| 00:09.23 | *** join/#asterisk war9407 (war@c-71-62-61-45.hsd1.va.comcast.net) |
| 00:13.38 | *** join/#asterisk WiretapWork (~Wiretap@unaffiliated/wiretap) |
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| 00:36.20 | damasceno | Anyone can help me? I need to know about money to do a service with asterisk. It's a simple server with a simple configuration. |
| 00:39.22 | *** join/#asterisk Cesare (~Adium@creati59.lnk.telstra.net) |
| 00:41.21 | carrar | use bitcoins |
| 00:46.04 | WiretapWork | damasceno, |
| 00:46.06 | WiretapWork | ~ask |
| 00:46.06 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 00:50.46 | *** join/#asterisk jeffspeff (~jeffspeff@173-11-144-149-houston.txt.hfc.comcastbusiness.net) |
| 00:51.34 | jeffspeff | has anybody used a Aastra phone or recommend them or Polycom? |
| 00:52.24 | carrar | Aastra phones work great |
| 00:52.26 | carrar | So fo polycom |
| 00:52.28 | carrar | d |
| 01:00.43 | *** join/#asterisk sam555 (~chatzilla@udp124488uds.hawaiiantel.net) |
| 01:00.47 | sam555 | hello all! |
| 01:01.11 | sam555 | Having trouble recording a basic IVR. |
| 01:01.35 | sam555 | The wav file is not saving (they are empty files - no sound) |
| 01:01.52 | WiretapWork | are you using freepbx? |
| 01:02.01 | sam555 | Also noticing a loop for the *99 menu. I respond with a button push and it's not registering |
| 01:02.09 | sam555 | WiretapWork: yes, freepbx |
| 01:02.16 | WiretapWork | in that case, yes... you want #freepbx |
| 01:02.22 | WiretapWork | #asterisk does not support freepbx installs |
| 01:02.36 | WiretapWork | as they have a buttload of precreated config that is a minefield to work with |
| 01:02.36 | sam555 | WiretapWork: gotcha. sorry |
| 01:02.44 | sam555 | i see |
| 01:02.54 | sam555 | thanks for the forward |
| 01:15.20 | *** join/#asterisk nightrid3r (~kvirc@91.176.172.17) |
| 01:25.48 | p3nguin | damasceno: What's the issue? |
| 01:27.19 | p3nguin | wiretapwork: Do you ever use your WRT54GL as a wireless bridge? I'm wondering about physical configuration. Curious if I disregard the WAN port and turn off the dhcpd in such a scenario. |
| 01:38.04 | WiretapWork | p3nguin, disregard WAN port (set it to be attached to the switch, and on the same VLAN as the rest of the ports), turn off the WAN connection, and set it to operate as a 'client bridge' |
| 01:46.33 | p3nguin | wiretapwork: And I'll have to specify an essid with which it should associate? |
| 01:50.24 | p3nguin | I guess I need to test it out with one of my neighbors' APs before I need to use it in a serious situation. |
| 01:50.35 | WiretapWork | p3nguin, use the site survey tool |
| 01:50.55 | p3nguin | I don't recall seeing that in the GUI. |
| 01:51.01 | WiretapWork | on status page |
| 01:51.07 | WiretapWork | status -> wireless |
| 01:51.12 | WiretapWork | I'm assuming you've put DD-WRT on itg |
| 02:07.55 | *** join/#asterisk saxa (~sasa@189.26.255.43) |
| 02:11.42 | p3nguin | wiretapwork: Once I join the other node as either a client or, as in this case, a client bridge, where do I get to input the security key for that AP? |
| 02:11.57 | WiretapWork | wireless->wireless security |
| 02:12.17 | p3nguin | Oh, so being a client takes over the security page of when it was an AP. Okay. |
| 02:12.30 | p3nguin | I guess that makes sense. |
| 02:13.00 | *** join/#asterisk nightrid3r (~kvirc@91.176.23.232) |
| 02:13.39 | p3nguin | Then any wireless clients that connect through my device have to also use the key from the master AP? |
| 02:14.21 | WIMPy | You try to use it as client and AP at the same time? |
| 02:15.53 | p3nguin | Maybe I am. Let's say my laptop cannot reach the master AP, but I have this wrt54gl here with a good antenna on it which can access the master AP. I'd like to be able to use my laptop on the master through this appliance. Maybe client bridge mode isn't the right mode. |
| 02:16.03 | p3nguin | Maybe repeater mode is what I need? |
| 02:16.19 | p3nguin | I don't know, I've never used dd-wrt before. |
| 02:16.48 | WIMPy | I think repeater sounds right. |
