat you to a better one... |
| 02:35.53 | [TK]D-Fender | x86: "Its everywhere you want to be" |
| 02:36.00 | x86 | hahaha |
| 02:36.17 | x86 | epic win |
| 02:37.03 | russellb | O.O |
| 02:37.08 | russellb | yay for #asterisk after-hours |
| 02:37.21 | x86 | *nod* |
| 02:37.26 | x86 | it's still SFW though :P |
| 02:37.54 | [TK]D-Fender | BRB... |
| 02:38.08 | russellb | alright, he's gone, everyone start talking about him |
| 02:38.28 | Entranced | tampons ? |
| 02:38.45 | russellb | no, D-Fender :-p |
| 02:40.08 | Entranced | white and fluffy |
| 02:40.30 | Entranced | Never seen him though |
| 02:40.38 | Entranced | just an assumption |
| 02:41.27 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 02:45.18 | *** join/#asterisk robba (n=rob@203.56.181.15) |
| 02:45.38 | robba | Hi guys |
| 02:46.05 | robba | does anyone know of a way of allowing only 30 sip devices to connect to a server? |
| 02:47.16 | [TK]D-Fender | robba: Why would you want to do that? Making a closed appliance? |
| 02:48.02 | robba | i know thats what i thought... why would anyone do that but a clients server is lock down like this for some unknown reason |
| 02:48.24 | [TK]D-Fender | robba: # of clients != security. Thats retarded |
| 02:48.49 | robba | i know this |
| 02:49.32 | robba | i'm trying to figure out whats stopping it from connecting |
| 02:49.45 | nick125 | [TK]D-Fender: Well, if you have limited resou...nevermind, yeah..it's retarded. |
| 02:49.58 | robba | could you point me in the direction of something that may be the cause |
| 02:50.13 | robba | i have googled for a while now and can't come up with anything |
| 02:50.40 | [TK]D-Fender | robba: Cause of what? You just asked if there was a way of blocking client connections. Whats the actual problem now? |
| 02:51.31 | robba | well the full story is we took over managing this server, and they want to add a few more sip devices, but when it hits 30 it won't allow any more |
| 02:51.41 | Entranced | licensing ? |
| 02:51.53 | Entranced | proprietary PBX? |
| 02:52.02 | robba | no sir |
| 02:52.32 | robba | hang on |
| 02:52.35 | robba | BARGH |
| 02:52.47 | robba | one of the guys just got back to me |
| 02:52.53 | robba | its running trixbox |
| 02:53.01 | [TK]D-Fender | robba: LOL... |
| 02:53.05 | robba | i'll go ask over there |
| 02:53.11 | [TK]D-Fender | robba: probably hacked into chan_sip or something. |
| 02:53.22 | robba | yeah |
| 02:53.35 | Entranced | if it's the ELITE version of Trixbox |
| 02:53.39 | nick125 | [TK]D-Fender: Don't you mean TrixSIP 3.0? *ducks and runs* |
| 02:53.59 | robba | sooo annoying |
| 02:54.22 | robba | just because trix runs a mashed asterisk backend they believe its asterisk |
| 02:56.39 | [TK]D-Fender | robba: But * is at the heart so clearly everything thats wrong is an Asterisk problem and we are obligated to help. |
| 02:56.46 | [TK]D-Fender | </sarcasm> |
| 02:57.26 | robba | lol |
| 02:58.19 | robba | hahahaha my boss has just informed me that they want to replace it with plain asterisk, they feel trixbox is causing too many headaches |
| 02:58.33 | NovceGuru | anybody use vitelity? |
| 02:58.40 | Entranced | yup |
| 02:59.04 | NovceGuru | hosted or sip trunking? |
| 02:59.08 | *** join/#asterisk ming_zym (n=ming_zym@222.241.202.220) |
| 02:59.10 | nick125 | robba: great idea |
| 02:59.15 | Entranced | IAX |
| 02:59.30 | NovceGuru | hmm |
| 02:59.35 | Entranced | trunking |
| 02:59.39 | NovceGuru | Looking to try them for their hosted service |
| 02:59.58 | NovceGuru | they have a similar pricing model to voicepulse connect |
| 03:00.31 | Entranced | no idea about the hosting part... but trunking works quite nicely for me |
| 03:01.04 | Entranced | good quality calls... international and local |
| 03:01.28 | NovceGuru | always a + |
| 03:01.46 | Entranced | ...and can't beat their pricing |
| 03:01.57 | Entranced | well.. at least for what I need |
| 03:02.02 | x86 | Entranced: what do you pay per minute? |
| 03:02.24 | Entranced | 1.4c |
| 03:02.29 | x86 | OUCH! |
| 03:02.38 | x86 | les.net, $0.008 per minute |
| 03:02.46 | x86 | (8 tenths of a cent) |
| 03:02.46 | Entranced | I don't use it too much for outgoing |
| 03:03.05 | Entranced | free incoming |
| 03:03.23 | Entranced | $0.014 |
| 03:03.28 | [TK]D-Fender | Entranced: So... how must does free incoming cost you? ;) |
| 03:03.42 | NovceGuru | for the hosted plan its $7.95 |
| 03:03.44 | Entranced | $3.79 /mo |
| 03:03.58 | Entranced | including DID |
| 03:04.05 | NovceGuru | http://www.vitelity.com/index.php?p=retailserv doesn't seem too bad but I wish they offered a "unlimited" plan |
| 03:04.16 | x86 | les.net ftw |
| 03:04.28 | Entranced | do they offer international ? |
| 03:04.45 | Entranced | lemme check |
| 03:04.49 | NovceGuru | les.net website makes me want to puke |
| 03:05.00 | x86 | support is GREAT |
| 03:05.05 | x86 | they have awesome service too |
| 03:05.12 | x86 | and it's cheap! $0.008 per minute |
| 03:05.36 | NovceGuru | if we mention #asterisk on freenode do you get kickbacks ;P |
| 03:05.46 | NovceGuru | *rephrase* |
| 03:05.55 | NovceGuru | if we mention #asterisk on freenode, can we split the kickbacks |
| 03:07.16 | Entranced | x86, the international rates are a bit higher... not by much though |
| 03:07.30 | x86 | Entranced: tis why you use LCR :) |
| 03:08.40 | *** join/#asterisk rabbs (n=qwer@4801F05A.ptr.provps.com) |
| 03:08.53 | Entranced | I want some of that LCR |
| 03:09.07 | x86 | least cost routing |
| 03:10.37 | Entranced | yeah... I need to get rid of my cell |
| 03:11.57 | Entranced | If I spent a lot of time on the phone I would consider having multiple providers... |
| 03:12.36 | x86 | make a lot of international calls? |
| 03:12.38 | Entranced | I will end up spending more per month that on actual minutes used |
| 03:12.54 | x86 | for les.net, you only pay for what you use |
| 03:12.56 | Entranced | couple a week |
| 03:12.58 | x86 | no monthly fees, et |
| 03:12.59 | x86 | etc |
| 03:13.09 | Entranced | that's nice |
| 03:14.15 | Entranced | that actually may work out then |
| 03:16.42 | Entranced | Where did you see the $0.008 per/min deal ? |
| 03:16.51 | Entranced | Usage Fee: $0.015/Minute |
| 03:25.59 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:26.41 | drmessano | les.net is pretty cool.. Doesn't look as high class as some of the other places (fancy website, etc), but they're damn good |
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| 03:35.23 | jameswf-home | dah dah dah |
| 03:42.42 | lmadsen | had had had |
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| 04:08.10 | *** join/#asterisk booh- (n=Sabrina@bas12-montrealak-1167977420.dsl.bell.ca) |
| 04:10.40 | booh- | is there a place where I can find the meaning (differences) between asterisk, asterisknow, zaptel, switchvox, freepbx, trixbox ? I think I need some definition to understand... |
| 04:12.02 | drmessano | Go to each IRC channel and ask about all the others |
| 04:12.02 | drmessano | You'll learn fast |
| 04:12.02 | booh- | also, wich distro is the best for asterisk? I plan to use netbsd or Ubuntu |
| 04:12.31 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 04:16.00 | [Outcast] | booh-, http://www.voip-info.org |
| 04:17.26 | lmadsen | booh-: whatever you're most comfortable with, but a Linux distribution is going to have a lot more documentation, and is going to be simpler to administer than a BSD distro |
| 04:18.07 | lmadsen | or check out the website of each project |
| 04:18.19 | lmadsen | (in re: to the differences between the various packages) |
| 04:18.26 | lmadsen | jbot_: tell booh- about thebook |
| 04:21.03 | booh- | ok. and for hardware, I should buy "digidum" stuff? (I have 5 cascading lines, 1 fax line, 19 fixed phone extensions. I don't know at this moment what will be my need for other site office or teleworkers or mobils... and for phones itselves, wich ip phone is the best for physical set at the office? |
| 04:21.34 | lmadsen | booh-: Digium hardware supports the project, so that would be the ideal :) |
| 04:21.51 | [TK]D-Fender | booh-: For IP Phones, Polycom <- |
| 04:22.01 | [TK]D-Fender | booh-: Model to choose is dependent on your needs. |
| 04:22.10 | [TK]D-Fender | booh-: www.telephonydepot.com |
| 04:22.28 | [TK]D-Fender | booh-: very fast shipping across the border. |
| 04:23.27 | booh- | I noted... |
| 04:24.26 | booh- | I own some Mitel 5212 phone sets... Can I do some testing with them before buying other ones? |
| 04:24.36 | lmadsen | yep |
| 04:24.50 | lmadsen | I have a 5220 and it works fine |
| 04:24.54 | booh- | At this moment I have Nortel Meridian phone system... ;) |
| 04:25.06 | lmadsen | as long as the 5212 supports SIP |
| 04:25.18 | booh- | yes 5212 support SIP. |
| 04:26.44 | booh- | In fact, we have moved last week ... and I made all cables with cat-5E for phone sets... so it will be easy to switch to IP Phone system. |
| 04:27.08 | booh- | (if I can make Asterisk working... and answer to my needs.. to replace the nortel Meridian) |
| 04:27.34 | booh- | I just don't know what kind of adventure it will be... |
| 04:28.13 | [TK]D-Fender | booh-: I got to ditch my company's Norstar system almost 3 years ago for *. |
| 04:28.59 | [TK]D-Fender | booh-: I snicker every time I see a company running with a 7310 in public view with the passwords defaulted :) |
| 04:29.13 | booh- | It's much more an interesting project to switch to asterisk/voip than a real need... (lines with my pstn will stay analog... I will use voip only inside the office and to reach branch office...) |
| 04:31.00 | booh- | I don't have the lan connected to my norstar system ... it's too old, I don't have this possibility... |
| 04:31.17 | booh- | and I hate to have to program on the phone screen :( |
| 04:35.20 | [TK]D-Fender | booh-: Norstar is a dead-end appliance. VM is a PITA, restore/setup too, AA? bleh |
| 04:44.19 | shtoom | hi, what is the audio format that is available on eagi's file descriptor 3 ? |
| 04:45.23 | Juggie | the 8 ball says, slin (RAW) |
| 04:46.38 | shtoom | juggie - I tried to play it with aplay all i can hear is noise |
| 04:47.46 | Juggie | of course |
| 04:47.47 | Juggie | its raw |
| 04:47.55 | Juggie | aplay has no idea what data is in the file |
| 04:48.10 | Juggie | 8 bit or 16bit, 8000hz, 16000hz etc |
| 04:48.30 | Juggie | (its 16bit, 8000hz) |
| 04:52.57 | shtoom | ok , aplay -t raw -f S16_LE testeagi did the trick |
| 04:53.35 | Juggie | your welcome |
| 04:54.48 | shtoom | juggie: slin is different from pcm ? |
| 04:56.23 | Juggie | i believe so yes |
| 04:56.41 | Juggie | yes it is |
| 04:57.23 | Juggie | slin = 16bit Signed Linear PCM |
| 04:58.23 | Maliuta | is bored |
| 04:58.47 | Maliuta | there is nothing interesting going on, and no stupid people to play head games with |
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| 06:47.06 | mike8901 | Is it normal to pay for minutes once you forward a call? |
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| 07:04.46 | dieno | hello is anyone fimiliar with this command L(x[:y][:z]) |
| 07:14.01 | kaldemar | actually that is not a command but an option for the dial application. |
| 07:14.12 | dieno | how do i add it |
| 07:14.21 | dieno | and where should i |
| 07:14.35 | kaldemar | have you taken a look at CLI output for "core show application dial"? |
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| 08:09.09 | mandh | Hi all , i have 7 zap chanelss and using them for outgoing call , but if some of them down , the asterisk still try connect to it how to fix that please |
| 08:12.49 | tzafrir_laptop | mandh, what type of zap channels? FXO? |
| 08:13.28 | tzafrir_laptop | Do you use Zap/g0 or something similar? |
| 08:13.49 | mandh | the zap channels connected to PSTN |
| 08:13.54 | mandh | yeah |
| 08:14.37 | mandh | group => 1 |
| 08:14.37 | mandh | channel => 1,2,3,4,5,6,7 |
| 08:15.45 | *** join/#asterisk matrix1233 (n=Administ@196.203.40.207) |
| 08:16.05 | matrix1233 | hello |
| 08:16.50 | synthetiq | are you using chanavail command? |
| 08:17.26 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 08:17.39 | matrix1233 | any one can help me with the Asterisk Call Manager/1.0 |
| 08:18.20 | *** join/#asterisk Swat2 (n=bler@218-214-169-112.people.net.au) |
| 08:19.19 | mandh | i use it like |
| 08:19.20 | mandh | ;Outcalls |
| 08:19.21 | mandh | exten => _9.,1,Dial(Zap/g1/${EXTEN:1}) |
| 08:22.05 | *** join/#asterisk mosty (n=mosty@124-168-216-41.dyn.iinet.net.au) |
| 08:24.26 | mandh | synthetiq, can chanavial check if the some channel alive? |
| 08:27.13 | tzafrir_laptop | mandh, what card do you use? What version of zaptel? |
| 08:27.16 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) |
| 08:27.47 | tzafrir_laptop | Also, what do you mean by "down"? |
| 08:28.05 | mandh | i mean i have 7 phone line |
| 08:28.14 | mandh | and all of them connected to the card |
| 08:28.25 | mandh | but some times there is problem in phone line |
| 08:28.34 | synthetiq | yea mandh |
| 08:28.41 | mandh | now for exmaple |
| 08:28.53 | mandh | the phone line number 7 have problem |
| 08:29.13 | mandh | but channel => 1,2,3,4,5,6,7 for outgoing call |
| 08:29.30 | mandh | so some times the system chosse the line that may be not work |
| 08:30.24 | mandh | i can change it manualy but to make system check lines status and if one not work , it will not send outgoing calls to it |
| 08:31.11 | mosty | is there an iax equivalent of the SIPCHANINFO function? |
| 08:31.39 | Swat2 | Can anyone tell me how to make it so that when a call comes in with no caller ID, how to set a caller ID of "Unknown" <00-000-000> for incomming Zaptel Calls |
| 08:31.58 | Swat2 | wow that was bad english |
| 08:32.24 | Swat2 | Can anyone tell me how to make it so that when a call comes in with no caller ID, set a caller ID of "Unknown" <00-000-000> for incomming Zaptel Calls |
| 08:33.03 | mosty | Swat2, use a GotoIf which jumps somewhere if the callerid is empty, then use Set and the CALLERID function to set callerid to whatever you like |
| 08:34.22 | Swat2 | hrmm |
| 08:37.26 | mandh | tzafrir_laptop, TDM2004p zaptle 1.4.5 |
| 08:37.58 | tzafrir_laptop | and what do you mean by "down"? |
| 08:38.22 | tzafrir_laptop | what type of problem? |
| 08:38.58 | mandh | down if some one of phone lines not work |
| 08:39.46 | tzafrir_laptop | if you connect a normal analog phone to that line, will it give ring tone? |
| 08:40.09 | mandh | tzafrir_laptop, yeah it will not ring |
| 08:41.05 | tzafrir_laptop | I think that with newer versions of Zaptel it should be able to report such events. But I'm not sure |
| 08:42.15 | mandh | tzafrir_laptop, so u think that at 1.4.5 it cant report that , do u think there is any application that can be used to check the status of the line? |
| 08:42.16 | *** join/#asterisk J4zen (n=Jeroen@a82-95-153-17.adsl.xs4all.nl) |
| 08:42.54 | tzafrir_laptop | not with zaptel 1.4.5 . Try zaptel 1.4.11, maybe |
| 08:44.18 | J4zen | Hi there, Does anyone know an existing PBX or Fork that supports multi-tenancy preferably with GUI(end-user GUI at least)? I'm sure this questions popups up fairly often; But bare with me, i've searched the magic world of google and can only find "Future plans". |
| 08:45.12 | *** join/#asterisk pa (n=pa@unaffiliated/pa) |
| 08:48.34 | *** join/#asterisk OcterA (n=octera@ks37167.kimsufi.com) |
| 08:48.41 | OcterA | hi there |
| 08:48.51 | *** join/#asterisk xenonex (n=xenonex@92.47.2.149) |
| 08:50.04 | *** join/#asterisk tcseke (n=chatzill@217.20.134.239) |
| 08:51.43 | *** join/#asterisk rcy (n=rcy@S010600003981572c.vc.shawcable.net) |
| 08:52.17 | OcterA | can anyone tell me what's is the prereq for the DTMF inband detection in SIP? Because I have two asterisk one on a i386 computer and one on a arm processor. The i386 detect the dtmf inband and not the arm. The two server have the same sip.conf and extensions.conf the phone are same model |
| 08:55.52 | *** join/#asterisk lenscape (n=lenscape@208.50.101.130) |
| 08:56.52 | lenscape | has anyone heard of sipproxd? A SIP proxy daemon. I can't find a URL for it |
| 08:59.51 | mosty | OcterA, sip inband dtmf is crappy, why aren't you using a better dtmf method? |
| 09:05.16 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 09:05.24 | mort_gib | Morning |
| 09:06.14 | mort_gib | I need to send voicemails to recepients via an external smtp server, that requires authentication -Can this be done, apart from configuring sendmail to do so?? |
| 09:06.42 | Swat2 | how do you manually enter an object into the asterisk internal Phonebook |
| 09:08.25 | OcterA | mosty : because my card is linked with a sip provider, wich give me the dtmf inband only |
| 09:09.23 | tzafrir_laptop | J4zen, Asterisk sure supports that. But you seem to want a GUI to wrap this in a pretty suite for suites, right? |
| 09:09.54 | tzafrir_laptop | I was told VoiceOne can handle this nicely. Never really tried it. |
| 09:10.29 | tzafrir_laptop | DeStar was also designed for multi-tennant support. But its development has rather stalled |
| 09:11.08 | tzafrir_laptop | J4zen, or simply make yourself familiar with the concept of multiple contexts |
| 09:11.34 | J4zen | True, i could put some guys on it but it'd cost a fair bit of time |
| 09:11.39 | J4zen | i need a rapid-deployment solution |
| 09:12.05 | J4zen | I'll look into VoiceOne, you don't happen to have any expierence with Thirdlane PBX MTE? |
| 09:12.38 | mosty | OcterA, card for what? |
| 09:12.52 | tzafrir_laptop | J4zen, as a rule I don't wish anybody to go the non-free route |
| 09:13.48 | J4zen | Naturally, but for now it was all i could find about solid multi-tenant support in GUI-based asterisk |
| 09:14.02 | J4zen | i'd be glad to go the free opensource way, if only there was one' |
| 09:14.20 | J4zen | If i remember correctly, VPS isn't the way to go with Asterisk right? |
| 09:14.28 | J4zen | even though im not using any BRI/PRI interfaces |
| 09:14.37 | J4zen | i remember there being some latency issues |
| 09:15.01 | tzafrir_laptop | People do use it |
| 09:15.38 | J4zen | There will always be people doing it anyway, what would you recommend? |
| 09:17.12 | OcterA | mosty : the devellopement board with the arm processor (i'm french, we tell this a "carte" => card) |
| 09:27.15 | mosty | OcterA, you should be able to do sip dtmf in software easily, without inband dtmf |
| 09:29.14 | *** join/#asterisk rootlogin (n=root@saturn2.franken.de) |
| 09:31.29 | OcterA | i need dtmf inband, because i have a provider sip, and he gimme the dtmf inband only, not in rfc. |
| 09:32.10 | OcterA | another things, on the board, the generating of dtmf don't work too (dtmfsend()) in the extensions.conf |
| 09:32.46 | *** join/#asterisk tuxd00d (n=tuxd00d@128.187.132.25) |
| 09:36.05 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
| 09:39.25 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 09:39.59 | OcterA | ztdummy is required to god detect dtmf? |
| 09:46.10 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 09:46.23 | dandre | Hello, |
| 09:47.06 | dandre | Is there a dialpattern like _X! but that can handle letters too ? |
| 09:48.51 | *** join/#asterisk Madkiss (i=madkiss@freenode/staff-emeritus/madkiss) |
| 09:48.54 | Madkiss | hi all. |
| 09:49.07 | Madkiss | after having made changes to extensions.ael, do I need to restart asterisk to make the changes take effect? |
| 09:49.37 | Aurs | Madkiss: isn't it enough with a "extensions reload" ? |
| 09:50.03 | Madkiss | how to get an overview over the currently configured extensions? |
| 09:51.44 | Aurs | hmm.. dialplan show? |
| 09:52.14 | Aurs | or show dialplan |
| 09:54.07 | *** join/#asterisk Rico29 (n=Rico@sto93-4-88-162-214-225.fbx.proxad.net) |
| 09:54.10 | *** join/#asterisk aksyn (n=aksyn@78.86.127.229) |
| 09:54.24 | lenscape | has anyone heard of sipproxd? A SIP proxy daemon. I can't find a URL for it |
| 09:55.39 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 09:55.51 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.135) |
| 09:55.59 | *** join/#asterisk kannan (n=kann@123.201.60.110) |
| 09:57.40 | JT | lenscape: never heard of it |
| 10:00.10 | lenscape | JT: no. Nor me |
| 10:00.27 | lenscape | What's the normal solution to several SIP phones behind a NAT gateway? |
| 10:01.14 | *** join/#asterisk aksyn (n=aksyn@78.86.127.226) |
| 10:06.13 | Madkiss | slaps himself |
| 10:06.24 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 10:06.48 | Madkiss | what a crap. well, whatever. i would like to make my asterisk support conference rooms. e.g. I call a certain number, then I enter a passcode and voila, I'm in a conference |
| 10:06.58 | Madkiss | is there some example available for this? |
| 10:09.09 | styelz | lenscape: you mean http://siproxd.sourceforge.net/ ? |
| 10:09.20 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 10:09.54 | lenscape | styelz: you hero! Thanks. I was looking for sipproxd which is how it's listed in some places |
| 10:12.25 | *** join/#asterisk rcy` (n=rcy@S010600003981572c.vc.shawcable.net) |
| 10:15.02 | *** join/#asterisk _ys (i=yuri@91.151.196.254) |
| 10:26.49 | *** join/#asterisk oej (n=olle@bkkb-gw.voop.net) |
| 10:28.09 | *** join/#asterisk icenick82 (n=pavlovni@83.244.78.241) |
| 10:28.17 | icenick82 | hi |
| 10:28.57 | icenick82 | i am new to asterisk and i want to ask questions abour codec support |
| 10:29.13 | icenick82 | my asterisk right now supports g711 codec |
| 10:29.27 | icenick82 | how much is it easy to install g729 and g723? |
| 10:29.41 | icenick82 | whar r the requirements? licenses? |
| 10:31.17 | Madkiss | what's the command to show meetme-information on the asterisk command line? |
| 10:35.55 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.135) |
| 10:35.59 | styelz | Madkiss: meetme ? |
| 10:36.04 | *** join/#asterisk kannan (n=kann@123.201.60.110) |
| 10:37.44 | styelz | Madkiss: http://www.voip-info.org/wiki-Asterisk+cmd+MeetMe |
| 10:37.50 | Madkiss | thanks. |
| 10:38.15 | Madkiss | now the only interesting question left is. why can't I type "901" on my sip-phone and take part in the conference, while calling the extension 901 from the outside works perfectly? |
| 10:39.37 | styelz | dont know, maybe its not in context |
| 10:39.38 | *** join/#asterisk vgster (n=vgster@93.96.221.240) |
| 10:42.34 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 10:47.39 | *** join/#asterisk imcdona (n=t@c-24-19-99-60.hsd1.mn.comcast.net) |
| 11:02.28 | JT | lenscape: solution? it should just work. |
| 11:03.21 | lenscape | JT: How? You can't forward port 5060 to a bunch of phones |
| 11:03.40 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 11:04.07 | JT | lenscape: phones are endpoints. endpoints do not neet port forwarding. |
| 11:04.17 | JT | s/neet/need/ |
| 11:04.45 | *** join/#asterisk Dr-Linux|home (n=Nothing@117.20.21.66) |
| 11:05.43 | Dr-Linux|home | I'm not getting CallerID on my analog lines, where it was working fine. Which thing needs to be checked? |
| 11:05.48 | lenscape | JT: I have a single SIP phone on a LAN behind NAT. the SIP gateway is out on the net somewhere. How do you address the phone? |
| 11:05.49 | icenick82 | g729 is free for asterisk? |
| 11:05.57 | Dr-Linux|home | there is nothing modified |
| 11:06.03 | JT | lenscape: get the phone to register. |
| 11:06.41 | lenscape | JT: yep. That works but it still can't make or receive calls (well it can - but there's no audio) |
| 11:07.11 | *** join/#asterisk l0verb0y (n=l0verb0y@202.128.53.3) |
| 11:07.12 | lenscape | JT: the phone rings but there's silence in both directions |
| 11:07.26 | JT | lenscape: it will if you have your sip server setup correctly and there is nothing stupid like a "sip fixup" mode enabled on the nat router |
| 11:07.38 | JT | also a small handful of nat routers just suck and do not work |
| 11:08.01 | lenscape | JT: sip server is www.sipgate.co.uk |
| 11:08.25 | lenscape | router is Linux (Suse 10.2 with Shorewall) |
| 11:08.38 | JT | ensure the ports aren't blocked then |
| 11:09.16 | lenscape | JT: port isn't blocked but where's it supposed to be forwarded? |
| 11:09.30 | JT | it's NOT supposed to be forwarded |
| 11:09.38 | JT | make sure the RTP ports are not blocked |
| 11:09.47 | lenscape | which are the RTP ports? |
| 11:09.59 | JT | check the SIP SDP to see what RTP ports (UDP) sipgate is trying to use |
| 11:10.03 | JT | the ports which the media uses |
| 11:10.04 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 11:10.08 | JT | no media = no audio |
| 11:10.51 | *** join/#asterisk packetstream (n=chatzill@87-194-74-202.bethere.co.uk) |
| 11:10.55 | lenscape | JT: so from what you're telling me - all SIP traffic is outbound? |
| 11:11.01 | JT | no |
| 11:11.06 | lenscape | sorry - all connections are outbound |
| 11:11.07 | JT | all sip is is signalling |
| 11:11.07 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 11:11.10 | JT | it's bi-directional |
| 11:11.34 | JT | you do not need to port forward to make sip work and endpoint sites unless the router is broken |
| 11:11.38 | JT | a/and/at/ |
| 11:11.42 | lenscape | JT: so how does an inbound connection find its target? |
| 11:11.58 | JT | the endpoint regularly registers |
| 11:12.03 | lenscape | right |
| 11:12.14 | lenscape | so the connection is initally made outbound then |
| 11:12.53 | JT | there are hundreds of thousands of consumers in homes with ATAs connecting to ITSPs, it would be a nightmare if the ITSPs had to guide all these non-technical users through setting up port forwarding on their myriad of routers |
| 11:12.54 | lenscape | there must be an active connection made outbound for external traffic to find its target |
| 11:13.11 | JT | sure, sip connections should stay alive |
| 11:13.23 | lenscape | ok. I understand a bit more. Thanks |
| 11:15.50 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 11:20.21 | *** join/#asterisk hackeron (n=hackeron@gentoo/user/hackeron) |
| 11:20.43 | redax | hi, |
| 11:21.11 | hackeron | hey, how would I execute an external command with asterisk? -- say I want to stop rtorrent when there is an incoming call and restart it when the call is over |
| 11:21.14 | redax | I'd like to interconnect an Ericsson BusinessPhone and Asterisk using a Digium WildCard B4S0 |
| 11:21.29 | redax | the call works from the ericcson to the asterisk |
| 11:21.53 | redax | but from the asterisk to ericsson it's going to be congestion |
| 11:22.05 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 11:22.07 | redax | do you have any idea? |
| 11:22.38 | redax | and I have `hfcmulti_rx: CRC-error' lines in my kernelmsgs :=/ |
| 11:22.51 | *** join/#asterisk gomel77 (n=gomel77@leased-line-244-82.telecom.by) |
| 11:23.52 | icenick82 | r g729 and g723 free for asterisk? |
| 11:26.56 | redax | quit |
| 11:27.01 | redax | sorry :) |
| 11:30.21 | NovceGuru | <PROTECTED> |
| 11:30.28 | NovceGuru | actually if the router follows RFC it breaks sip :( |
| 11:34.12 | icenick82 | > r g729 and g723 free for asterisk? |
| 11:35.51 | J4zen | What would be the best aproach to actually calculate or estimate the computing power needed for a specified amount of concurrent calls/registered extensions? |
| 11:36.20 | J4zen | For example, 20 sip phones; 5 concurrent calls. How'd i go by calculating the needed hardware? |
| 11:37.20 | NovceGuru | http://www.voip-info.org/wiki/view/Asterisk+dimensioning |
| 11:37.41 | icenick82 | anyone for my question about the codecs? |
| 11:37.51 | icenick82 | <PROTECTED> |
| 11:38.00 | JT | NovceGuru: what are you talking about? the router should treat sip/rtp traffic as if it were any other udp traffic |
| 11:38.59 | J4zen | Thanks NovceGuru :) |
| 11:39.46 | NovceGuru | I think something to do with source-port rewriting |
| 11:41.07 | NovceGuru | I can't get more then 1 sip device behind one public IP connecting to an external proxy using openbsd/pf |
| 11:41.17 | JT | NovceGuru: all nat routers need to do is to treat sip like any other udp packet and not to change any of the payload |
| 11:41.49 | JT | NovceGuru: that's weird, i have no problem getting multiple sip devices working behind nat with iptables, and lots of cheap crappy consumer embedded routers |
| 11:42.33 | NovceGuru | Yeah, it's a known issue :\ |
| 11:42.50 | NovceGuru | Sucks when the provider says "replace your soekris box with a wrt54g" |
| 11:43.35 | NovceGuru | though can be fixed with multiple public IPs using 1:1 nat or using the siproxd package |
| 11:45.26 | JT | or by using a nat device setup correctly |
| 11:45.32 | JT | 1:1 nat is a stupid solution |
| 11:45.42 | JT | ditto for siproxd, pretty impractical |
| 11:50.41 | icenick82 | anyone familiar with the asterisk codecs?\ |
| 11:54.08 | icenick82 | JT i asked a lot of times |
| 11:54.24 | icenick82 | anyone familiar with asterisk codecs please/ |
| 11:54.26 | icenick82 | ? |
| 11:55.21 | JT | ~ask |
| 11:55.22 | jbot | [ask] Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 11:56.31 | icenick82 | g723 and g729 are free or should i pay to get them? |
| 11:57.03 | JT | they are not free to transcode |
| 11:57.44 | icenick82 | can i enable asterisk with free g723 and g729 without having to pay? |
| 11:58.24 | ltd | sure, but just don't get caught using it commercially :P |
| 12:00.27 | redax | hm. seems like the asterisk isn't allocate B1-Channel to the ericsson call |
| 12:04.41 | redax | aaa. maybe misdn in NT mode and PTP is still buggy? |
| 12:07.21 | JT | VERY buggy |
| 12:09.24 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 12:09.24 | *** mode/#asterisk [+o russellb] by ChanServ |
| 12:10.24 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:11.13 | *** join/#asterisk c909 (n=c909@p4FC4CE4F.dip.t-dialin.net) |
| 12:11.28 | c909 | hello @ all |
| 12:11.45 | c909 | I've trouble with my asterisk gui |
| 12:13.53 | gr0mit | and? |
| 12:16.13 | russellb | c909: which one are you using? |
| 12:18.03 | *** join/#asterisk RoyK (n=roy@ip-192-27-149-91.dialup.ice.no) |
| 12:18.24 | *** join/#asterisk shido6 (n=shido6@74-130-224-188.dhcp.insightbb.com) |
| 12:18.45 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:20.55 | c909 | can s.o. help me troubleshooting my gui login??? |
| 12:22.12 | tzafrir_laptop | ~ask |
| 12:22.15 | jbot | [ask] 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. |
| 12:22.15 | russellb | i tried asking a question and you didn't answer, so I guess not |
| 12:26.55 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 12:27.55 | *** join/#asterisk msetim (n=msetim@200.195.161.164) |
| 12:28.08 | redax | JT: seems like misdn NT mode +PTP not allocating B1 channel @ Channel Identification phase |
| 12:29.19 | [TK]D-Fender | c909: GUI's are not supported in this channel. Please look at the channel topic for the appropriate one. |
| 12:30.26 | JT | redax: do you have to use misdn? |
| 12:30.51 | c909 | I tried this, but there seems to be nobody awake ;-) |
| 12:31.41 | c909 | but I think its a well known problem, I'm getting a 404 error during login process... |
| 12:32.57 | russellb | and you _still_ haven't answered the first question I asked you! |
| 12:33.10 | russellb | and without that, nobody can even begin to help |
| 12:33.23 | russellb | or even point you in the right direction |
| 12:33.35 | russellb | i know tzafrir_laptop just did this, but ... |
| 12:33.35 | russellb | ~ask |
| 12:33.38 | jbot | it has been said that ask is 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. |
| 12:35.09 | [TK]D-Fender | c909: Just because "nobody is awake" there doesn't make this a fall-back. Wait some more, or get Googling. |
| 12:39.37 | c909 | k |
| 12:42.48 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 12:44.08 | x86 | [2008-06-10 07:42:45] NOTICE[18731] chan_iax2.c: Out of idle IAX2 threads for I/O, pausing! |
| 12:44.14 | x86 | wtf does this mean? |
| 12:46.32 | *** join/#asterisk grEvenX (n=even@ap39pb.ip.ssc.net) |
| 12:54.57 | russellb | x86: you can ignore it ... |
| 12:55.00 | russellb | it's a debug message ... |
| 12:56.22 | *** join/#asterisk km2 (n=x@c-24-23-255-173.hsd1.mn.comcast.net) |
| 12:56.33 | [TK]D-Fender | x86: ... You "noticed"! |
| 12:58.19 | x86 | well kind of hard to ignore that when it happens, I can no longer originate or terminate any iax2 calls, and unloaded chan_iax2.so and re-loading it does not solve the problem, I have to kill asterisk completely |
| 12:59.22 | *** join/#asterisk Prog-javApplet (n=Prog-Jav@41.224.253.239) |
| 12:59.30 | Prog-javApplet | hello |
| 12:59.43 | Prog-javApplet | can someone help to install asterisk server |
| 12:59.49 | Prog-javApplet | on fedora |
| 12:59.49 | Prog-javApplet | ? |
| 13:00.42 | russellb | download asterisk-1.4.21-rc2.tar.gz |
| 13:00.49 | russellb | tar -xvzf asterisk-1.4.21-rc2.tar.gz |
| 13:00.50 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 13:00.53 | russellb | cd asterisk-1.4.21-rc2 |
| 13:00.54 | russellb | ./configure |
| 13:00.55 | russellb | make |
| 13:00.57 | russellb | sudo make install |
| 13:00.58 | russellb | done! |
| 13:01.55 | Prog-javApplet | checking how to run the C++ preprocessor... /lib/cpp |
| 13:01.55 | Prog-javApplet | configure: error: C++ preprocessor "/lib/cpp" fails sanity check |
| 13:01.55 | Prog-javApplet | See `config.log' for more details. |
| 13:02.06 | russellb | you need to install gcc |
| 13:02.06 | Prog-javApplet | i have this erroro on ./configure process |
| 13:02.09 | [TK]D-Fender | Prog-javApplet: Time to install the pre-requisites |
| 13:02.21 | Prog-javApplet | ok |
| 13:02.23 | russellb | you can run the contrib/scripts/install_prereq script |
| 13:02.31 | russellb | actually, is that in 1.4? *looks* |
| 13:02.33 | [TK]D-Fender | Prog-javApplet: you seem to have not bothered installing the basic compiling tools |
| 13:02.46 | [TK]D-Fender | russellb: yup, last I checked |
| 13:02.53 | russellb | i don't see it |
| 13:03.19 | russellb | oh well, there aren't many things to install ... |
| 13:03.20 | [TK]D-Fender | russellb: Actually... wasn't that a "make" option? |
| 13:03.27 | russellb | don't think so ... |
| 13:03.27 | redax | heh. the mISDN debug log seems like it's trying to create a new channel; meanwhile the ericsson trace shows there no channel allocation |
| 13:04.03 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
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| 13:08.31 | Prog-javApplet | [root@localhost scripts]# apt-get install gcc |
| 13:08.32 | Prog-javApplet | Reading Package Lists... Done |
| 13:08.32 | Prog-javApplet | Building Dependency Tree... Done |
| 13:08.32 | Prog-javApplet | gcc is already the newest version. |
| 13:08.52 | Prog-javApplet | it's not gcc the problem |
| 13:09.58 | *** part/#asterisk DanyWalker (n=ww@201.230.26.10) |
| 13:10.33 | [TK]D-Fender | Prog-javApplet: Yes, you are missig G++ |
| 13:10.42 | [TK]D-Fender | Prog-javApplet: do "yum install gcc*" |
| 13:11.05 | [TK]D-Fender | Prog-javApplet: that should grab the rest of them. |
| 13:12.16 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:13.02 | x86 | [TK]D-Fender: funny how he's running debian and you tell him to use yum ;) |