| 02:17.29 | *** join/#asterisk nightrid3r (~kvirc@91.176.23.234) |
| 02:17.37 | WIMPy | That will leave you with <50% speed, off course. |
| 02:17.54 | p3nguin | I'll have to accept that. |
| 02:18.16 | WiretapWork | p3nguin, you can't use a single radio device that is not atheros based to do that properly, other than repeater mode, which the remote end has to support |
| 02:18.55 | p3nguin | Now that I think about it, it sort of does make sense that bridge mode would bridge the wireless interface to the wired ports. I don't know what I was thinking. |
| 02:19.24 | WIMPy | Yes, just client mode and continue via wire, optionally to the next AP would be some sort of better. |
| 02:19.41 | nightrid3r | the fact that you talk in your head doesn't mean your thinking |
| 02:19.47 | p3nguin | What's the difference between client mode and client bridge mode? |
| 02:20.05 | WIMPy | nfi |
| 02:21.14 | WIMPy | Probably client mode uses routing, possibly with nat. |
| 02:21.16 | WiretapWork | p3nguin, one bridges it to the lan ports, the other just acts as client-only, |
| 02:21.54 | p3nguin | Realistically, what good would it do with that appliance being just a client with no other devices connecting to it? |
| 02:22.02 | p3nguin | Or did I misinterpret your statement? |
| 02:22.09 | *** join/#asterisk timahvo1 (~rogue@41.223.57.75) |
| 02:22.21 | WiretapWork | some devices of high power than the WRT54GL might be used as print servers, etc |
| 02:23.15 | p3nguin | So you're suggesting that the client mode isn't necessarily intended for this router appliance? |
| 02:24.19 | p3nguin | Maybe client mode is more of an Ah-hoc type of configuration? |
| 02:24.39 | WiretapWork | no |
| 02:24.55 | WiretapWork | p3nguin, all modes are available on ALL DD-WRT capable devices |
| 02:25.02 | WIMPy | Ad hoc is ad hoc. |
| 02:25.38 | p3nguin | is lost again |
| 02:26.45 | *** join/#asterisk sourcode (~code@ppp-58-8-126-117.revip2.asianet.co.th) |
| 02:30.17 | p3nguin | If I start changing settings to see what does what, I'll get kicked off. |
| 02:33.25 | *** join/#asterisk vinhdizzo (~vinh@pool-173-51-129-161.lsanca.fios.verizon.net) |
| 02:47.58 | p3nguin | wiretapwork: So using this appliance with a Broadcom card, I can use repeater mode... but the master AP has to support repeaters? |
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| 03:24.21 | brack9 | Hey |
| 03:24.36 | brack9 | Anyone know if there's any hosted asterisk out there that runs chan_gtalk for google voice integration? |
| 03:24.46 | brack9 | besides pbxes |
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| 05:37.55 | sawgood | Why is Digium cancelling the Skype-Asterisk plug in coming in July? |
| 05:38.33 | irroot | sawgood its the other way around Skype AKA M$ has pulled the plug |
| 05:38.45 | sawgood | oh ... thanks ... |
| 05:38.52 | sawgood | pulled the plug ... |
| 05:39.06 | sawgood | so, what happens to the people with licenses existing? |
| 05:39.32 | irroot | read the release think they covered |
| 05:39.38 | irroot | but only for while |
| 05:39.46 | sawgood | ok |
| 05:42.07 | *** join/#asterisk ajkaanbal (~ajkaanbal@189.181.110.9) |
| 05:44.33 | p3nguin | Everyone know that Microsoft would discontinue that project as soon as they acquired Skype. |
| 05:45.15 | p3nguin | s/know/knew/ |
| 05:45.16 | sawgood | Well, will there be other ways to bring in the Skype protocol other then the Digium product you think? |
| 05:54.36 | nightrid3r | one possible way is that reverse engineering is legal in russia, once the protocol of sky is known someone could write a module |
| 05:57.06 | irroot | there is a openskype driver of sorts already |
| 05:57.21 | irroot | got to love the russians .... |
| 05:57.29 | nightrid3r | yeah |
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| 06:04.01 | kaldemar | sawgood: there were other ways too before chan_skype existed, but they were icky. google away. |
| 06:06.25 | irroot | was asked the other day if asterisk will have a lync ?? driver ... think it was at this times i showed symptoms of tourettes syndrome |