| 13:13.25 | [TK]D-Fender | x86: [08:59]<Prog-javApplet>on fedora |
| 13:13.32 | [TK]D-Fender | x86: O RLY? |
| 13:13.38 | tzanger | *sigh* |
| 13:13.41 | x86 | 08:08 < Prog-javApplet> [root@localhost scripts]# apt-get install gcc |
| 13:13.42 | tzanger | I love descriptive error messages |
| 13:13.49 | tzanger | "you hae a problem with your zaptel config. fuck off." |
| 13:13.59 | [TK]D-Fender | x86: You read the smallprint, I'll rad the BIG PRINT ;) |
| 13:14.23 | x86 | I'll rad whatever I wanna! |
| 13:14.25 | Prog-javApplet | i have apt-get installed on fedora |
| 13:14.27 | Prog-javApplet | :) |
| 13:14.35 | x86 | Prog-javApplet: you should probably use yum |
| 13:14.44 | Prog-javApplet | lol |
| 13:14.51 | Prog-javApplet | the package is already installed |
| 13:15.00 | x86 | gcc is, yes, but that's not g++ |
| 13:15.14 | x86 | gcc is a C compiler, g++ is a C++ compiler |
| 13:15.16 | [TK]D-Fender | Prog-javApplet: Do not forget the "*" I put in there. It was on purpose <- |
| 13:15.47 | dominic1 | Why doesn't that work as expected? if("${CALLERID(NUM)}" != "${EXTEN:4}) |
| 13:16.03 | Prog-javApplet | ok |
| 13:19.19 | x86 | anyone know if amoxicillin expires? and if so, is it ok to take it after expiration? |
| 13:19.26 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 13:20.53 | tzanger | ok that weird error has something to dow ith the zaptel timer not doing its thing. It'd be nice if it said something to the effect. |
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| 13:32.40 | tzafrir_laptop | x86, yum is available on Debian, if you really want. Though of little use |
| 13:32.43 | *** join/#asterisk DavidR2008 (n=chatzill@fw1.safedataisp.net) |
| 13:32.56 | x86 | tzafrir_laptop: he's running fedora and using apt |
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| 13:36.04 | DavidR2008 | I have written a small bash script that every four seconds executes asterisk -rx "core show channels" to get a count of active connections and display a list of active channels. Before I go and write a script to parse the output and write it into a table, is there a way to have asterisk directly write the data to a database table? I need the real time data, so I don't think the CDR records... |
| 13:36.06 | DavidR2008 | ...will do it because I understand that they are written after the fact. |
| 13:36.10 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 13:39.17 | *** join/#asterisk xnixan (n=xnixan__@unaffiliated/xnixan) |
| 13:42.50 | [TK]D-Fender | DavidR2008: You should probably be doing this via AMI |
| 13:43.07 | [TK]D-Fender | DavidR2008: And no, there is no ready made way to DB this. |
| 13:43.55 | DavidR2008 | [TK]D-Fender: thx! |
| 13:43.55 | *** join/#asterisk leanshen (n=bob@ool-43546a71.dyn.optonline.net) |
| 13:49.09 | jblack | Are questions about polycoms appropriate here? |
| 13:49.37 | jblack | DavidR2008: Did you ever look at AMI ? |
| 13:50.00 | jblack | AMI is a way to get/change the status of asterisk, particularlly, call status |
| 13:50.21 | DavidR2008 | jblack: actually I had forgotten that I had already written a basic AMI framework for a previous project. I'm reworking it to handle this |
| 13:51.32 | [TK]D-Fender | jblack: Shoot. |
| 13:51.37 | jblack | so, if you make a simple script that feeds off * via AMI, you can dump that into a db. i'm about to write one myself |
| 13:51.40 | jblack | [TK]D-Fender: Who? |
| 13:51.53 | [TK]D-Fender | smirks and prepares his hit-list |
| 13:52.27 | jblack | I'm having a bad xen/* week, so going out in a blaze of glory seems like a grand idea this morning. :) |
| 13:53.54 | DavidR2008 | jblack: it's only tuesday give it at least one more day |
| 13:54.54 | x86 | asterisk in a VM... what a joke ;) |
| 13:56.57 | jblack | so, word has it that there was a snowstorm in spokane last night |
| 13:58.26 | J4zen | Thanks for the help today :) Bye |
| 14:01.19 | jblack | looks for a copy of "Insane : The Future of Going Postal" |
| 14:09.22 | Oy90 | is it possible to change dtmf payload type (rfc2833) on a sip channel? |
| 14:09.54 | *** join/#asterisk brodiem (n=brodiem@rrcs-24-227-171-186.sw.biz.rr.com) |
| 14:11.37 | [TK]D-Fender | Oy90: On a channel, no, on a device, yes |
| 14:13.34 | brodiem | Can anyone think of why I constantly get "RTCP SR transmission error, rtcp halted" errors on console? The only references I can find all are users with broken MOH, but I have no problems there (single class, mode=files, format is g729). It's mostly just an annoyance because even with verbose/debug=0, it writes frequently to the console |
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| 14:22.42 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:22.42 | *** mode/#asterisk [+o lmadsen] by ChanServ |
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| 14:26.49 | [TK]D-Fender | jblack: So... your Polycom question? |
| 14:26.52 | *** join/#asterisk Defraz (i=t0tal@69.92.19.83) |
| 14:27.04 | *** join/#asterisk frocos11292 (n=ask@firewall.vipvoz.com) |
| 14:27.39 | frocos11292 | Hi! |
| 14:27.50 | frocos11292 | anyone played with retrieve_conf? |
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| 14:40.42 | lmadsen | if the 'thebook' went up in price to match the number of pages (i.e. 800-1000 pages), would you still buy it? |
| 14:41.24 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.140.193) |
| 14:41.29 | brodiem | no |
| 14:43.29 | [TK]D-Fender | lmadsen: If the book went up in quality I'd say it might warrant a small bump. |
| 14:43.30 | seanbright | if it was a cent a page, maybe. |
| 14:43.46 | frocos11292 | anyone played with retrieve_conf? |
| 14:43.51 | [TK]D-Fender | lmadsen: To match / page.... not at that scale. |
| 14:44.06 | tzafrir_laptop | frocos11292, I did. But I think you should ask this in #freepbx / #freepbx-devel |
| 14:44.12 | lmadsen | well this is the problem we're running into -- adding more content causes the cost of producing the book to go up, and thus, the price of the book needs to go up |
| 14:44.54 | lmadsen | so either we just update what we have and leave the thickness the same (and thus the same price), or we add content *and* update what is in the book, which increases thickness... etc.... |
| 14:44.59 | frocos11292 | tzafrir> could you just point me in the right direction? |
| 14:45.20 | frocos11292 | tzafir-> it seems that i miss a pre-reload or something... |
| 14:45.30 | frocos11292 | Use of uninitialized value in string eq at /var/lib/asterisk/bin/retrieve_op_conf_from_mysql.pl line 415 |
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| 14:52.23 | a1fa | hm.. why is recorded audio choppy in asterisk running on xen? |
| 14:52.25 | hsv-al | hola |
| 14:54.18 | x86 | dude |
| 14:54.22 | a1fa | what |
| 14:54.25 | x86 | asterisk sucks in a VM |
| 14:54.27 | x86 | don't do it |
| 14:54.29 | a1fa | no shit ;P |
| 14:54.39 | *** join/#asterisk blop (i=blop@219-186-112-217.dyn.adsl.belcenter.be) |
| 14:54.44 | a1fa | limited resources.. no choice |
| 14:54.45 | blop | hi there |
| 14:54.58 | a1fa | its either bandwidth restrictions, or vm |
| 14:55.09 | blop | i've got a weird behaviour with my tdm400p :/ |
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| 14:55.21 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:56.08 | x86 | a1fa: my bet is timing |
| 14:56.16 | a1fa | someone said that |
| 14:56.30 | a1fa | d-fender says timing is only used for meetme |
| 14:56.37 | a1fa | and iax2 trunk |
| 14:57.17 | [TK]D-Fender | And helps optimize MoH, and MeetMe is used in app_page. |
| 14:57.35 | a1fa | :) |
| 14:57.35 | blop | when i receive a call on a fxo port, its working normally, makes 2 fxs ring, then its hanging up and behave like a second call is incoming on that same fxo (altough its always the same call ringing), any idea why? |
| 14:57.42 | a1fa | ivr is all f000keeed |
| 14:57.44 | a1fa | clicky and crap |
| 14:57.50 | *** join/#asterisk enemy^x (n=enemy@c213-158-248-202.static.sdsl.no) |
| 14:57.53 | a1fa | it never bothered me until someone pointed it out to me |
| 14:58.10 | a1fa | i wonder if 1.6 suffers from premature clickination |
| 14:58.26 | brodiem | maybe I'll ask this again since there's more activity now... |
| 14:58.32 | brodiem | Can anyone think of why I constantly get "RTCP SR transmission error, rtcp halted" errors on console? The only references I can find all are users with broken MOH, but I have no problems there (single class, mode=files, format is g729). It's mostly just an annoyance because even with verbose/debug=0, it writes frequently to the console |
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| 15:09.46 | blop | asterisk[25872]: segfault at 081fc000 eip 080a8f8a esp b6b50a34 error 4 |
| 15:09.48 | blop | :D |
| 15:17.44 | *** part/#asterisk nny_1 (n=Scott_My@64.203.239.83) |
| 15:17.46 | tzafrir_laptop | That's a blop, indeed |
| 15:18.03 | tzafrir_laptop | What version? |
| 15:19.06 | blop | 1.4.20 |
| 15:19.26 | blop | it crashed 6 times today while doing some tests |
| 15:20.36 | blop | again : |
| 15:20.36 | blop | asterisk[758]: segfault at 081c8278 eip b75e048f esp bf8837c0 error 4 |
| 15:20.37 | blop | asterisk[1246]: segfault at 081fc000 eip 080a8f8a esp b6a3ca34 error 4 |
| 15:20.38 | blop | :) |
| 15:20.46 | blop | grrrrrrrrrrrrrrrrrrrrrr :) |
| 15:20.51 | *** join/#asterisk AlienPenguin (n=Miranda@213.188.207.153) |
| 15:22.34 | AlienPenguin | hi, can anyone can point me towards the function that is called when the "attended transfer" feature is used? |
| 15:23.05 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 15:23.20 | *** join/#asterisk n|cotine (n=nicotine@biological.warningg.com) |
| 15:23.41 | n|cotine | Is it possible for asterisk to receive presense about remote extensions over a SIP trunk? |
| 15:23.50 | [TK]D-Fender | AlienPenguin: There is no "function" unless you're talking about some deeper procedure in the source code. |
| 15:24.02 | n|cotine | IE, can I have Asterisk know about the status of SCCP phones registered to a call manager that supports presence notification? |
| 15:24.23 | [TK]D-Fender | n|cotine: * does not receive presence, it only reports it for devices its responsible for |
| 15:25.08 | n|cotine | So there's no way for * to know the hook status of a phone registered to CCM ? |
| 15:25.10 | *** join/#asterisk jpeeler (n=jpeeler@216.207.245.1) |
| 15:25.14 | [TK]D-Fender | n|cotine: And it is not device based, it is * based. So it doesn't matter what the phone thinks it's doing, only what * thinks its doing based on its own call processing. |
| 15:25.23 | [TK]D-Fender | n|cotine: Correct. |
| 15:25.34 | n|cotine | [TK]D-Fender: Thx for the info, much appreciated. |
| 15:25.41 | *** join/#asterisk deeperror (n=deeperro@adsl-76-226-148-247.dsl.sfldmi.sbcglobal.net) |
| 15:26.45 | AlienPenguin | [TK]D-Fender : i was actually talking of source code... i tried to set a breakpoint in local_attended_trasfer in chan_sip.c (i am interested in SIP channels only) but it didt pass there as i had hoped |
| 15:28.12 | pjz | best error message so far: rtp.c:2885 bridge_native_loop: Oooh, |
| 15:28.14 | pjz | something is weird, backing out |
| 15:28.15 | *** join/#asterisk greek_user (i=ath@adsl39-104.kln.forthnet.gr) |
| 15:28.25 | [TK]D-Fender | AlienPenguin: What are you trying to accomplish? |
| 15:29.50 | AlienPenguin | [TK]D-Fender : i want to change the behaviour of the sip transfer, i'd like asterisk to show another sip from instead of the extension name |
| 15:30.09 | AlienPenguin | "sip from header" not "sip transfer" |
| 15:30.36 | [TK]D-Fender | AlienPenguin: Ok, wierd... what would you want it to show for example? |
| 15:31.00 | AlienPenguin | the sip from of the actual caller |
| 15:32.41 | *** join/#asterisk MuffinMan (n=muffinma@mctnnbsa24w-142167052094.pppoe-dynamic.nb.aliant.net) |
| 15:32.50 | file | M8824 |
| 15:33.00 | MuffinMan | [ready for testing] [Asterisk] Channels/NewFeature 0008824: [patch] Remote (called) Party Identification - chan_sip & chan_skinny implementation reported by gareth (Karma: +9.00) http://bugs.digium.com/view.php?id=8824 |
| 15:33.08 | file | you want that. |
| 15:33.12 | *** part/#asterisk MuffinMan (n=muffinma@mctnnbsa24w-142167052094.pppoe-dynamic.nb.aliant.net) |
| 15:33.25 | enemy^x | if rxfax is not generating faxtones, is that related to spandsp? |
| 15:34.33 | greek_user | Hello, I 'm trying to use the Dial(SIP\mybuddy@hissipprovider.com) although I am not a member of hissipprovider.com .. can I still call him? It seems that when I do so, and he answers my call, my Asterisk PBX doesn't understand the signal of answering, and keeps waiting until timeout. (using Asterisk 1.4.17 on ubuntu, behind a nat -- i have already solved the nat problems) |
| 15:35.32 | deeperror | Is it a known issue about cdr warnings on 3-way calls? I get the following warnings pretty much all day and reproduce this when in a 3-way call. www.pastebin.ca/1043920 |
| 15:36.19 | Strom_M | greek_user: SIP is SIP... |
| 15:36.25 | Strom_M | show the full Dial() line |
| 15:37.06 | greek_user | ok |
| 15:37.18 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 15:37.52 | greek_user | Dial(SIP/mybuddy@voipdiscount.com,30) |
| 15:37.55 | greek_user | that's all |
| 15:38.13 | Strom_M | my first instinct is to blame voipdiscount |
| 15:38.20 | greek_user | hmm |
| 15:38.31 | Strom_M | but...pastebin a sip debug |
| 15:38.50 | JT | possibly a codec negotiation issue |
| 15:39.13 | blop | why the hell is zaptel hanging up for no reason ......... on a incoming call ? :/ |
| 15:39.38 | Strom_M | blop: at what point is it hanging up? |
| 15:39.54 | Strom_M | JT: if it was a codec negotiation issue, perhaps the call wouldn't ever get set up in the first place |
| 15:40.03 | [TK]D-Fender | greek_user: Who said the voipdiscount allows you to dial their URI's like that? |
| 15:40.14 | blop | well, i call my own pstn number -> fxo ; its ringing 2 times, then asterisk hangup .. then it detects again the incoming call |
| 15:40.41 | Strom_M | blop: pastebin the relevant part of extensions.conf as well as zaptel.conf and zapata.conf |
| 15:40.43 | [TK]D-Fender | AlienPenguin: who is the "actual" caller? Sounds like you're trying to make an attended transfer into a blind transfer. |
| 15:40.46 | greek_user | i have a huge debug file here, Asterisk keeps sending INVITE requests, even when mybuddy presses the Green button, and asterisks never recieves any "green" button !! in addition, while still waiting, and he presses the red button, i immediatly get a busy message, and all is done. |
| 15:41.08 | blop | <PROTECTED> |
| 15:41.14 | blop | i'll past that, :) |
| 15:41.18 | [TK]D-Fender | greek_user: Green button? Red button? Are these some miracle controls I've somehow never heard of? |
| 15:41.45 | greek_user | i am refering to green and red as "Accept" and "Decline" a call |
| 15:42.13 | Strom_M | greek_user: pastebin a sip debug |
| 15:42.35 | [TK]D-Fender | greek_user: very poor description, and never ask any of this stuff without a pastebin of the full SIP debug to back it. |
| 15:43.09 | greek_user | i understand |
| 15:43.15 | AlienPenguin | [TK]D-Fender : i get a call from an extenal number, that is answered by a user in an extesion that matches _1X., this user uses attended trasfer to transfer to a phone: on the phone it appears _1X. instead of say: 102 (the user that picked up) |
| 15:43.22 | greek_user | where can i paste a debug file? |
| 15:43.26 | Strom_M | ~pb |
| 15:43.27 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 15:43.41 | greek_user | oh, thank you |
| 15:43.43 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 15:43.58 | [TK]D-Fender | AlienPenguin: did the user who picked up have a callerid set in their peer definition? |
| 15:44.04 | *** part/#asterisk frocos11292 (n=ask@firewall.vipvoz.com) |
| 15:45.13 | blop | oooooooooooooooooh, i just found where the problem was :P |
| 15:45.24 | blop | while i was copy/pasting to pastebin :) |
| 15:45.42 | blop | stupid ringtimeout :) |
| 15:45.48 | Strom_M | ...? |
| 15:46.23 | AlienPenguin | [TK]D-Fender : just checked, it did not, but i just added and tried, makes no difference |
| 15:46.30 | blop | i had ringtimeout=1000 , i just removed it, now its working again |
| 15:46.53 | blop | what's the default value for the ringtimeout btw? |
| 15:46.55 | deeperror | I get the following warnings pretty much all day and reproduce this when in a 3-way call and hangup. www.pastebin.ca/1043920 |
| 15:47.02 | [TK]D-Fender | AlienPenguin: perhaps you should pastebin your peers, and a complete call. This works fine for pretty much everyone else using *, so something must be off on your side... |
| 15:51.03 | AlienPenguin | [TK]D-Fender : i will soon, however, please note that i am not calling directly.. i mean the user who picks up the first call arrives there from a "s" or a pattern_matching extension dont know if that can make a difference |
| 15:51.38 | *** join/#asterisk rcy (n=rcy@S010600003981572c.vc.shawcable.net) |
| 15:52.17 | [TK]D-Fender | AlienPenguin: Shouldn't. What matters is the device. |
| 15:52.40 | *** join/#asterisk geoaxis (n=hatim@unaffiliated/geoaxis) |
| 15:53.38 | geoaxis | hello people, i am having issue with incoming call , i am unable to hear sound on the other end, i see IVRs getting played on asterisk console and i can also hear the IVRs if i call locally, i suspect its a NAT problem |
| 15:55.01 | geoaxis | will appreciate any help |
| 15:56.34 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 15:57.04 | blop | i still get that segfault tough :) |
| 15:57.14 | Strom_M | geoaxis: see the following |
| 15:57.16 | Strom_M | ~sipnat |
| 15:57.17 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 16:01.20 | *** join/#asterisk henrique (n=henrique@unaffiliated/henrique) |
| 16:01.53 | rootlogin | I have set up a HFC-S (hfcpci with misdn) card in TE mode on * 1.4 and kernel 2.6.24 .. it "works" but i get kernel oops/crashes after 1-2 calls .. i have no clue what makes sense to try next (different module versions maybe?) |
| 16:01.56 | *** part/#asterisk henrique (n=henrique@unaffiliated/henrique) |
| 16:02.11 | *** part/#asterisk greek_user (i=ath@adsl39-104.kln.forthnet.gr) |
| 16:02.34 | blop | will oslec echo canceller be merged into zaptel main sources one day? |
| 16:03.09 | rootlogin | or maybe somebody has a working *-misdn setup and can tell me about his versions |
| 16:03.39 | *** join/#asterisk quark_ (n=quark@pool-96-229-61-249.lsanca.fios.verizon.net) |
| 16:03.41 | *** join/#asterisk gaetronik (n=gaetan@164.77.74.18) |
| 16:03.51 | gaetronik | Hi all! |
| 16:04.14 | [TK]D-Fender | blop: Extremely unlikely. |
| 16:04.43 | quark_ | russellb: I guess I don't understand what asterisk does if it doesn't handle controling what get sent to what phone |
| 16:05.14 | russellb | hmm ... well .... it must do something |
| 16:05.28 | russellb | but if you want to send a message to the phone, you don't have to send it through asterisk |
| 16:05.35 | quark_ | are you at all familiar with sipsak? |
| 16:05.40 | russellb | no. |
| 16:05.41 | blop | D-Fender: why ? ;) license problem ? |
| 16:05.41 | enemy^x | when rxfax gets a voice call instead of a fax call and it get's hung up, then asterisk crashes (just exits) (asterisk v1.4.17) ... tested several spandsp`s. anyone seen this before? |
| 16:05.52 | quark_ | i'm pretty new to this voip stuff in general |
| 16:06.27 | quark_ | russellb: http://pastebin.com/m64e24ab0 |
| 16:07.07 | [TK]D-Fender | blop: Because to get included with *, the author would have to sign off rights to Digium which I can't imagine ever happening. IIRC some of their code was aquired through other OSS projects (SpanDSP, etc) and the CAN'T forward that onwards. |
| 16:08.16 | quark_ | russellb: the server obviously gets the message, I don't understand why it says that its dropping it |
| 16:08.28 | russellb | but you said you want to send it to the phone |
| 16:08.30 | russellb | so send it to the phone |
| 16:08.34 | blop | uhm ok :) |
| 16:08.44 | quark_ | russellb: I guess I don't know how to do that |
| 16:08.46 | russellb | have to run to lunch ... |
| 16:08.49 | quark_ | alright |
| 16:08.51 | quark_ | me too actually |
| 16:09.47 | quark_ | has anyone here ever used sipsak? |
| 16:09.50 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:10.45 | alrs | quark_ not for awhile, I usually use the sofia sip-options utility when I'm sip debugging |
| 16:11.17 | hsv-al | does anyone know off the top of their head, what the command on a pix 515 is, that squelches in between hops |
| 16:11.26 | hsv-al | between origin, and final destination |
| 16:11.41 | hsv-al | from a traceroute |
| 16:12.45 | *** join/#asterisk JenniferAkemi (n=akemi@69-196-131-228.dsl.teksavvy.com) |
| 16:15.37 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.193) |
| 16:16.46 | *** join/#asterisk Prog-javApplet (n=Prog-Jav@196.203.32.109) |
| 16:16.54 | gaetronik | quark_, i use tcpdump to capture and wireshark to analyze what is send and received |
| 16:17.04 | gaetronik | it's sufficient in most of the case |
| 16:18.58 | *** join/#asterisk lupino3 (n=andrea@217-133-45-108.b2b.tiscali.it) |
| 16:19.49 | lupino3 | hello everybody! when I start Asterisk, the members of my queues are marked as INVALID, I have to issue a "module reload app_queue.so" in order to see them marked as "Not in use"... How can I fix it? Thanks in advance |
| 16:21.44 | *** join/#asterisk nny_1 (n=Scott_My@64.203.239.83) |
| 16:22.28 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.140.193) |
| 16:23.53 | gaetronik | russellb, the mpodifications in the asterisk 1.6.1 for distributed device state will permit to have queues over various * servers or i'm totally wrong? |
| 16:24.43 | nny_1 | anyone have a preferred ITSP that has good customer service and works well with asterisk? |
| 16:24.44 | lmadsen | just the device state |
| 16:24.47 | lmadsen | not the queue itself |
| 16:25.09 | gaetronik | there is no way to achieve this? |
| 16:25.34 | lmadsen | you can have a Queue() on a single box and distribute the calls to multiple servers, but the Queue() app can only run on 1 server |
| 16:25.38 | Strom_M | nny_1: i like teliax if you're in north america |
| 16:26.05 | nny_1 | Strom_M: i'll check them out thanks. Having some strangeness with my current one, trying to figure out if it's us or them |
| 16:26.22 | Strom_M | nny_1: what kind of strangeness? |
| 16:26.50 | lupino3 | I'm using Asterisk 1.4.19.. please help :'( |
| 16:26.58 | nny_1 | DTMF is being buggy (they suggested inband, which hasn't solved the issue. I also get a lot of registration timeout (they have stated this is an asterisk bug), and getting some call failures and delays |
| 16:27.19 | nny_1 | it basically *just worked* up until a week ago or so |
| 16:28.24 | nny_1 | chan_sip.c: -- Registration for 'wildcatsteel@inbound7.companyXX.net' timed out, trying again |
| 16:28.34 | Defraz | who are you using nny_1 |
| 16:28.47 | nny_1 | Vitelity atm |
| 16:28.51 | nny_1 | I don |
| 16:29.25 | nny_1 | 't want to have to switch, but getting ready to have company do 20K minutes a month in outbound long distance, and don't want them to think the system is borked |
| 16:30.15 | nny_1 | still may be an issue with my ISP's network or ours, so I am going to exhaust that route first |
| 16:30.37 | *** join/#asterisk Prog-JavApplet (n=Prog-Jav@41.224.253.239) |
| 16:30.42 | Prog-JavApplet | hey |
| 16:30.59 | Prog-JavApplet | please i have problem with ./configure astrisk |
| 16:31.01 | Prog-JavApplet | configure: error: *** termcap support not found |
| 16:31.11 | nny_1 | the DTMF issue I am unsure on. I think their last suggestion was to have all devices use inband DTMF, but I don |
| 16:32.34 | Prog-JavApplet | can someone help please? |
| 16:32.43 | NovceGuru | nny_1: I'm getting ready to try vitelity |
| 16:33.01 | NovceGuru | was going to try teliax but they can't port my # |
| 16:33.08 | nny_1 | NovceGuru: I should know by the end of the day if this issue is local to my network or not |
| 16:33.11 | nny_1 | the main is DTMF right now |
| 16:33.12 | Defraz | I personally love Vitelity, they have a great UI. |
| 16:33.46 | NovceGuru | Their pricing model works for me because I don't think it'll reach the teliax unlimited price/month |
| 16:34.01 | nny_1 | Yeah their interface is nice indeed |
| 16:34.07 | NovceGuru | nny_1: I've had a few issues with some other ITSP's lately, so i'm shopping around |
| 16:34.17 | NovceGuru | but like you, i'm not sure its the network |
| 16:34.28 | NovceGuru | running a broadvoice account on line2 of this phone to see what it does |
| 16:34.29 | [TK]D-Fender | Prog-JavApplet: "yum install libtermcap" |
| 16:34.35 | nny_1 | I am gonna exhaust my troubleshooting skills today, ill keep you all informed |
| 16:36.34 | Prog-JavApplet | [TK]D-Fender: libtermcap is already intalled :( |
| 16:37.55 | [TK]D-Fender | Prog-JavApplet: "yum install libtermcap-devel" |
| 16:37.56 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 16:38.11 | *** join/#asterisk akira2014 (n=chatzill@19.Red-88-1-135.dynamicIP.rima-tde.net) |
| 16:38.28 | [TK]D-Fender | Prog-JavApplet: While you're at it : "yum -y install gcc gcc-c++ kernel-devel bison openssl-devel libtermcap-devel ncurses-devel doxygen curl-devel" |
| 16:39.17 | spokra | NovceGuru: I tried broadvoice and dumped them during the trial.. they are a reseller and have no clue someone here suggested les.net and I'm very happy with them.. |
| 16:39.37 | hsv-al | does * store incoming phone call #'s in a log? |
| 16:39.41 | x86 | spokra: I suggest les.net to everyone... they're awesome |
| 16:39.43 | hsv-al | im remoted into home, and need my home # since i forget it |
| 16:39.51 | nick125 | hsv-al: Check the CDR |
| 16:39.53 | x86 | spokra: support is the best i've ever dealt with |
| 16:39.59 | Qwell | hsv-al: you forgot your home #? |
| 16:40.04 | hsv-al | yeah, i just got it installed heh |
| 16:40.12 | hsv-al | primary = cell |
| 16:40.13 | Qwell | oh |
| 16:40.19 | x86 | didn't document it? :) |
| 16:40.26 | nick125 | has too many phone numbers to count |
| 16:40.26 | spokra | x86: even before your a customer they rock. |
| 16:40.38 | x86 | I usually document circuit IDs, POTS numbers, etc in my zaptel configs |
| 16:41.03 | nny_1 | les.net does seem pretty good |
| 16:41.15 | x86 | they rock |
| 16:41.27 | x86 | I used them for 3 years |
| 16:41.28 | nny_1 | |m| (o_0) |m| |
| 16:41.36 | x86 | and I was an ITSP |
| 16:41.43 | nny_1 | nice |
| 16:41.58 | nny_1 | they can even say les.net = x86 compatible! |
| 16:42.05 | nny_1 | ok bad joke, back to diagnostics |
| 16:42.20 | lmadsen | shakes his head |
| 16:42.36 | x86 | heh |
| 16:42.56 | x86 | they claim price per minute is $0.015, but I was getting $0.008 |
| 16:43.02 | jeev | fender, my rma'd WIP330 comes in thoday, lets see how bad it is! |
| 16:43.07 | x86 | not sure if they raised the prices or what |
| 16:43.20 | outtolunc | dips his hightops in bright orange paint and calls it sneakervoice |
| 16:43.29 | hsv-al | Master.csv isnt that large |
| 16:43.35 | hsv-al | within /var/log/asterisk/cdr-csv |
| 16:43.42 | hsv-al | doesnt have phone #'s |
| 16:43.42 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 16:43.46 | *** join/#asterisk knobo` (n=user@40.80-202-89.nextgentel.com) |
| 16:44.02 | nny_1 | the suggestion from my ITSP for my registration errors is to "switch to IP routing". and "Asterisk is known to spit out false registration errors". Anyone wanna chime in on that statement? |
| 16:45.05 | nny_1 | that's gonna be my first nut to crack. The no route to host stuff may or may not be resultant of that error |
| 16:46.21 | nny_1 | although "ip routing" seems ambiguous. |
| 16:46.32 | x86 | http://les.net/products/product_ipwholesale.php |
| 16:46.45 | x86 | that's what I was doing to get $0.008/min from les.net |
| 16:46.56 | nny_1 | damn .008 |
| 16:47.04 | nny_1 | heh 31337 |
| 16:47.05 | nick125 | nny_1: When they say "switch to IP routing", I think they mean statically route to your IP instead of registering. |
| 16:47.34 | nny_1 | is this normal for using an ITSP? |
| 16:47.39 | jeev | do all providers calculate based on 6 seconds? or only mine |
| 16:48.05 | nick125 | jeev: Most of them bill to 6 second increments. |
| 16:48.10 | jeev | ahh ok cool |
| 16:48.18 | nick125 | Least for domestic calling |
| 16:48.40 | nny_1 | hrrm les.net has .015 to US |
| 16:49.47 | nny_1 | nick125: know of any asterisk specific documentation for that? |
| 16:50.14 | nick125 | nny_1: It's usually something that your ITSP has to do, I believe. |
| 16:50.34 | knobo` | What are important protocols for voip in additional to: h.323 sip iax enum mgcp rtp stun ss7? |
| 16:50.37 | nny_1 | so basically they mean allow connections based on my IP? |
| 16:50.50 | knobo` | well "important" is relative... |
| 16:50.52 | _ShrikE | lots of itsp's will not allow you to register |
| 16:51.01 | nick125 | nny_1: Basically, they'll route calls to your IP, just like if you were registering, except you cut out the registration part. |
| 16:51.09 | nny_1 | nick125: gotcha |
| 16:52.11 | nny_1 | nick125: seems a little more reliable, although it would require static ips for the customers I have that use ITSP for LD, which is trivial |
| 16:52.13 | nny_1 | i'll make that my first point today to address then |
| 16:52.48 | nny_1 | hmm here's a though, what if i have two * boxes behind the same IP address? This very well may be breaking my network here |
| 16:52.51 | nny_1 | thought* |
| 16:53.05 | nick125 | nny_1: If you run them on different ports, you should be fine. |
| 16:53.27 | nny_1 | nick125: just the registration port or rtp and sip registration? |
| 16:53.50 | nick125 | nny_1: You probably want to change the SIP port and the RTP ranges, just to make sure they won't conflict. |
| 16:53.56 | nny_1 | nick125: gotcha |
| 16:55.58 | nny_1 | nick125: (thanks btw), what about NAT as far as the registrant? I have one box that sits on the WAN (it's the gateway) and the other (for testing) has NAT (different subnet) behind the first box. I imagine this is bad, although it is a linux based gateway, so the NAT may be one of the more forgiving types |
| 16:56.56 | nny_1 | this is where i think my issues lie. I may need to grab another IP for testing, although a portion of my smaller clients would benefit more from being able to just have their pbx sit behind their current edge device |
| 16:57.08 | nick125 | It should work. You have to make sure that nat=yes, though. |
| 16:57.28 | nny_1 | nick125: yes I have that in both sip entries for the ITSP |
| 16:57.55 | nick125 | Might want to do some port forwarding while you're at it. |
| 16:58.08 | nny_1 | nick125: never hurts :) |
| 16:58.28 | nny_1 | I think i know where my error is though |
| 16:58.57 | nny_1 | so both boxes use the standard sip registration ports, maybe this is where i have made my mistake |
| 17:01.00 | hsv-al | by default should Master.csv |
| 17:01.11 | hsv-al | within /var/log/asterisk/cdr-csv contain inbound #'s in the viewable logs? |
| 17:03.25 | *** part/#asterisk a1fa (n=a1fa@unaffiliated/a1fa) |
| 17:06.22 | *** join/#asterisk los415 (i=los415@LAX-DHCP-64-201-109-237.race.com) |
| 17:06.50 | Prog-JavApplet | how can i add urser to the sip server? |
| 17:07.49 | *** join/#asterisk l0verb0y (n=l0verb0y@119.111.96.120) |
| 17:08.34 | *** join/#asterisk tobias (n=tobias@cpe-069-134-205-184.nc.res.rr.com) |
| 17:09.53 | [TK]D-Fender | Prog-JavApplet: ..... |
| 17:09.56 | [TK]D-Fender | ~book |
| 17:09.57 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 17:09.58 | [TK]D-Fender | ^^^^^^^^^^^^^ |
| 17:13.28 | akira2014 | hello |
| 17:16.24 | *** join/#asterisk Micko113 (n=micko113@c-67-183-170-166.hsd1.wa.comcast.net) |
| 17:16.28 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 17:16.54 | jameswf-home | Cough |
| 17:17.05 | *** join/#asterisk Micko113 (n=micko113@c-67-183-170-166.hsd1.wa.comcast.net) |
| 17:17.09 | akira2014 | i'm in troubles with atxfer in asterisk 1.4.18 |
| 17:17.12 | *** join/#asterisk nny_2 (n=Scott_My@64.203.239.83) |
| 17:17.38 | akira2014 | i've defined the #2 key sequensa as atxfer.... |
| 17:18.14 | akira2014 | but atsfer is working like blindtransfer |
| 17:18.19 | akira2014 | any ideas ? |
| 17:18.39 | [TK]D-Fender | akira2014: maybe you're dialing too slow. |
| 17:18.54 | [TK]D-Fender | akira2014: Did you set the dynamic features var BEFORE you dialed? |
| 17:19.18 | akira2014 | [TK]D-Fender: no..... |
| 17:19.38 | akira2014 | i haven't found anything about this in manual |
| 17:19.39 | [TK]D-Fender | akira2014: Go read the WIKI page on features.conf and those dialing features then. |
| 17:20.48 | jameswf-home | Manuel is my gardener |
| 17:20.57 | akira2014 | thk's a lot [TK]D-Fender |
| 17:21.24 | anonymouz666 | akira2014: is it an applicatiomap? |
| 17:21.52 | akira2014 | no it's not an app map |
| 17:22.00 | anonymouz666 | or just atxfer => #2 featuremap? |
| 17:22.27 | anonymouz666 | you don't need to set any var unlesse it's applicationmap |
| 17:22.46 | *** join/#asterisk enemy^x (n=enemy@c213-158-248-202.static.sdsl.no) |
| 17:23.25 | *** join/#asterisk B1ST (i=overdose@d51A411B6.access.telenet.be) |
| 17:23.29 | anonymouz666 | what 'feature show' says? |
| 17:23.33 | akira2014 | it's just atxfer => #2 |
| 17:23.56 | enemy^x | anyone have a good way to debug rxfax in a good fashion? it kills my asterisk process if I call rxfax by voice. |
| 17:24.03 | akira2014 | attended transfer #2 |
| 17:24.12 | [TK]D-Fender | akira2014: pastebin your complete call at verbose 10, and your features.conf |
| 17:24.13 | [TK]D-Fender | ~pb |
| 17:24.14 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 17:24.14 | akira2014 | o my god |
| 17:24.24 | akira2014 | i've see my n00berror |
| 17:24.31 | akira2014 | blind transfer is set to # |
| 17:24.39 | akira2014 | and assisted is set to #2 |
| 17:27.25 | akira2014 | thk's to everybody for your time |
| 17:31.05 | *** join/#asterisk deeperror (n=deeperro@99.149.251.91) |
| 17:31.17 | B1ST | i got a problem whit registry, hope someone can help me whit that |
| 17:31.27 | jameswf-home | ~ask |
| 17:31.28 | jbot | i guess ask is 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. |
| 17:31.28 | B1ST | [Jun 10 19:31:20] NOTICE[3931]: chan_sip.c:7511 sip_reg_timeout: -- Registration for 'ZHY3J169DIM2SR0LW5X@voipsolutions.be' timed out, trying again (Attempt #11) |
| 17:31.59 | B1ST | well, want to know what i did wrong in my sip.conf file |
| 17:32.48 | B1ST | register => ZHY3J169DIM2SR0LW5X:passwd@voipsolutions.be/kevin is the line i added in sip.conf |
| 17:33.03 | sbingner | jbot_, are you there? |
| 17:33.42 | enemy^x | anyone here run rxfax? and had some issues with it? |
| 17:36.58 | nny_2 | for those involved with my ITSP questions earlier. Thanks. It seems that IP routing is muy bueno and suggested (this I did not know) and double NAT, even for testing, is bad... bad...! |
| 17:37.20 | deeperror | B1ST, looks like you didn't type in the password ;) |
| 17:37.38 | hsv-al | in exten => t,1,Playback(vm-goodbye) |