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| 06:09.35 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 06:09.47 | *** join/#asterisk Ad-Hoc (~nimbus@62.169.216.185) |
| 06:15.29 | sawgood | I've seen several via Google ... |
| 06:17.36 | nightrid3r | the problem is that once we have a good sky channel driver M$ will change the protocol, so basicaly we will end up with a ratrace |
| 06:18.26 | nightrid3r | note to self, replace keyboard |
| 06:18.44 | irroot | or M$ will take the FOSS enhancements for themselves and black box us |
| 06:18.50 | sawgood | Do you think there is any 'advantage' to using Fax for Asterisk when faxes arrive over a SIP trunk to an IP PBX ... and then from the IP PBX to an extension (which is an ATA) hooked directly to a fax machine ... |
| 06:19.09 | sawgood | vs ... just using the SIP trunk to call the ATA directly by passing Asterisk? |
| 06:19.52 | irroot | nightrid3r no letters left on the keyboard ?? worn through a few ... teaches touch typing quickly |
| 06:19.52 | sawgood | Basically, could using Fax for Asterisk 'improve' the changes of a FAX working to a stand alone fax machine? |
| 06:20.12 | sawgood | not changes ... I meant 'chances' |
| 06:20.14 | sawgood | sorry |
| 06:20.59 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 06:21.07 | irroot | sawgood it depends i have used both methods bypassing asterisk to avoid nat / reinvite issues can bennigit |
| 06:21.11 | nightrid3r | does your ata speak t.38 ?? if not it will be impossible to get a fax trough |
| 06:21.25 | sawgood | Yes, the ATA supports T.38 ... for sure |
| 06:21.40 | irroot | shamelessly punts the T38 Gateway patch to res_fax .... |
| 06:21.40 | *** join/#asterisk mandla (~mandla@168.167.180.161) |
| 06:22.08 | sawgood | I have been using the SIP trunk DID number to the ATA ... but faxing is not always working ... breaks on receiving most of the time |
| 06:22.53 | sawgood | I figured maybe if the Fax for Asterisk process was involved rather then just using a T.38 ATA ... then it might improve the changes for faxes to work |
| 06:25.22 | irroot | wonders if doing reviews on RB counts as good deed of the day |
| 06:28.50 | *** join/#asterisk Ad-Hoc (~nimbus@62.169.216.185) |
| 06:29.13 | mandla | irroot, |
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| 06:34.19 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 06:37.57 | irroot | mandla lo there |
| 06:38.06 | irroot | will have info for you in a bit |
| 06:38.15 | mandla | Ok. |
| 06:38.41 | mandla | And the account?? |
| 06:39.01 | irroot | yeah |
| 06:39.36 | irroot | coffee first |
| 06:41.05 | mandla | Ok. |
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| 07:22.42 | *** join/#asterisk caban|2 (~caban@office.rih.ru) |
| 07:28.48 | caban|2 | Why Asterisk does not play musiconhold ? http://paste.org.ru/?hzkhpf |
| 07:28.59 | *** join/#asterisk sgimeno (~sgimeno@163.117.206.10) |
| 07:30.50 | p3nguin | Based on your paste... |
| 07:30.50 | p3nguin | How the hell would we know? |
| 07:30.57 | cneb3000 | caban|2: what does the logs say when it 'doesn't play'? |
| 07:31.05 | cneb3000 | your logs just show us the file directory :) |
| 07:31.12 | cneb3000 | your paste* |
| 07:31.20 | cneb3000 | woke up an hour ago. |
| 07:32.33 | p3nguin | Just because you have music files does not make music play. You have to have phones and stuff. |
| 07:32.42 | caban|2 | http://paste.org.ru/?2yl2bs |
| 07:33.17 | CareBear\ | duh. didn't think about skype for asterisk when I heard MS bought them |
| 07:33.19 | CareBear\ | stupid me |
| 07:33.24 | CareBear\ | of course they shut it down |
| 07:33.44 | cneb3000 | caban|2: Do you know how to log into the AsteriskCLI? |
| 07:33.50 | CareBear\ | bloody f-ing morons :( |
| 07:33.50 | p3nguin | Just wait to see what they do to skype. |
| 07:34.22 | p3nguin | Bing Communicator, here we come! |
| 07:34.22 | CareBear\ | p3nguin : the one hope is that they break it so bad that users will be open for open solutions |