| 17:37.50 | hsv-al | is that a prebuilt variable that stands for time out? |
| 17:38.12 | deeperror | hsv-al, extension |
| 17:38.32 | deeperror | hsv-al, along with i, T, and h |
| 17:38.39 | hsv-al | calls will be sent to the t extension if the caller takes to long to press a digit after WaitExten() |
| 17:38.44 | [TK]D-Fender | B1ST: if it keeps retrying thats likely because you're not getting an error. This is either a firewall issue, bad host, issue, or NAT issue. |
| 17:39.04 | Strom_M | hsv-al: that's only IF auto-fallthrough isn't on |
| 17:39.05 | B1ST | i dont use issue |
| 17:39.08 | B1ST | err |
| 17:39.10 | B1ST | nat |
| 17:39.22 | Strom_M | hsv-al: otherwise, it'll fall directly through to t without waiting |
| 17:39.32 | hsv-al | im looking at page 130 of the book |
| 17:39.36 | hsv-al | and trying to understand the logic in it |
| 17:39.39 | B1ST | [TK]D-Fender: i really need to use nat? |
| 17:39.47 | [TK]D-Fender | sbingner: Don't talk to the bot.... its too early for that kind of humour. |
| 17:40.01 | Strom_M | hsv-al: i'd advise you not use t or T and instead have the goodbye and hangup occur in the same extension as calling voicemail |
| 17:40.08 | [TK]D-Fender | B1ST: I never said to USE it, I'm saying your problem could be symptomatic of it. |
| 17:40.09 | hsv-al | if anything other then 1, 2, or 3 is pressed, it says invalid # |
| 17:40.28 | [TK]D-Fender | B1ST: pastebin SIP debug of the registration attempt. |
| 17:40.30 | [TK]D-Fender | ~pb |
| 17:40.31 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 17:40.33 | [TK]D-Fender | ^^^^^^^^^ |
| 17:40.36 | B1ST | ok |
| 17:40.43 | Strom_M | hsv-al: that's the "i" extension -- "invalid" |
| 17:41.45 | hsv-al | so if I press a 4-9, instead of 1-3 |
| 17:41.55 | hsv-al | it says invalid, then goes back to the top and waits for initial input again |
| 17:42.05 | hsv-al | and if I dont put in input, it will say good bye |
| 17:42.06 | Strom_M | hsv-al: look at your "i" extension |
| 17:42.17 | Strom_M | hsv-al: mostly, yes |
| 17:42.19 | hsv-al | what is telling it how long to wait |
| 17:42.27 | hsv-al | before it says good bye, ie: miliseconds |
| 17:42.30 | hsv-al | or seconds, etc |
| 17:42.31 | Strom_M | do you have auto-fallthrough enabled? |
| 17:42.33 | hsv-al | yes |
| 17:42.40 | Strom_M | the duration of WaitExten() |
| 17:42.49 | B1ST | [TK]D-Fender: http://www.pastebin.ca/1044018 |
| 17:42.56 | Strom_M | but I would avoid doing the goodbye in the T or t extension |
| 17:43.06 | Strom_M | do the goodbye immediately after the WaitExten() |
| 17:43.10 | Strom_M | or program in some looping logic |
| 17:43.31 | hsv-al | default WaitExten() = 10s |
| 17:43.33 | hsv-al | see it |
| 17:43.43 | Strom_M | see what? |
| 17:43.54 | hsv-al | i just saw it a few pages bakc, default timeout in waitexten() = 10s |
| 17:43.55 | Strom_M | i always specify an argument to that just for clarity's sake |
| 17:44.34 | hsv-al | ~book |
| 17:44.34 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 17:44.37 | sevard | I always specify catsex to that just for dogballs's sake. |
| 17:44.51 | Strom_M | sevard: <3 |
| 17:44.57 | Strom_M | hsv-al: yes, I know the book |
| 17:45.05 | hsv-al | no for me, im looking something up |
| 17:45.15 | [TK]D-Fender | B1ST: check your firewall. |
| 17:45.17 | Strom_M | you couldnt message the bot privately? :/ |
| 17:45.45 | B1ST | i don't have one |
| 17:46.05 | [TK]D-Fender | B1ST: pastebin the backup. |
| 17:46.38 | B1ST | hm? |
| 17:46.43 | B1ST | backup from what? |
| 17:47.59 | [TK]D-Fender | B1ST: that your firewall is clear. |
| 17:48.29 | hsv-al | strom: http://pastebin.com/m272544bc |
| 17:49.17 | hsv-al | am i correct to say when i read that basic plan, anything other then 1,2, or 3 pressed after please enter the extension is played....it says: sorry invalid #, then loops to the top of the context |
| 17:49.22 | *** part/#asterisk knobo` (n=user@40.80-202-89.nextgentel.com) |
| 17:49.43 | Strom_M | hsv-al: it doesn't "loop" so much as it goes back to the extension |
| 17:50.07 | Strom_M | but that's a mediocre IVR design |
| 17:50.10 | B1ST | heh, [TK]D-Fender it's a fresh install from yesterday, i did not set iptables and such |
| 17:50.41 | hsv-al | so goes back to exten => 123,n,Background(enter-ext-of-person) |
| 17:50.54 | hsv-al | if [4-9] is pressed? |
| 17:51.12 | Strom_M | no |
| 17:51.14 | Strom_M | look at the goto |
| 17:51.17 | [TK]D-Fender | B1ST: verify it. |
| 17:51.22 | Strom_M | ~brain |
| 17:51.23 | jbot | brain is, like, a very useful device. It is advised that you ensure your brain is switched to the "on" position before continuing. If you need assistance, please consult your directory. This is a recording. |
| 17:51.27 | hsv-al | why would it answer() again |
| 17:51.32 | hsv-al | if the phone is already picked up |
| 17:51.35 | Strom_M | look at the goto |
| 17:51.43 | Strom_M | that's what it's being told to do |
| 17:51.52 | keith4 | Strom_M: say it again, I don't thing it's clear yet ;-) |
| 17:51.58 | keith4 | s/thing/think |
| 17:52.00 | Strom_M | answer() doesn't do anything if the channel is already answered |
| 17:52.05 | hsv-al | ok i didnt know that |
| 17:53.02 | hsv-al | so in that snippit, what causes the t extens to be triggered |
| 17:53.08 | hsv-al | since I understand the i's now |
| 17:53.31 | Strom_M | the "t" happens when the WaitExten() times out |
| 17:53.38 | *** join/#asterisk rabbs (n=qwer@4801F05A.ptr.provps.com) |
| 17:53.44 | Strom_M | but like I said, that's a really pathetically dumb way to design an IVR |
| 17:53.58 | hsv-al | well thats the intro newbie teaching method |
| 17:54.01 | hsv-al | in the book |
| 17:54.27 | Strom_M | if the book isn't teaching you what everything does, perhaps it's not written very well ;) |
| 17:55.19 | hsv-al | it says calls will be sent to the t extension of waitexten() duration is achieved without input |
| 17:55.37 | hsv-al | but thats assumed to be understood, by how t extensions are interpreted in dialplans? |
| 17:55.44 | Strom_M | t == timeout |
| 17:55.44 | quark_ | is anyone familiar with the sofia sip stack from nokia? |
| 17:55.48 | Strom_M | it doesn't say that? |
| 17:55.57 | hsv-al | so you cant alter what t references to |
| 17:56.06 | hsv-al | in custom dialplans, if someone wants to use t for something else? its locked down? |
| 17:56.07 | Strom_M | no |
| 17:56.15 | Strom_M | it's a reserved special extension |
| 17:57.59 | hsv-al | I have still in my beginner dialplan from early on exten => s,1,Verbose(1|Unrouted call handler) |
| 17:58.03 | hsv-al | safe to rm -rf that? |
| 17:58.46 | *** join/#asterisk keulin (n=cray@bne75-5-82-231-224-34.fbx.proxad.net) |
| 17:58.46 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 17:58.52 | Strom_M | don't be a hopeless unix geek...don't use "rm -rf" as a synonym for "delete" |
| 17:59.19 | Strom_M | and it's probably safe, but i can't just tell you these things in a vacuum |
| 18:04.07 | *** join/#asterisk ManxPower (n=manxpowe@163.sub-70-220-190.myvzw.com) |
| 18:04.32 | nick125 | Or at least use sane flags...you don't need -r. |
| 18:05.12 | ManxPower | The "r" option to Dial is EVIL AND SHOULD BE DESTROYED. |
| 18:05.18 | ManxPower | But I don't have strong feelings on the matter |
| 18:05.20 | ManxPower | 8-) |
| 18:05.59 | Strom_M | ManxPower: he was talking about the "rm" command |
| 18:06.11 | Strom_M | but yes, i share your sentiment |
| 18:06.57 | ManxPower | Ah |
| 18:07.18 | ManxPower | never miss a chance to diss that option. |
| 18:07.38 | Strom_M | DON'T MISS THE DISS |
| 18:07.51 | Strom_M | this message brought to you by ManxPower and the Ad Council |
| 18:08.16 | ManxPower | And the Illuminati |
| 18:08.41 | hsv-al | strom, in that example, for incoming calls, I can replace that |
| 18:08.45 | hsv-al | 123, with s? |
| 18:08.59 | hsv-al | so general users could test that basic menu |
| 18:09.20 | hsv-al | exten 123 does not exist, so i just want it applied to an incoming phone call |
| 18:09.21 | Strom_M | hsv-al: that depends |
| 18:09.27 | Strom_M | how are the calls coming in? |
| 18:09.33 | hsv-al | pots analog |
| 18:09.41 | Strom_M | then yes |
| 18:09.48 | ManxPower | hsv-al: "s" means "did not receive a dialed extension", which is almost only for FXO ports. |
| 18:09.58 | ManxPower | You will NEVER EVER EVER received the dialed number on a FXO port. |
| 18:10.01 | Strom_M | assuming that's the context which the FXO port is set to in zapata.conf |
| 18:10.03 | ManxPower | Therefore you can ONLY use "s" |
| 18:10.29 | [TK]D-Fender | ManxPower: No not "not ever". just extremely rarely. |
| 18:10.35 | hsv-al | ill paste zapata.conf, one sec |
| 18:10.44 | Strom_M | hsv-al: NO |
| 18:10.45 | Strom_M | don't paste |
| 18:10.46 | ManxPower | "s" means "port is too stupid to get the dialed number." |
| 18:10.48 | Strom_M | just think and look |
| 18:10.49 | hsv-al | pastebin. |
| 18:10.50 | hsv-al | durr |
| 18:10.55 | Strom_M | no, not pastebin either |
| 18:10.58 | Strom_M | think and look |
| 18:11.02 | Strom_M | use your head please |
| 18:11.03 | ManxPower | [TK]D-Fender: trust me, hsv-al doesn't need any more confusion. |
| 18:12.17 | hsv-al | set to [incoming] |
| 18:12.20 | hsv-al | non exten = s, so s it is |
| 18:12.30 | [TK]D-Fender | hsv-al: Analog zap = almost always "s" unless you paided for a DID Analog service |
| 18:12.35 | ManxPower | hsv-al: please speak clearly |
| 18:12.37 | Strom_M | paided? |
| 18:12.50 | ManxPower | [TK]D-Fender: FXO signalled channels on a Channalised T-1 also |
| 18:12.56 | Strom_M | je m'excuse, [TK]D-Fender...paided ain't a word in English ;) |
| 18:12.59 | keith4 | Strom_M: yah, past tense of "paid" |
| 18:13.06 | Strom_M | paid IS past tense already |
| 18:13.16 | [TK]D-Fender | I'm pre-occupied, couldn't care less for typos ATM. |
| 18:13.18 | keith4 | yah, but paided is the PAST past tense |
| 18:13.28 | keith4 | it's like.... paid a *really* long time ago |
| 18:13.33 | Strom_M | bludgeons keith4 with a telephone |
| 18:13.34 | ManxPower | Exactly! Pay attention, Strom_M! |
| 18:13.59 | keith4 | Strom_M: I hope it's an ancient bakelite one, or it's just not going to hold up to that abuse |
| 18:14.14 | Strom_M | keith4: it's a WECo 1A1 coin telephone |
| 18:14.20 | keith4 | ah, that'll do it |
| 18:14.29 | ManxPower | you can kill people with that thing! |
| 18:14.34 | Strom_M | indeed |
| 18:14.41 | Strom_M | that's what makes it such an effective bludgeoning phone |
| 18:14.41 | hsv-al | strom, somehow weaseals was in the [default] context in extensions.conf |
| 18:14.52 | hsv-al | even though incoming calls were pointing to [incoming] |
| 18:15.02 | hsv-al | and yet it played somehow |
| 18:15.34 | *** join/#asterisk cmantito (n=gphreak@pool-71-188-82-138.cmdnnj.east.verizon.net) |
| 18:15.37 | Strom_M | hsv-al: if there was no "s" extension in incoming, it likely fell back to default |
| 18:15.44 | Strom_M | ~brain |
| 18:15.45 | jbot | rumour has it, brain is a very useful device. It is advised that you ensure your brain is switched to the "on" position before continuing. If you need assistance, please consult your directory. This is a recording. |
| 18:16.25 | hsv-al | i dont think your attitude will change until something bad happens to you |
| 18:16.35 | hsv-al | you need to calm down to those that havent been exposed to this point in the book yet |
| 18:17.07 | Strom_M | hsv-al: I'm encouraging you to think logically and not just go "I don't know what this is so therefore I'm going to ask someone to tell me" |
| 18:17.33 | hsv-al | .... whateva bigboi |
| 18:18.07 | nick125 | Strom_M: I've heard that brains are going out of style. |
| 18:18.10 | hsv-al | well it works now for incoming calls :) |
| 18:18.54 | hsv-al | [Jun 10 12:18:23] WARNING[9137]: pbx.c:2483 __ast_pbx_run: Channel 'Zap/1-1' sent into invalid extension 's' in context 'default', but no invalid handler |
| 18:18.54 | hsv-al | <PROTECTED> |
| 18:19.01 | ManxPower | hsv-al: you are so confused at this point that you need to stop blindly following stuff and learn WHY you need to do stuff. We are not a DO-IT-FOR-YOU, we are a TEACH-YOU-HOW group of people |
| 18:19.16 | ManxPower | hsv-al: That message is pretty clear. |
| 18:19.49 | hsv-al | [default] is blank |
| 18:19.59 | hsv-al | all my stuff is in [incoming] |
| 18:20.10 | ManxPower | Well it's not landing in incoming, it's landing in default |
| 18:20.10 | hsv-al | and fxo calls are pointed to [incoming] |
| 18:20.17 | hsv-al | in zapata.conf |
| 18:20.19 | ManxPower | hsv-al: not according the THAT message they are not. |
| 18:20.26 | ManxPower | is the context= line before of after the channel=1 |
| 18:20.29 | Strom_M | hsv-al: did you reload zapata.conf? |
| 18:20.34 | Strom_M | did you reload extensions.conf? |
| 18:20.38 | hsv-al | dialplan reload |
| 18:20.39 | Strom_M | did you do a sanity check? |
| 18:20.57 | ManxPower | hsv-al: dialplan reload reloads the dialplan, it does not reload zapata.conf |
| 18:21.19 | keith4 | just retart asterisk |
| 18:21.27 | keith4 | it'll be easier than thinking, for you |
| 18:21.39 | ManxPower | hsv-al: HINT: all /etc/zapata.conf options are applied to all FOLLOWING channel = lines. |
| 18:21.59 | hsv-al | n/m it works, [incoming_calls] was my context in extensions.conf |
| 18:22.03 | hsv-al | zapata.conf pointed to [incoming] |
| 18:22.08 | hsv-al | coffee ftw, brb :) -grin- |
| 18:22.08 | ManxPower | hsv-al: be nore careful |
| 18:22.22 | ManxPower | How much of our life was wasted"? |
| 18:22.41 | Strom_M | ManxPower: clearly it's an attitude problem on our end that's causing all the problems |
| 18:22.51 | *** join/#asterisk dalbaech (n=dalbaech@c-98-200-231-201.hsd1.tx.comcast.net) |
| 18:23.15 | hsv-al | everything else was fine, but i had an invalid named context |
| 18:23.24 | dalbaech | hey everyone |
| 18:23.26 | *** join/#asterisk De_Mon (i=de_mon@fl-71-55-24-227.dhcp.embarqhsd.net) |
| 18:24.50 | keith4 | Strom_M: yes. you should seek IRC anger counseling, clearly |
| 18:25.10 | Strom_M | dialing 10-10-heroin overdose |
| 18:25.53 | hsv-al | heh, weird, when I typed an invalid extension it did a forever loop of the lady saying: I am sorry, that is an invalid extension please try again |
| 18:27.27 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-46-104.lns10.syd7.internode.on.net) |
| 18:27.53 | dalbaech | has anyone here had much experience sending outbound calls to an openser box? |
| 18:36.44 | NovceGuru | stabs 1 way audio |
| 18:39.59 | NovceGuru | asterisk server is on a public IP, client is behind a 1:1 nat with it's own public IP (but device has private ip) has audio when calling the extension from outside but when extension makes a call = call completes but no audio |
| 18:40.34 | NovceGuru | shoot. I fail |
| 18:41.13 | NovceGuru | call from the outside = audio both ways, when the client calls out = only audio from the called party |
| 18:42.38 | NovceGuru | nat=no fixed it |
| 18:51.58 | *** join/#asterisk ZaVoid (n=zavoid@75.147.121.177) |
| 18:54.40 | nny_2 | anyway to change it from "asterisk" if CID is unavailable? |
| 18:55.28 | [TK]D-Fender | nny_2: "core show application set" , "core show function CALLERID" |
| 18:56.46 | nny_2 | [TK]D-Fender: hmm yeah i know that one, just noticed if all else fails it says asterisk (ex: my cellphone has CID blocked),... no big deal |
| 18:57.10 | nny_2 | i am betting that still is the answer lol |
| 18:57.19 | nny_2 | ill shall do what you say :) |
| 18:58.26 | nny_2 | starting to hate this sidecar a little |
| 18:58.48 | ZaVoid | is there an easy way to strip "#" in the dialplan if sometimes it comes in and sometimes doesn.t.. sometimes i'll see 5551212# and sometimes just 5551212 |
| 18:59.02 | nny_2 | noticed if asterisk is restarted, i have to restart the sidecar.. going to see if i can increase the re registration time for it or something to that effect.. |
| 19:03.42 | *** join/#asterisk blq (i=Bl@dslb-088-067-021-184.pools.arcor-ip.net) |
| 19:11.41 | enemy^x | anyone here want to setup asterisk for me? msg me and I'll pay you! (You need to have skills & paypal) |
| 19:12.29 | *** join/#asterisk kamh (n=q@host-81-190-236-85.wroclaw.mm.pl) |
| 19:12.42 | *** join/#asterisk bkruse (n=bkruse@phoenixii.total-vision.net) |
| 19:12.42 | *** mode/#asterisk [+o bkruse] by ChanServ |
| 19:15.43 | nny_2 | any linksys 932 users know how to have the sidecar stay registered through an asterisk restart |
| 19:15.46 | [TK]D-Fender | ZaVoid: "core show function CUT" , "cor show application set". |
| 19:15.59 | [TK]D-Fender | ZaVoid: And basic variable expressions. |
| 19:16.00 | *** part/#asterisk PTorres (n=PTorres@200.68.87.146) |
| 19:17.08 | ManxPower | ZaVoid: just play a message saying not to use # |
| 19:17.23 | ZaVoid | lol |
| 19:17.35 | ManxPower | I'm not joking. |
| 19:17.48 | ManxPower | Users find it easier to remember ONE way to dial instead of TWO ways to dial |
| 19:18.05 | ManxPower | I am assuming you don't have overlapping dialplan entries that would require a timeout or a # |
| 19:18.20 | [TK]D-Fender | ManxPower: They do... each of them remembers THEIR way, it jsut happens to DIFFER from that of their compatriots :) |
| 19:18.29 | ManxPower | make the use the same way |
| 19:19.07 | ManxPower | Oh, they'll whine and cry for a while, they always do when anything changes -- but it's for their own good, like jail time for drunk drivers. |
| 19:21.12 | ZaVoid | i got one carrier that chokes on it |
| 19:21.14 | ZaVoid | rest don't care |
| 19:21.30 | ZaVoid | i support jail time for drunk drivers.. unless they are cute |
| 19:22.16 | *** join/#asterisk talntid (n=erict@66.208.251.170) |
| 19:22.18 | Strom_M | i wouldn't recommend punishing users for dialing # to complete a number, since # is the "I'm finished dialing so put the call through now" key |
| 19:22.35 | ZaVoid | yeah well tehir ata's aren't conrfigured right and sending the damn # |
| 19:22.37 | Strom_M | let me see if there's a way to strip it |
| 19:23.00 | [TK]D-Fender | ZaVoid: And whose job is it to configure those ATAs was it? |
| 19:23.15 | ZaVoid | the customer owns it |
| 19:23.27 | [TK]D-Fender | ZaVoid: And their job to configure? |
| 19:23.28 | ZaVoid | they stupidly play with it all the time |
| 19:23.46 | ZaVoid | in fact they tell me that changing sip to port say 100 frm 5060 improves quality :) |
| 19:24.00 | ZaVoid | so their obviously qualified to be touching their own device |
| 19:24.24 | jameswf-home | doesnt toech his own device |
| 19:24.26 | jameswf-home | often |
| 19:24.46 | ManxPower | ZaVoid: In theory I can see how that would be, but that's only if there is poorly configured QoS somewhere that's based on source port numbers rather than destinations numbers |
| 19:25.12 | ZaVoid | right that and the fact media doesn't go over 5060...... |
| 19:25.25 | *** part/#asterisk WildPikachu (n=WildPika@about/linux/staff/wildpikachu) |
| 19:25.25 | ManxPower | *nod* |
| 19:25.35 | ManxPower | There IS that, I guess 8-) |
| 19:25.41 | ZaVoid | minor details i call it i know |
| 19:25.47 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 19:26.20 | Strom_M | ZaVoid: i don't think there's a way to specify "arbitrary number of digits followed by #" but you could do something like "_NXXNXXXXXX#" and then just use variable substrings to Goto() the right extension |
| 19:26.21 | ManxPower | ZaVoid: Get a known working configuration on their ATA, export/save/backup the config. Any time they complain make them restore the settings from the saved file. |
| 19:26.40 | ManxPower | If that fixes it you can tell them to go screw themselves and stop bothering you with problems they create. |
| 19:26.48 | ZaVoid | we do now that now manx |
| 19:26.52 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 19:26.53 | ManxPower | Cool! |
| 19:26.54 | ZaVoid | i got 20k devices though... |
| 19:26.58 | ZaVoid | just easier to cut the # off i think |
| 19:27.10 | ManxPower | start charging them if restoring from backup fixes the issue |
| 19:27.18 | ZaVoid | yeah strom thats the problem its an arbritrary number of digits first |
| 19:27.20 | sbingner | [TK]D-Fender, pfft never too early to be retarded ;) |
| 19:27.32 | ZaVoid | nah i got provisioning server they can use get the correct fix |
| 19:27.36 | ManxPower | ZaVoid: How exatly is there variable number of digits. |
| 19:27.41 | ZaVoid | doesn't stop them from thinking they are smart and changing stuff anyway |
| 19:27.45 | ManxPower | NANPA should be all 11 digit dialing |
| 19:27.45 | *** join/#asterisk Ward1983 (n=ward@91.178.41.103) |
| 19:27.49 | ZaVoid | well some countries have say 10 digits |
| 19:27.51 | ZaVoid | others have 11 |
| 19:27.53 | ZaVoid | some have 12 |
| 19:27.57 | ZaVoid | china has 15 for example |
| 19:28.08 | ZaVoid | etc etc etc |
| 19:28.08 | ManxPower | all others would be using a time out |
| 19:28.15 | ManxPower | Just play an invalid message to them |
| 19:28.24 | ZaVoid | yeah that doesn't go over well |
| 19:28.33 | ManxPower | You will NEVER be able to handle all dialing users will try. |
| 19:28.51 | pjz | tell them to treat their deskphone like a cellphone and press 'Dial' when they're done |
| 19:29.03 | ZaVoid | heh they like hitting # for some reason |
| 19:29.04 | ManxPower | pjz: Where is the Dial button on ATAs? |
| 19:29.10 | ZaVoid | culture thing i think |
| 19:29.16 | ManxPower | ZaVoid: so force them to always use # |
| 19:29.19 | ZaVoid | its the # key manx ;) |
| 19:29.23 | *** join/#asterisk tobias (n=tobias@cpe-069-134-205-184.nc.res.rr.com) |
| 19:29.33 | ManxPower | ZaVoid: no, the # key is the # key |
| 19:29.38 | ZaVoid | not to them |
| 19:29.43 | ManxPower | Some ATAs will pass it along, others will strip it. |
| 19:30.07 | pjz | ManxPower: oh, ick, ATAs? hmm |
| 19:30.12 | ZaVoid | well if the dialplan is set right it strips it |
| 19:30.17 | ManxPower | ZaVoid: To me a dump truck is always a MACK Truck, but that doesn't mean it's true. |
| 19:30.33 | pjz | ManxPower: I guess you could do sometehing fun with a dialing context and a bunch of single digit state machine stuff |
| 19:30.35 | ZaVoid | now we're just arguing semantics :) |
| 19:30.41 | ManxPower | ZaVoid: I can see no way to do what you want without complex scripts that parse each digit |
| 19:30.47 | ZaVoid | yeah i agree manx |
| 19:30.59 | ZaVoid | maybe i'll try and do it on my registration server or somthing |
| 19:31.03 | ZaVoid | before it hits da asterisk network |
| 19:31.05 | ManxPower | I'm just SO glad I'm not in the ITSP business. |
| 19:31.21 | ManxPower | The corporate world is so much better in so many things like this. |
| 19:31.31 | ManxPower | "Dial the way we tell you or pack up your desk and GET OUT!" |
| 19:31.43 | ZaVoid | :) |
| 19:31.55 | *** join/#asterisk Segnale007 (n=Segnale0@cheetah-tiv-ppp1182.bmts.com) |
| 19:33.11 | *** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler) |
| 19:33.11 | *** mode/#asterisk [+o angler] by ChanServ |
| 19:34.02 | lmadsen | ManxPower: yay ITSPs -- job security for 2 months! |
| 19:34.12 | Ward1983 | is there any way to realtime decode and listen to rtp streams? |
| 19:34.13 | ManxPower | You can always play a message to them, then let the call go thru. That's what we did when we started forcing 11 digit dialing for any calls not destined for the local area code. |
| 19:34.31 | *** join/#asterisk oej (n=olle@114.62-97-206.bkkb.no) |
| 19:34.54 | ZaVoid | lol lmadsen |
| 19:35.16 | ManxPower | They will forget the 1 sometimes, and they still get the message, but the call won't go thu anymore. |
| 19:35.40 | ManxPower | lmadsen: I want to get into teaching/training for Asterisk. |
| 19:35.42 | ZaVoid | its bad enuff i get 00 and 011 sent to me that i strip off |
| 19:35.51 | ManxPower | At least I won't get calls at 9am memorial day about a single broken phone. |
| 19:36.17 | ZaVoid | lol |
| 19:36.17 | lmadsen | ManxPower: heck ya |
| 19:36.26 | lmadsen | ok, luncht time |
| 19:36.32 | ManxPower | lmadsen: so if you hear of anything...... eric@fnords.org is my e-mail |
| 19:36.32 | ZaVoid | sushi? |
| 19:36.34 | Ward1983 | "replace the broken phone" |
| 19:36.35 | Ward1983 | lol |
| 19:36.48 | Ward1983 | and then *click* |
| 19:36.51 | ZaVoid | It works! |
| 19:37.14 | ManxPower | Ward1983: I cursed several times, turned off the ringer then went back to bed. I didn't know it was a broken phone until the next business day. |
| 19:37.32 | ManxPower | All their offices are supposed to be closed on that day. |
| 19:38.27 | Ward1983 | ManxPower, lol |
| 19:38.28 | lupino3 | hello everybody, Asterisk doesn't load queue members in the correct order specified in file queues.conf, so strategies roundrobin and linear (via patch 7279) don't work correctly. Any hints? I use Asterisk 1.4.19 |
| 19:38.33 | lupino3 | thanks in advance :) |
| 19:39.16 | Ward1983 | ManxPower, liek some idiot i know who thinks he makes great music, on first first show his macbook crashed and he was unable to switch it off.... |
| 19:39.27 | Ward1983 | thats even worse then being unable to replace a phone lol |
| 19:39.31 | *** join/#asterisk ReD-MaN (i=rox-ur-s@172-220.static.golden.net) |
| 19:39.48 | Ward1983 | he actually stopped the show lol |
| 19:40.25 | Ward1983 | some peopel are jsut not made to handle electronic devices |
| 19:40.46 | *** join/#asterisk tripps (n=sean@72.20.150.196) |
| 19:41.27 | *** part/#asterisk dalbaech (n=dalbaech@c-98-200-231-201.hsd1.tx.comcast.net) |
| 19:41.34 | Ward1983 | is there any way to realtime decode and listen to rtp streams? |
| 19:41.46 | ZaVoid | sure |
| 19:42.44 | ZaVoid | yu can even record and play back with these http://www.touchstone-inc.com |
| 19:44.03 | Ward1983 | also realtime? |
| 19:44.35 | ZaVoid | think their tools are only win based |
| 19:45.55 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
| 19:46.37 | Ward1983 | yep i saw |
| 19:46.43 | Ward1983 | and not free :) |
| 19:48.21 | *** join/#asterisk BCS-Satori (n=somewher@75.148.21.113) |
| 19:49.19 | BCS-Satori | Is there a way to listen to saved/old voicemail messages other then having to listen to all the new voicemail messages? |
| 19:49.57 | *** join/#asterisk nny_1 (n=Scott_My@64.203.239.83) |
| 19:50.04 | *** join/#asterisk LiNeTuX|Home (n=LiNeTuX@171.117.8.67.cfl.res.rr.com) |
| 19:50.48 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 19:51.03 | lupino3 | heeeeeeeeeeeeeeeeeeeeeeeeeeelp |
| 19:51.08 | lupino3 | :) |
| 19:53.53 | *** join/#asterisk hfb (n=hfb@pool-71-118-254-245.lsanca.dsl-w.verizon.net) |
| 19:55.49 | *** join/#asterisk dieno (i=771e625e@gateway/web/ajax/mibbit.com/x-bc0a2444ea30c589) |
| 19:56.48 | dieno | hello is anyone fimiliar with AboluteLimit and L(x[:y][:z]) |
| 19:56.55 | dieno | how to use them |
| 19:57.08 | lmadsen | what do you mean... the docs are pretty straight forward |
| 19:57.47 | dieno | dont know where to put and how to limit every new call |
| 19:58.00 | Strom_M | the L is an option to Dial() |
| 19:58.43 | dieno | can you please provide me more guide like should i put it local/1NXXNXXXXX@from-intenal,L(10000(; |
| 19:59.36 | [TK]D-Fender | dieno: "core show application dial" <- go read the instructions. The parameter order is clear. |
| 20:00.00 | dieno | ok googlin |
| 20:00.54 | *** join/#asterisk vgster (n=vgster@93.96.221.240) |
| 20:02.27 | Strom_M | whoever thought up this "from-internal" nonsense needs to be shot |
| 20:02.44 | Strom_M | apparently they've never heard of the words "inbound" or "outbound" |
| 20:02.56 | dieno | its outbound |
| 20:03.11 | dieno | i made call using PHP script by originate |
| 20:03.20 | dieno | if you want it then i can send you |
| 20:03.29 | [TK]D-Fender | Strom_M: You should know the most popular source of dialplans conatining a context with that name.... |
| 20:03.44 | Strom_M | [TK]D-Fender: oh, I do |
| 20:04.04 | Strom_M | it starts with A and ends with P and has some kind of faggotry in the middle |
| 20:08.06 | dieno | ok one more thing |
| 20:08.10 | dieno | how to use absolutelimit |
| 20:08.21 | dieno | please provide me a short example |
| 20:08.29 | [TK]D-Fender | dieno: Set it and when the time's up, the call ends. |
| 20:08.43 | dieno | unable to find the appropriate thing to use with parameters |
| 20:08.45 | dieno | for PHP |
| 20:09.06 | *** join/#asterisk msetim (n=msetim@200.195.161.164) |
| 20:09.07 | [TK]D-Fender | dieno: that isn't a PHP function, thats an * dialplan function. |
| 20:09.10 | dieno | where should i put hits AbsoluteTimout(10) means 10 sex |
| 20:09.29 | *** join/#asterisk phy (n=phy@funky.monkey.org) |
| 20:10.03 | dieno | hmm but if use fput Action Originate means can make call then we can use this as well like fput Action AbsoluteTimeout(10) |
| 20:11.14 | *** join/#asterisk bootc (n=bootc@arcadia.prv.bootc.net) |
| 20:11.15 | [TK]D-Fender | dieno: it needs to be executed in the dialplan, this is NOT something you'd do in a separate AMI request or passed as some sort of "parameter" |
| 20:12.00 | bootc | is there any way I can influence what gets passed in OPTIONS when Asterisk qualifies a SIP friend? |
| 20:12.26 | tzanger | hmm did anyone else get an email from voipsupply about their 101 uses of asterisk blog entry? |
| 20:12.48 | Qwell | tzanger: yep |
| 20:13.00 | tzanger | heh |
| 20:13.02 | Qwell | form letter |
| 20:13.16 | [TK]D-Fender | bootc: You've got the source, jsut like the rest of us. |
| 20:13.48 | bootc | [TK]D-Fender: I have been reading through it, but I figured there might be a configuration parameter I'd missed or something |
| 20:14.00 | *** join/#asterisk d-k-t (n=dt@125.120.129.131) |
| 20:14.06 | [TK]D-Fender | bootc: chan_sip.c |
| 20:14.18 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 20:14.40 | dieno | [TK]D-Fender have you tried SippySkype ? |
| 20:14.51 | [TK]D-Fender | dieno: No. |
| 20:15.05 | hsv-al | d-fender |
| 20:15.06 | hsv-al | http://pastebin.com/m272544bc |
| 20:15.21 | hsv-al | I have everything working fine now, but originally I had infinite loops when the goto statements contained invalid extensions |
| 20:15.24 | [TK]D-Fender | dieno: You are painting an incrementally worse picture of yourself with every passing question... |
| 20:15.31 | bootc | it sends OPTIONS sip:<host>, but I'd like OPTIONS sip:<foo>@<host> |
| 20:15.41 | hsv-al | it would loop the "invalid extension wav file" forever, when I had non existant entries in there, what caused that? |
| 20:15.43 | [TK]D-Fender | bootc: You know where to go... |
| 20:16.10 | bootc | fair enough, maybe you can explain how the fullcontact is generated for a SIP friend? |
| 20:16.21 | [TK]D-Fender | hsv-al: Stop making invalid goto's then. |
| 20:16.29 | hsv-al | i know, its all good now |
| 20:16.36 | hsv-al | but originally, why was asterisk looping forever like that |
| 20:16.39 | hsv-al | instead of just hanging up? |
| 20:16.46 | [TK]D-Fender | bootc: Its all in the source, and few mess around in there. |
| 20:17.00 | [TK]D-Fender | hsv-al: because things keep being invalid. |
| 20:17.37 | nny_1 | any Linksys 932 users here? I am trying to address an issue: When I restart asterisk or any connectivity is lost, i have to restart the phone. Is there a way around this? Seems like it should just reestablish the sip registration and go. |
| 20:18.18 | nny_1 | clarify: i have to restart the phone to get blfs to work |
| 20:18.19 | ManxPower | nny_1: put an ip in the phone, not a hostname |
| 20:18.27 | ManxPower | nny_1: change your reg interval on the phone to be shorter |
| 20:18.30 | nny_1 | ManxPower: thanks will try |
| 20:19.54 | oej | bootc: What do you mean with "gets passed" in the OPTIONs? |
| 20:21.58 | bootc | well, the actual OPTIONS request is "OPTIONS sip:<hostname>" |
| 20:22.06 | SuPrSluG | hi all |
| 20:22.09 | *** join/#asterisk javar (n=javar@200.118.170.56) |
| 20:22.33 | ManxPower | waves to oej |
| 20:22.47 | bootc | oej: and I need it to be "OPTIONS sip:<foo>@<hostname>" |
| 20:23.01 | oej | waves back |
| 20:23.12 | oej | bootc: For Asterisk it doesn't matter. |
| 20:23.12 | bootc | either that or I'm barking up completely the wrong tree |
| 20:23.18 | ManxPower | Is "OPTIONS sip:<foo>@<hostname>" permitted in the RFCs? |
| 20:23.20 | oej | bootc: Why does this matter for you? |
| 20:23.24 | bootc | oej: no, but I'm talking to OpenSER |
| 20:23.27 | oej | ManxPower: Yes |
| 20:23.34 | ManxPower | oej: Ah. |
| 20:23.46 | oej | bootc: Well, again, ASterisk doesn't bother with the answer, as long as we get one |
| 20:23.50 | ManxPower | bootc: why not just disable OPTIONS |
| 20:23.51 | SuPrSluG | <PROTECTED> |
| 20:23.59 | bootc | I want to qualify an OpenSER peer, and it's returning 404's constantly and filling my logs |
| 20:24.00 | oej | bootc: And if you configure OpenSER properly, OpenSER doesn't bother either. |
| 20:24.13 | bootc | oej: any hints on that last bit then? |
| 20:24.14 | [TK]D-Fender | SuPrSluG: "core show function GROUP_COUNT" |
| 20:24.37 | SuPrSluG | k |
| 20:24.48 | *** join/#asterisk l2cache (n=chatzill@97.101.178.117) |
| 20:24.50 | oej | bootc: I see. So you don't want to qualify OpenSER, but something behind it, right? |
| 20:25.15 | bootc | oej: no I do actually want to qualify OpenSER :-/ |
| 20:25.42 | ManxPower | bootc: Qualify in Asterisk SIP's stack is horrid and I suggest you not use it |
| 20:25.56 | oej | bootc: So why is the username important then? Just check if the request URI = myself and then answer the OPTIONs request with 200 OK. OpenSER has a module for that if you want to do it properly too. |
| 20:25.56 | ManxPower | See the IAX2 qualify smoothing feature for why. |
| 20:26.21 | oej | ManxPower: I don't understand you there. Please tell me why it's horrid |
| 20:26.27 | bootc | oej: yeah I guess, I imagined it worked somehow otherwise |
| 20:27.05 | oej | bootc: I guess it's hard coding to user's imagination, but it would certainly be cool :-) |
| 20:27.20 | ManxPower | oej: For all the reasons IAX2 needed qualify smoothing. i.e. even one lost packet will make a device unreachable, if the device makes the OPTIONS a low priority and is busy, then the host could easily become lagged, even when it's not really lagged. |
| 20:27.26 | oej | bootc: The important thing for you to know is that for Asterisk, we don't care about the answer. For log files, it doesn't look nice, I agree. |
| 20:27.38 | ManxPower | qualify smoothing would either significantly reduce these issues or eliminate them |
| 20:27.48 | oej | ManxPower: That's wrong. We do retransmit OPTIONs several times, according to the RFCs |
| 20:28.06 | oej | ManxPower: So ONE transmission, with several retransmits, will make the peer unreachable |
| 20:28.12 | oej | ManxPower: But one lost packet won't. |
| 20:28.14 | ManxPower | oej: then why do hosts on a link with no loss suddenly go lagged? |
| 20:28.41 | ManxPower | oej: did that change between 1.2 and 1.4? |
| 20:28.49 | oej | ManxPower: That's something we need to debug. For some devices (Nokia Phones and Grandstream) they respond slowly when they're busy talking. |