| 07:34.56 | caban|2 | cneb3000: yes |
| 07:35.06 | CareBear\ | but, of course users will not, because they see nothing beyond the brand |
| 07:35.15 | CareBear\ | no |
| 07:35.17 | CareBear\ | ok |
| 07:35.22 | CareBear\ | get a hold of yourself now bear |
| 07:35.30 | CareBear\ | didn't sleep enough and am grumpy |
| 07:35.38 | CareBear\ | like bears are |
| 07:35.42 | cneb3000 | caban|2: access the CLI, and type 'core set verbose 5' (to make sure logging is turned on). put a phone on hold, and copy and paste what the CLI outputs. |
| 07:35.50 | p3nguin | Don't be grumpy because of that... |
| 07:36.13 | irroot | sends carebear |
| 07:36.20 | p3nguin | Be grumpy because people DESERVE IT! |
| 07:36.25 | irroot | sends carebear to redmond spread some cheer |
| 07:36.40 | *** join/#asterisk MDesade (~chatzilla@ip24-251-93-137.ph.ph.cox.net) |
| 07:36.56 | CareBear\ | p3nguin : is true, but then I would get depressed and mad because so many deserve it :) |
| 07:37.33 | caban|2 | http://paste.org.ru/?gvbofc |
| 07:37.50 | CareBear\ | ok, back op topic: |
| 07:38.41 | CareBear\ | how can I debug why my asterisk ends up with State Failed for a registration that works perfectly if I kick it with sip reload ? |
| 07:40.21 | cneb3000 | caban|2: it's trying to translate the codec between g729 anxd slin. |
| 07:41.02 | cneb3000 | try converting the source files to different file types, put them in that folder and try again. |
| 07:41.04 | cneb3000 | you use... |
| 07:41.38 | cneb3000 | Sox: http://www.voip-info.org/tiki-index.php?page=Convert+WAV+audio+files+for+use+in+Asterisk or.. |
| 07:41.57 | cneb3000 | http://www.digium.com/en/products/ivr/audio-converter.php if you're not up to the linux side of things |
| 07:42.30 | cneb3000 | in that logs case. if you had a slin file there, it wouldn't have to translate, and shoudl just work,. |
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| 07:56.39 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 07:58.20 | *** join/#asterisk ChannelZ (channelz@burner.com) |
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| 08:04.49 | *** join/#asterisk Stese (~Stese@adsl.ntsols.com) |
| 08:04.52 | Stese | Hey all |
| 08:05.19 | *** join/#asterisk salimb (~3laz3r@109.231.227.100) |
| 08:08.31 | Stese | Can someone look @ this Pastebin, and tell me why, on Priority 9, the wrong value is set? specifically, it's sets the value in the astDB to 1, instead of 0. it also does this on Priority 15... there it will set the valie to 0, instead of 1. |
| 08:08.34 | Stese | http://pastebin.com/QtsM7QuL |
| 08:17.57 | ChannelZ | line 18 - Set(NSStatusSet=1) |
| 08:18.11 | ChannelZ | and similar on line 11 |
| 08:18.27 | kaii | # |
| 08:18.30 | kaii | exten => s, 8, Set(${NSStatusSet}=0) |
| 08:18.47 | mandla | Hey irroot , i have some cable engineers here, they have configured a crossover cable. |
| 08:18.57 | Stese | I spotted it myself a few moments ago... |
| 08:18.59 | kaii | results it Set(1=0) or Set(0=0), depending on which value you have read from the DB |
| 08:19.45 | Stese | I was trying to set the variable name, instead of the actual variable... |
| 08:20.03 | Stese | slaps himself with a wet, slimy trout |
| 08:20.17 | ChannelZ | I'm glad we could all have this talk. |
| 08:22.17 | *** join/#asterisk mzahariev (adminimini@piem-vsichko-v.unixsol.org) |
| 08:24.39 | *** join/#asterisk hetii (~Grzegorz@194.181.154.25) |
| 08:24.42 | hetii | hello :) |
| 08:25.22 | ChannelZ | ohell |
| 08:25.46 | hetii | Q: did someone of you know how to do it when i send fax to hylafax that he instead of process it by some modem (to send him to destination) will send postcript file to some mail |
| 08:29.07 | kaii | Stese: to be correct, you tried to use the actual _value_ of your variable as the variables name for Set()ing a value. |
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| 08:43.06 | *** part/#asterisk kriegerod (~krieger@user-204.45.infomir.com.ua) |
| 08:53.14 | *** join/#asterisk jacc0 (~jacco@D522448D.static.ziggozakelijk.nl) |