| 20:28.56 | ManxPower | I do know that 1.4 does not have qualify smoothing -- it's one of the things that pisses me off about the 1.4 release |
| 20:28.58 | oej | So they're really lagged. |
| 20:29.27 | oej | I don't think this has changed in 1.4, it's been there for a long time. I've been discussing this with young Madsen several times and proven to him he was wrong here. |
| 20:29.44 | Strom_M | but...1-800-URINE isn't even eleven digits |
| 20:29.49 | ManxPower | If you proved blitzrage wrong, then that's good enough for me. |
| 20:29.59 | oej | It's just one of these rumours, Like for the first couple of years everyone on the mailing list said "Yes, you have to have a sound card to run Asterisk" |
| 20:30.14 | ManxPower | It does not change the fact that frequently SIP devices go lagged or unreachable when they do not appear to be. |
| 20:30.16 | oej | I'm still curious on why you're getting LAGGED though. |
| 20:30.26 | x86 | oej: don't _have_ to, but it makes for some cool applications using the console :) |
| 20:30.33 | ManxPower | Not me, I turned off qualify YEARS ago and never looked back |
| 20:31.18 | oej | ManxPower: Don't look back. You propably have a massive amount of old OPTION messages following your tracks.... Evil OPTION messages. |
| 20:31.50 | oej | A lot of things has happened to the SIP stack in those years, so we've propably changed something for the better. I hope. |
| 20:32.26 | oej | I have a server that monitors hundreds of SIP devices and don't see that problem. I guess I have to check log files a bit more. |
| 20:32.27 | ManxPower | oej: almost all of my devices are either on a local LAN with the Asterisk server or are on QoS'd links to the Asterisk server. |
| 20:32.28 | bootc | oej: a simple if (method=="OPTIONS" && uri==myself) sl_send_reply("200", "Still alive"); did indeed fix my problem |
| 20:32.36 | bootc | barking up the wrong tree again, thanks |
| 20:32.49 | oej | bootc: No problem. :-) |
| 20:33.14 | oej | I like sending "200 Still alive". YOu should add a random "699 I'm dead tired" |
| 20:33.17 | ManxPower | oej: at least the SIP stack, with all of it's problems, is much more reliable than the IAX2 stack for many people. |
| 20:33.39 | oej | ManxPower: ...but there's still a ton of fixes that needs fixing in the SIP stack. |
| 20:34.04 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 20:34.07 | ManxPower | We switched to SIP for inter-asterisk stuff several years ago when we could not make the random disconnects on IAX2 calls stop. |
| 20:34.37 | oej | ...and you can send channelv ariables and stuff across easily. With MD5 checksums too :-) |
| 20:34.45 | ManxPower | Oddly, some people have NO problems with IAX2, others have random disconnects. I'm sure it's a bitch to try to debug. |
| 20:34.52 | oej | There was a reason I added MD5 at the same time as sipaddheader |
| 20:35.21 | oej | IAX2 has so many problems with NAT. |
| 20:35.25 | oej | (ha ha ha) |
| 20:36.30 | ManxPower | oej: I find the rumor "SIP and NAT cause all sorts of problems" to be very funny. |
| 20:36.50 | *** join/#asterisk LiNeTuX|Home (n=LiNeTuX@171.117.8.67.cfl.res.rr.com) |
| 20:37.15 | oej | SIP and NAT is very good for my consulting business |
| 20:37.17 | ManxPower | It took me all of 30 mins to set up my Asterisk server and SIPura client to transparently move between local network, internet not behind NAT and internet behind NAT with my Asterisk server on NAT. |
| 20:37.24 | oej | I like NATs. More NATs in the network boys and girls! |
| 20:37.30 | ManxPower | Granted, I had a Cisco router and a static IP at the time. |
| 20:37.32 | oej | And badly configured firewalls. Tons of them. |
| 20:37.46 | oej | Those are the best. |
| 20:38.07 | *** join/#asterisk bminish (n=bminish@brenbox.home.minish.org) |
| 20:38.34 | florz | Any suggested way for limiting a sip client to only being able to call toll-free numbers (^0800.*) on a specific line? |
| 20:38.46 | ManxPower | I used DNS SRV records to allow the ATA to roam between non-NAT and NAT |
| 20:38.48 | Strom_M | 1-800-URINE |
| 20:39.21 | ManxPower | florz: SIP clients don't actually have lines, they have call appearances. If you have each call appearance on your phone use a different SIP user/pass then what you want to do is trivbial |
| 20:39.26 | *** join/#asterisk _MrSeb_ (n=SebaX@87.253.113.240) |
| 20:39.27 | _MrSeb_ | Hi to all |
| 20:39.30 | ManxPower | and trivial to. Using contexts and a good dialplan |
| 20:39.42 | florz | ManxPower: by "line" I mean the uplink to dial out through |
| 20:39.43 | ManxPower | *whine* My arms hurt. |
| 20:40.01 | *** join/#asterisk seva (i=seva@sevatech.com) |
| 20:40.04 | ManxPower | florz: use the correct terms. |
| 20:40.10 | ManxPower | line and uplink are not. |
| 20:40.16 | seva | i would like to execute a command in the dialplan that gets the caller id as the argument |
| 20:40.26 | seva | what documentation should i be looking at? |
| 20:40.36 | ManxPower | "Can I limit my softphone to only dial certian numbers, and for the numbers it is allowed to dial, use a specific Zap line?" is a good question. |
| 20:40.55 | ManxPower | seva: "core show application system" |
| 20:40.58 | *** join/#asterisk nny_2 (n=Scott_My@64.203.239.83) |
| 20:41.06 | ManxPower | florz: is that what you want? |
| 20:41.10 | florz | ManxPower: well, it's a E1 "line", which connects to the PSTN, so one could call it the uplink ... |
| 20:41.18 | florz | ManxPower: yes |
| 20:41.19 | ManxPower | florz: so it is a Zap port? |
| 20:41.23 | florz | yep |
| 20:41.26 | seva | thanks http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+System |
| 20:41.27 | *** part/#asterisk seva (i=seva@sevatech.com) |
| 20:41.29 | ManxPower | What zap channel or group do you want to use? |
| 20:41.39 | ManxPower | florz: put that one client in a separate context |
| 20:41.49 | florz | well, just assume g1 |
| 20:42.16 | ManxPower | seva: The wiki is usually out of date. My way gives you the CURRENT docs for THAT application on the Asterisk VERSION you are using. |
| 20:42.17 | florz | ManxPower: well, yeah, and then? |
| 20:42.45 | ManxPower | florz: [toll-free-access] exten => _0800.,1,Dial(Zap/g1/${EXTEN}) |
| 20:42.55 | ManxPower | where the devices in in the context=toll-free-access |
| 20:43.05 | florz | ManxPower: well, that obviously doesn't work |
| 20:43.22 | ManxPower | florz: why not? |
| 20:43.26 | ManxPower | It does what you want to do. |
| 20:43.56 | *** join/#asterisk deeperror (n=deeperro@adsl-76-234-138-5.dsl.sfldmi.sbcglobal.net) |
| 20:44.10 | florz | ManxPower: because you can dial, say, sip:0800&Zap/g1/whatever@asteriskbox |
| 20:44.34 | ManxPower | florz: your statement makes no sense |
| 20:44.36 | nny_2 | hmm |
| 20:45.04 | ManxPower | florz: what you typed is not valid in any part of Asterisk |
| 20:45.11 | florz | ManxPower: well: If you dial that URI on the client to be limited ... see what happens? |
| 20:45.20 | ManxPower | Asterisk will reject that URI |
| 20:45.31 | florz | ManxPower: why should it? |
| 20:45.57 | ManxPower | Because it does not match and exten => line. |
| 20:45.59 | ManxPower | Why don't you just TRY it an see. |
| 20:46.13 | florz | Well, I tried and saw =:-) |
| 20:46.30 | ManxPower | You tried it in your CURRENT config, not in the config I suggested. |
| 20:46.33 | nny_2 | ok i have both retry in the 932 settings as well as retry in the 962 SIP entry set to 5. I have changed the proxy to an IP vs. DNS. the phone reregisters as soon as I restart asterisk, but the 932 sidecar stays "inacitve" any further advice? |
| 20:46.42 | *** join/#asterisk CVirus (n=GoD@82.201.174.159) |
| 20:46.56 | ManxPower | nny_1 contact Linksys |
| 20:47.02 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 20:47.12 | florz | sure did I try it in the config you suggested - because that's exactly what I tried before asking |
| 20:47.14 | nny_2 | ManxPower: hah gotcha |
| 20:47.36 | jblack | Hey, is there a way to reset billsec during a call? |
| 20:47.44 | ManxPower | florz: I cannot help you further, best of luck. |
| 20:47.52 | ManxPower | jblack: I don't bill for calls. |
| 20:48.11 | ManxPower | My users get free calling |
| 20:48.14 | jblack | Nor do I. |
| 20:48.27 | ManxPower | Granted, they pay $40,000/yr to work for the company |
| 20:48.43 | *** join/#asterisk knobo` (n=user@ti0153a380-0024.bb.online.no) |
| 20:48.50 | ManxPower | That gives them a desk, phone, internet connection, and access to the marketing department. |
| 20:48.54 | jblack | We use billsec as one of several metrics to determine call agent effectiveness. |
| 20:49.18 | ManxPower | jblack: we don't track that either. If they don't make enough money per month they are fired. |
| 20:49.22 | knobo` | Can Asterisk be a h323 gatekeeper. |
| 20:49.46 | jblack | I'd like to make them more competitive by listing their ranking for the day, but that would futz up billsec. |
| 20:50.24 | florz | So, anyone else got any ideas as to how to (easily) limit to toll-free? |
| 20:50.26 | ManxPower | jblack: we stopped using queues years ago. Our needs are simple and better handled by the dialplan. |
| 20:50.41 | jblack | I'm not using queues either, at least for the agents. |
| 20:51.14 | jblack | What I'd like, is whenever they dial a new customer, say "You are # xx", then complete the call. |
| 20:52.08 | jblack | which is easy to do, but will cause an answer, which will start the billsecs clock |
| 20:52.08 | ManxPower | jblack: Uh, almost all things that play audio in Asterisk have a noanswer option, but you knew that because you checked the docs, right? |
| 20:52.20 | jblack | Did I do something to make you so hostile today? |
| 20:53.01 | ManxPower | jblack: only when you ask questions that are covered in the application docs |
| 20:53.31 | ManxPower | I assume you would use Playback(), check the docs for that in the CLI, look for the noanswer options. |
| 20:53.48 | jblack | you've seen me help a lot of people and refer them to the book dozens, if not hundreds, of times with the book, often giving page numbers in the process. Surely you can realize that I'd have checked the book if I had any idea such an option existed at all. |
| 20:54.25 | ManxPower | *sigh* The book is out of date before it is printed for some things, like application docs. |
| 20:54.48 | ManxPower | type "core show application playback" in the CLI. That is the docs for Playback for whatever version you are using. |
| 20:55.44 | jblack | Ok. Playback doesn't have any reference about avoiding an implicit answer as well. |
| 20:55.47 | ManxPower | The INCLUDED docs with Asterisk are always better than the book or the Wiki. The drawback of the built in docs is that they don't have much general information -- that's where the book and the Wiki are great. |
| 20:55.58 | jblack | I imagine you're looking at a newer version of asterisk than I'm using. |
| 20:56.16 | Strom_M | playback has had a "noanswer" flag forever |
| 20:56.32 | Strom_M | get with the program, all of you :) |
| 20:56.34 | jblack | I don't see it here in the show applicaiton playback page. |
| 20:56.49 | ManxPower | jblack: Strange as it DOES have that option in MY asterisk CLI |
| 20:56.56 | Strom_M | what the hell are you running? 0.7 |
| 20:56.57 | Strom_M | ? |
| 20:57.00 | jblack | 1.4.17 |
| 20:57.18 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 20:57.48 | Strom_M | read the block of text |
| 20:57.50 | ManxPower | " If 'skip' is specified, the application will return immediately should the channel not be off hook. Otherwise,...." Playback is talked about starting with the word after my quote |
| 20:58.23 | Strom_M | "Otherwise, unless 'noanswer' is specified, the channel will |
| 20:58.23 | Strom_M | be answered before the sound is played. Not all channels support playing |
| 20:58.23 | Strom_M | messages while still on hook. " |
| 20:58.36 | ManxPower | Strom_M: Handing him a fish again? |
| 20:58.37 | jblack | I'm blind |
| 20:58.49 | Strom_M | ManxPower: actually, trying to bludgeon him with the fish |
| 20:58.59 | ManxPower | jblack: get a better screen reader! *tease* |
| 20:59.10 | jblack | Heh. |
| 20:59.13 | ManxPower | Strom_M: A NERF bat doesn't smell as bad. |
| 20:59.26 | Strom_M | ManxPower: if it doesnt smell bad then what's the point? |
| 20:59.35 | ManxPower | A NERF bat shaped like a fish...... |
| 20:59.40 | ManxPower | It would be a hit. |
| 20:59.49 | _MrSeb_ | someone can help me and explain how i can change the ip in contact info in sip packet? |
| 21:00.07 | ManxPower | _MrSeb_: It sounds like you have a NAT problem. |
| 21:00.14 | *** join/#asterisk Cyorxamp (n=Cyorxamp@212.57.229.111) |
| 21:00.16 | Cyorxamp | Does anyone in here have a Nokia E51 and can tell me if the built in SIP client can work over a 3G connection as well as the WiFi? |
| 21:00.32 | jblack | mrseb: Is that what you're dealing with? Trying to get through nat? |
| 21:00.35 | ManxPower | Cyorxamp: what is the latency on your "3G" connection? |
| 21:01.01 | Cyorxamp | ManxPower: at WCMDA it's 384kbps I think, at HSDPA its up to 3.4mbps |
| 21:01.15 | _MrSeb_ | ManxPower: yes, but all is working, only asterisk don't know what interface to use... I must bind it only on external interface to make it working, and I need it on internal too for pc and voip phone |
| 21:01.17 | jblack | mrseb: And if so, what's behind the nat? *, or your clients. |
| 21:01.38 | jblack | mrseb: If it's an option, you can bind to all interfaces with 0.0.0.0 |
| 21:02.18 | ManxPower | _MrSeb_: Do not bind to any specific interface and turn off reinvites. That should fix it for you. |
| 21:02.44 | _MrSeb_ | ManxPower: I will try it now, thanks |
| 21:02.48 | ManxPower | bindaddr=0.0.0.0 is EXACTLY the same as not having a bindaddr |
| 21:03.08 | ManxPower | _MrSeb_: If that does not work there are several other things to try. |
| 21:03.26 | ManxPower | _MrSeb_: The OPERATING system will correctly pick which interface to use. |
| 21:03.29 | *** join/#asterisk s0ck (n=m@unaffiliated/s0ck) |
| 21:04.04 | _MrSeb_ | ManxPower: before I put a router all was working, now nothing... I go to try |
| 21:04.29 | ManxPower | _MrSeb_: you said your Asterisk is on a public IP, idn't you? |
| 21:04.45 | ManxPower | Cyorxamp: NO! That is the BANDWIDTH. Not LATENCY |
| 21:04.59 | Cyorxamp | you don't actually know if it can or not... nevermind] |
| 21:05.08 | Cyorxamp | thanks for trying to understand the question - you did your best |
| 21:05.10 | *** part/#asterisk Cyorxamp (n=Cyorxamp@212.57.229.111) |
| 21:05.22 | _MrSeb_ | ManxPower: no, my asterisk is on dmz, all traffic on router is redirected to asterisk server, it is a firewall too |
| 21:05.45 | ManxPower | _MrSeb_: Ah! Then you need the externip= and localnet= options in sip.conf [general] |
| 21:05.56 | _MrSeb_ | ManxPower: already configured |
| 21:05.59 | ManxPower | since the Asterisk server IS behind NAT -- DMZ means almost nothing to SIP. |
| 21:06.22 | ManxPower | _MrSeb_: what is your externip= setting? |
| 21:06.37 | ManxPower | and localnet= should be your INTERNAL PHONE network. |
| 21:08.05 | _MrSeb_ | ManxPower: externip = 87.253.113.240 |
| 21:08.20 | ManxPower | Good. |
| 21:08.26 | _MrSeb_ | ManxPower: localnet=192.168.0.0/255.255.0.0 |
| 21:08.27 | ManxPower | And your localnet= setting? |
| 21:08.45 | ManxPower | Ah. And what is the internal network and what is the DMZ addressing? |
| 21:08.59 | _MrSeb_ | ManxPower: and localnet=10.0.0.0/255.0.0.0 |
| 21:09.17 | ManxPower | No! only use one localnet option for now. |
| 21:09.45 | ManxPower | At least you did not screw up the netmask and you used an IP for exterip. Most people screw those up. |
| 21:09.45 | _MrSeb_ | ManxPower: here is my network topology http://sebhack.007sites.com/Asterisk.png |
| 21:10.03 | *** join/#asterisk EricL (n=eric@jarbeeg.chal.net) |
| 21:10.20 | _MrSeb_ | ManxPower: I've only one pc in dmz, before it was the firewall, with ppp0 |
| 21:10.29 | ManxPower | 192.168.0.0/255.255.0.0 is different from 192.168.0.0/24 |
| 21:10.37 | _MrSeb_ | yes |
| 21:10.45 | ManxPower | so which is correct? |
| 21:10.59 | _MrSeb_ | it must be the same? I've the vpn too |
| 21:11.06 | ManxPower | Try removing the localnet=10.0.0.0/255.0.0.0 and do a reload_chan_sip.so |
| 21:11.28 | _MrSeb_ | ok |
| 21:11.29 | ManxPower | _MrSeb_: get it working for local clients before trying to make it work for VPN clients. |
| 21:11.40 | hsv-al | analog phone plugged into slot 2 on the tdm card |
| 21:11.42 | hsv-al | = Zap/2? |
| 21:11.43 | ManxPower | sorry, core reload chan_sip.so I think |
| 21:11.53 | ManxPower | hsv-al: only one card? |
| 21:12.00 | hsv-al | have an fxo and fxos |
| 21:12.05 | hsv-al | landline into slot 1 |
| 21:12.07 | hsv-al | fxs in slot 2 |
| 21:12.15 | ManxPower | hsv-al: only one card? |
| 21:12.16 | hsv-al | s/fxos/fxs |
| 21:12.19 | hsv-al | yep |
| 21:12.22 | hsv-al | 1 TDM411P |
| 21:12.28 | ManxPower | then yes, port 2 on the TDM card would be Zap/2 |
| 21:13.09 | _MrSeb_ | ManxPower: I've restarted, no problem for now |
| 21:13.34 | _MrSeb_ | ManxPower: no, I get [Jun 10 23:13:12] NOTICE[13899]: chan_sip.c:7492 sip_reg_timeout: -- Registration for '5334264@sip.messagenet.it' timed out, trying again (Attempt #1) |
| 21:13.36 | ManxPower | _MrSeb_: Great! Do some more testing, then change the netmask to include your VPN users. |
| 21:14.02 | hardwire | I wonder how much of an overhaul it would be to have zap cards identified uniquely or at least as card 1 and 2, so that spans could be card 1 span 1, etc.. |
| 21:14.11 | ManxPower | _MrSeb_: I suspect that is a firewall issue |
| 21:14.48 | *** join/#asterisk Defraz (i=t0tal@69.92.19.83) |
| 21:14.51 | ManxPower | _MrSeb_: BTW, make a backup copy of your WORKING config. |
| 21:15.12 | *** join/#asterisk flush (n=SYN_SENT@ip216-239-89-134.vif.net) |
| 21:15.16 | ManxPower | _MrSeb_: Also try adding in the 2nd localnet with the 10.xxxxx address in it. Only change one thing at a time, of course. |
| 21:15.19 | _MrSeb_ | ManxPower: this is in sip packet: Contact: <sip:01123416043@192.168.0.127>... all work if I bind asterisk only on 10.0.0.1, so the packet contain @10.0.0.1 |
| 21:15.59 | _MrSeb_ | ManxPower: (all is not inclusive of clients...) |
| 21:16.12 | [TK]D-Fender | _MrSeb_: You've clearly not configured your system to work properly with NAT. Read up : |
| 21:16.12 | [TK]D-Fender | ~sipnat |
| 21:16.13 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 21:16.23 | x86 | gah... 30 minutes left at work... and i'm sick... this sucks |
| 21:16.23 | ManxPower | _MrSeb_: any chance of not making Asterisk an unneeded gateway? |
| 21:16.32 | x86 | the last 30 minutes always seem to snail along |
| 21:17.31 | _MrSeb_ | [TK]D-Fender: I've already visited this sita and all 9 topology, but nothing has worked |
| 21:17.42 | [TK]D-Fender | _MrSeb_: 1st link |
| 21:17.53 | [TK]D-Fender | _MrSeb_: And pastebin your sip.conf masking only passwords |
| 21:17.55 | [TK]D-Fender | ~pb |
| 21:17.56 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 21:18.20 | _MrSeb_ | [TK]D-Fender: my server is client for sip provider, but server for internal network |
| 21:18.56 | _MrSeb_ | ManxPower: no, this is the only possibility, I've problem with network cables |
| 21:19.16 | [TK]D-Fender | _MrSeb_: Yes, now pASTEBIN IT |
| 21:19.17 | _MrSeb_ | [TK]D-Fender: ok, wait a moment |
| 21:19.53 | hsv-al | cant find a sip client |
| 21:19.57 | hsv-al | for 8703e, bleh |
| 21:22.56 | *** join/#asterisk KenLee (n=k3leland@bg-fw2out.monmouth.com) |
| 21:22.56 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 21:23.24 | _MrSeb_ | [TK]D-Fender and ManxPower: my sip conf is at http://rafb.net/p/fuJl0L20.html |
| 21:24.44 | [TK]D-Fender | _MrSeb_: you have to put your "register" lines AFTER all the rest of [general] |
| 21:25.13 | [TK]D-Fender | _MrSeb_: and you should not be specifying externhost AND externip at the same time. |
| 21:25.40 | _MrSeb_ | ok, I comment exterhost |
| 21:26.05 | [TK]D-Fender | _MrSeb_: And you are missing "nat=yes" , and "canreinvite=no" under [general] |
| 21:26.17 | [TK]D-Fender | _MrSeb_: You don't appear to have read the guides properly at all. |
| 21:26.37 | *** join/#asterisk tobias (n=tobias@cpe-069-134-205-184.nc.res.rr.com) |
| 21:26.56 | _MrSeb_ | [TK]D-Fender: I've tryed many thing, this is the better working if I bind to "external" ip only |
| 21:27.26 | quark_ | is there any reason that every time I try to send an OPTIONS request to the asterisk server I get a 404? |
| 21:27.29 | _MrSeb_ | with other setting I was not able to register on sip provider |
| 21:27.36 | quark_ | something misconfigured perhaps? |
| 21:28.40 | quark_ | it does not matter at all where the request comes from |
| 21:28.42 | [TK]D-Fender | _MrSeb_: fix everything I told you to, READ the guide AGAIN. Test. If that failes, pastebin your "new" sip.conf |
| 21:29.05 | _MrSeb_ | [TK]D-Fender: ok, thanks |
| 21:29.12 | [TK]D-Fender | quark_: * always 404's Options packets. It is only expcting a response to it, not necessarily a POSITIVE one. |
| 21:29.56 | quark_ | but I've seen it not send a 404 |
| 21:29.59 | quark_ | and give a 200 |
| 21:29.59 | *** join/#asterisk ajricoveri (n=ajricove@190.37.169.212) |
| 21:30.04 | quark_ | let me see here |
| 21:30.06 | quark_ | I have a video |
| 21:30.16 | quark_ | http://blip.tv/file/621538 |
| 21:30.36 | quark_ | when he does it in the video he gets a 200 OK rather than a 404 |
| 21:30.47 | ajricoveri | hi, im currently testing AsteriskNOW and i think it is actually great, here is my doubt |
| 21:31.35 | ajricoveri | can i add some extension and "separate" them in groups so an extension from group A cant call an extension from group B ??? |
| 21:32.15 | quark_ | [TK]D-Fender: any idea how I would not get a 404? |
| 21:32.57 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 21:33.05 | keith4 | ~asterisknow |
| 21:33.06 | jbot | somebody said asterisknow was based on Asterisk, but it is not Asterisk, and it is unlikely to live up to Asterisk's standards. Only Asterisk is supported on #asterisk. Use #AsteriskNow instead. Even if the channel happens to be less helpful, support for systems other than Asterisk is offtopic on #asterisk |
| 21:33.53 | [TK]D-Fender | quark_: ok, not sure here.... |
| 21:34.53 | ajricoveri | i know asterisknow is not the same thing .... but can i at least know if asterisk is capable of doing that?? thank u |
| 21:35.07 | KenLee | asterisk is absolutely capable of doing that |
| 21:36.12 | [TK]D-Fender | ajricoveri: Yes you can do that quite easily with straight configuration you do yourself. |
| 21:36.35 | ajricoveri | noob question, is there an special term for that?? i mean, i have been googling that for hours, the nearest thing i have to be a solution candidate is doing virtualization with various pbx's |
| 21:36.36 | *** part/#asterisk brodiem (n=brodiem@rrcs-24-227-171-186.sw.biz.rr.com) |
| 21:39.04 | KenLee | you could accomplish that through virtualization but unless you had alterior motives that would be overkill |
| 21:39.06 | ajricoveri | noob question, is there any special term for that?? i mean, i have been googling that for hours, the nearest thing i have found as a solution candidate is doing virtualization with some asterisknow instances running (sorry for my english) |
| 21:39.25 | anonymouz666 | quark_: if you have a extension to match the SIP Options you will get 200 OK, otherwise 404 for SIP OPTIONS. |
| 21:39.32 | ajricoveri | i know it will be overkill, it is actually the last thing i want to do ... |
| 21:39.57 | [TK]D-Fender | ajricoveri: You need to build proper CONTEXTS in your dialplan. This is all in CHAPTER % of the BOOK. |
| 21:39.59 | *** part/#asterisk hubguruJR (n=hubguruJ@mail.ntegratedsolutions.com) |
| 21:39.59 | [TK]D-Fender | ~book |
| 21:40.00 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 21:40.17 | _MrSeb_ | [TK]D-Fender: always the same thing, timeout on sip provider registration and the sip packet contain wrong ip (Contact: <sip:01123416043@192.168.0.127> and to get it working it must be 10.0.0.1)... here is the current configuration http://rafb.net/p/zcAwQD94.html |
| 21:40.38 | [TK]D-Fender | ajricoveri: AsteriskNOW and the GUI it comes with will not let you do jsut anything you want with it. It has a rather limited flow in its design. |
| 21:41.01 | *** part/#asterisk KenLee (n=k3leland@bg-fw2out.monmouth.com) |
| 21:41.07 | luke-jr | hsv-al: why are you impersonating my bot? |
| 21:42.34 | [TK]D-Fender | _MrSeb_: you also failed to tell * that that other subnet was LOCAL <- |
| 21:42.57 | [TK]D-Fender | _MrSeb_: And please insert some blank lines between sections. Thats 1 gian t mess. |
| 21:43.23 | [TK]D-Fender | _MrSeb_: and why would * need to report back a PRIVATE return IP for that register? |
| 21:43.58 | _MrSeb_ | [TK]D-Fender: I've grepped only useful line... |
| 21:44.59 | _MrSeb_ | [TK]D-Fender: the complete file sip.conf has many comments |
| 21:46.08 | _MrSeb_ | [TK]D-Fender: for other subnet local I've tryed before with 10.0.0.0/255.0.0.0 too |
| 21:47.40 | _MrSeb_ | [TK]D-Fender: the registration work if I bind asterisk server on 10.0.0.1, so the sip packet contain "Contact: <sip:01123416043@10.0.0.1>" |
| 21:49.58 | _MrSeb_ | [TK]D-Fender: I've not found other manner to get it working... when I go in debug mode I know the result reading sip packet info |
| 21:51.46 | *** part/#asterisk EricL (n=eric@jarbeeg.chal.net) |
| 21:55.34 | *** join/#asterisk spokra (n=spokra@host093-179-178.sea0.speakeasy.net) |
| 22:02.52 | *** join/#asterisk UnixDog (n=UnixDog@254.69.118.70.cfl.res.rr.com) |
| 22:03.35 | *** join/#asterisk eXistenZ (i=pectic@unaffiliated/existenz) |
| 22:04.58 | [TK]D-Fender | _MrSeb_: You should NOT be telling an ITSP to contact you at a PRIVATE IP. |
| 22:06.01 | [TK]D-Fender | ok, stepping out for a while. |
| 22:06.16 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 22:06.49 | enemy^x | does anyone know if fax (rxfax) works with asterisk v1.6 |
| 22:07.24 | _MrSeb_ | [TK]D-Fender: yes, but contact info must contain natted IP, the udp packet contain public ip, and then natted, but contact should be correct... the registration work in this manner and I don't get any timeout |
| 22:07.30 | UnixDog | not that I know off |
| 22:07.31 | UnixDog | of |
| 22:07.42 | UnixDog | you also need spandsp |
| 22:08.12 | spokra | any AGI experts here? |
| 22:08.14 | *** join/#asterisk bminish (n=bminish@brenbox.home.minish.org) |
| 22:09.07 | UnixDog | what lang you writing the agi in |
| 22:09.17 | UnixDog | perl/php/python ? |
| 22:09.17 | spokra | looking at adhearsion |
| 22:09.25 | UnixDog | never touched it |
| 22:09.38 | spokra | well might be a generic problem. |
| 22:10.34 | spokra | the adhearsion (ruby) script tries to play files after the line disconects and asterisk spews errors.. |
| 22:11.28 | spokra | how does a AGI script know the line has disconnected? |
| 22:12.51 | *** join/#asterisk deeperror (n=deeperro@d149-67-253-63.try.wideopenwest.com) |
| 22:24.08 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:26.16 | russellb | AGI should stop running in that case |
| 22:26.25 | *** join/#asterisk korihor (n=korihor@190.199.171.145) |
| 22:26.43 | spokra | does asterisk frop the tcp connection to the AGI script? |
| 22:26.56 | spokra | s/frop/drop/ |
| 22:27.05 | russellb | it should |
| 22:27.37 | spokra | Hmmm any known problems with 1.6.beta and AGI? |
| 22:27.49 | russellb | not known issues, no |
| 22:28.10 | russellb | feel free to report it to bugs.digium.com |
| 22:28.15 | spokra | well i will do some TCP dumps.. and see why it's not stopping |
| 22:28.30 | russellb | but it would be most helpful if you could reproduce it with a simple AGI, and not a framework on top of it |
| 22:28.39 | spokra | or just move back to a more stable version!! |
| 22:29.02 | spokra | I'll try that!! |
| 22:30.43 | jaytee | anyone using * with click2call in MS Outlook? |
| 22:37.23 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 22:42.29 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 22:43.05 | [hC] | anyone know much about a bug in older 1.4 builds that caused 0 byte voicemail files to be written (gsm, wav) which hang up on a caller when checking their voicemail with these files in them? |
| 22:44.55 | ManxPower | [hC]: you ran out of disk space |
| 22:45.22 | [hC] | I have 124GB free |
| 22:45.36 | [hC] | this problem comes up occasionally every couple months |
| 22:45.37 | ManxPower | I've run into that exact problem 3 times, each time the voicemail spool partition was out of space. Then I had to go thru and clean up all the .lock files before voicemail would work |
| 22:45.48 | ManxPower | Ah, then that's not the same issue. |
| 22:46.05 | [hC] | user will wind up with a voicemail file that has a 0 byte gsm file, and 60 byte wav file (empty clearly) and when asterisk tries to play the message, it app_voicemail dies and the user is hung up on |
| 22:46.27 | [hC] | It would be better if asterisk handled it with a soft error, and skipped the message |
| 22:46.38 | ManxPower | I think I recall some bug like that, but I thought it was fixed in 1.2.mumble |
| 22:47.05 | [hC] | this box is on 1.4.14 |
| 22:47.38 | [hC] | I'll upgrade it, but it would be nice to fix the way asterisk handles this type of thing.. I wonder if i could find it in the code. |
| 22:48.45 | ManxPower | [hC]: I'm SURE it's a race condition |
| 22:49.36 | _MrSeb_ | ManxPower: I've tried with can reinvite=no and nat=yes, changing localnet and some other thing, but the registration don't work |
| 22:50.26 | ManxPower | _MrSeb_: I still think you should just fix your network design |
| 22:50.39 | ManxPower | now that you have a real router, there is no reason for Asterisk to act as a router. |
| 22:53.10 | lmadsen | anyone know if it is possible to make GotoIfTime() route to something after a certain date indefiniately |
| 22:53.13 | _MrSeb_ | ManxPower: I can't have an asterisk that work as client for ITSP and as server for internal network? I've found here a person that has a configuration like mine, SteveTotaro, but I've never found again |
| 22:54.11 | ManxPower | _MrSeb_: I don't know. Asterisk is not very good at dealing with multiple IPs, your oddity is that BOTH Asterisk interfaces are NAT'd addresses. I have never ever run into an Asterisk install like that. |
| 22:54.40 | ManxPower | Asterisk is one of those programs where you can fight it's oddities and have a miserable life, or accept it's oddities, work around them, and be healthy, happy, and wise. |
| 22:54.55 | _MrSeb_ | ManxPower: ok, thanks |
| 22:56.44 | d-k-t | NAT=bad |
| 22:57.38 | nick125 | NAT=fun! |
| 22:57.43 | ManxPower | d-k-t: You're wrong, but you can't be blamed for believing the propaganda. |
| 22:57.57 | lmadsen | aha! |
| 22:57.59 | lmadsen | I think I know how to do it |
| 22:58.15 | ManxPower | lmadsen: if anyone knows, it would be you. 8-) |
| 22:58.18 | lmadsen | STRPTIME(...) > ${EPOCH_I_WANT_TO_USE} |
| 22:58.33 | lmadsen | that'd make a good cookbook recipe |
| 22:59.00 | ManxPower | I thought epochs were tough and stringy? |
| 22:59.06 | d-k-t | ManxPower, surely the propaganda is that NAT is good? I don't believe that though... I've had enough headaches caused by it to prove otherwise... |
| 22:59.19 | lmadsen | :) |
| 22:59.23 | ManxPower | Nope. |
| 22:59.40 | lmadsen | no one thinks NAT is good |
| 22:59.48 | d-k-t | NAT is an evil hack |
| 23:00.43 | ManxPower | NAT helps protect people too stupid to have a firewall, it is pretty transparent for almost all common protocols, except those dealing with RTP and other protocols that break the rules and embed address information inside the data payload. |
| 23:01.06 | ManxPower | It simplifies addressing as well, which is what I like. |
| 23:02.03 | [hC] | Does asterisk by default attach wav or wav49 voicemails to email? |
| 23:02.09 | lmadsen | wav |
| 23:02.10 | lmadsen | I think |
| 23:02.17 | lmadsen | check the sample voicemail.conf :) |
| 23:02.24 | ManxPower | [hC]: there is no default |
| 23:02.55 | [hC] | if i have voicemail saving in wav, wav49, and gsm, which of the two wav attachments make it into the email, how's that? :) |
| 23:02.55 | ManxPower | Or are you doing something silly like recoding messages in multiple formats? |
| 23:03.06 | [hC] | i dont even know why that is there |
| 23:03.07 | ManxPower | Yup, you are. |
| 23:03.14 | [hC] | Held over from ages ago im sure. |
| 23:03.21 | ManxPower | We use only WAV49 |
| 23:03.55 | lmadsen | [hC]: I'm pretty sure it uses the first one in the list |
| 23:04.03 | [hC] | and WAV49 is saved as .WAV right, whereas wav is saved as '.wav' .. |
| 23:04.06 | ManxPower | That's what I would assume |
| 23:04.15 | lmadsen | wav49 is like a wav encoded GSM |
| 23:04.19 | [hC] | lmadsen: okay, cool.. Thats what i need to know |
| 23:04.32 | [hC] | im going to record in WAV49 and g729 then. |
| 23:04.39 | lmadsen | basically use wav49 because its small like GSM, but plays on Windows computers without external software |
| 23:04.48 | ManxPower | It's the best of both worlds, the small audio file size of GSM, the Windows media player compat of .wav |
| 23:04.52 | [hC] | precisely. |
| 23:04.53 | lmadsen | :D |
| 23:05.08 | [hC] | Ive already got the functionality, but I dont see any purpose to record 'wav' as well if im not using it. |
| 23:05.12 | ManxPower | of course, converting from one format to another can be....problamatic |
| 23:07.01 | *** join/#asterisk coppice (n=chatzill@11.166.17.210.dyn.pacific.net.hk) |
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| 23:14.25 | delparnel | When I call a DID and get a DISA.. and then dial a number and call out from that DISA.. am I getting charged for both incoming and outgoing that entire call from the ITSP I'm using? |
| 23:15.03 | *** join/#asterisk PepOSX (n=angeldav@200.93.28.168) |
| 23:15.27 | ManxPower | [hC]: remember to do a find for all the now unused files and remove them. |
| 23:15.29 | ManxPower | delparnel: yes, that is 2 calls |
| 23:15.38 | [hC] | ManxPower: already done :) |
| 23:16.15 | ManxPower | [hC]: if any users had their greeting ONLY in one of the removed formats they no longer have a custom greeing |
| 23:17.23 | [hC] | ManxPower: yeah, it was in all three. s'all good. |
| 23:19.52 | ManxPower | [hC]: I switched from GSM to WAV49 and did not research it well enough. |
| 23:20.09 | ManxPower | People had all the GSM files in their mailbox, but they could not listen to them. Asterisk got very upset with me. |
| 23:23.05 | [hC] | ManxPower: haha i'll bet :) I have a number of gotchas like that that I've been bit with.. constantly reminding my staff how asterisk wants things to be .. :) |
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| 23:29.57 | _MrSeb_ | I leave... bye to all |
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| 23:36.30 | skyggen | Oi, anyone around? |
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| 23:38.18 | skyggen | Has anyone had a problem when setting caller id long distance calls get fast busy and 41? |
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