| 08:53.23 | jacc0 | hi all :0 |
| 08:54.05 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 08:54.07 | schmidts | hello |
| 08:54.15 | jacc0 | hi |
| 08:55.48 | *** join/#asterisk frawd (~francois@115.Red-81-44-76.dynamicIP.rima-tde.net) |
| 09:09.18 | Stese | I'm beginning to hate the way Variables are handled in Asterisk |
| 09:10.28 | *** join/#asterisk bluOxigen (~noreply@unaffiliated/bluOxigen) |
| 09:10.31 | Stese | I'm using GotoIF.... checking if a value is 0. it IS 0, so it's going, it's false... drop to the second destination |
| 09:10.40 | Stese | arg! |
| 09:10.52 | kaldemar | Stese: what's the problme? |
| 09:12.36 | Stese | Pastebin incoming |
| 09:12.51 | hetii | can be tbat 0 and false are intepreted on the same way :) |
| 09:12.58 | hetii | *that |
| 09:13.06 | Stese | http://pastebin.com/Dw4hJY2f |
| 09:13.23 | *** join/#asterisk e7e5 (~rudenko@188.134.2.33) |
| 09:13.29 | Stese | Hetii > possibly... would be nice for the CLI to say thats happened |
| 09:13.31 | kaldemar | Stese: GotoIf(${NSStatusChk}=0?vmoff:vmon) is missing $[]. |
| 09:14.06 | kaldemar | $[] is needed to evaluate an expression. so, use GotoIf($[${NSStatusChk} = 0]?vmoff:vmon) |
| 09:16.42 | Stese | <PROTECTED> |
| 09:16.52 | Stese | I'm getting there... |
| 09:16.53 | Stese | :P |
| 09:16.59 | Stese | Thanks Kal |
| 09:20.06 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 09:22.13 | *** join/#asterisk Gugge (~gugge@91.208.16.1) |
| 09:28.34 | Stese | Anyone know if it's possible to program buttons on a Linksys SPA922? |
| 09:30.42 | irroot | Stese define program .... |
| 09:31.06 | irroot | BLF is not enabled |
| 09:31.20 | irroot | we set them up as line keys |
| 09:31.42 | irroot | so not "programed" |
| 09:31.52 | Stese | Set them to do something else, other than what they already to... I'm thinking the 4 soft keys. |
| 09:32.11 | irroot | yeah |
| 09:32.23 | Stese | basically, use one of them to dial a number, instead of its default action |
| 09:34.11 | wdoekes2 | the 922? not 942? |
| 09:35.00 | Stese | 922 |
| 09:36.41 | wdoekes2 | I don't think you can do anything with those soft buttons.. on the 942 you can reuse the line-keys (green buttons on the right) for some purposes (e.g. blf / quick dialing) |
| 09:37.16 | Stese | ok, no worries. They'll just have to dial the code then, instead of pushing a single button. |
| 09:37.28 | *** join/#asterisk timahvo1 (~rogue@41.223.57.74) |
| 09:38.42 | wdoekes2 | you can use the "speed dial" |
| 09:38.57 | wdoekes2 | 2, Dial |
| 09:38.58 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 09:39.05 | wdoekes2 | that's two presses |
| 09:39.40 | *** join/#asterisk hrolf (~hrolf@202.61.49.9) |
| 09:41.56 | *** join/#asterisk Stese (~Stese@adsl.ntsols.com) |
| 09:42.01 | Stese | grr |
| 09:47.57 | *** join/#asterisk sameerynho (~lxsameer@178.131.121.181) |
| 09:48.36 | sameerynho | hi, i want to build an ACD , what software do i need? does a simple fax modem in enough |
| 09:50.35 | *** join/#asterisk bluOxigen (~noreply@unaffiliated/bluOxigen) |
| 09:55.54 | sameerynho | what kind of hardware do i need for running a ACD ? |
| 09:57.56 | Stese | What do you mean, by ACD? |
| 09:58.18 | sameerynho | Stese, Automatic Call Distributors |
| 09:58.40 | sameerynho | Stese, i want auto answering too |
| 09:58.47 | Stese | ok. then, not really my field sorry |
| 09:59.03 | *** join/#asterisk vfabi (~fabi@194.247.164.231) |
| 09:59.10 | irroot | sameerynho app_queue |
| 09:59.15 | irroot | ~thebook |
| 09:59.15 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 09:59.48 | sameerynho | irroot, thanks, i just want to order hardware |
| 10:00.00 | sameerynho | irroot, after that i'll surly read the book |
| 10:00.27 | sameerynho | irroot, but i wonder if i can do this with just a normal fax modem |
| 10:00.33 | kaldemar | sameerynho: a fax modem is useless. |
| 10:01.22 | sameerynho | kaldemar, aha, what do you suggest , i'm newbie to this, i just want to create an ACD for an small business |
| 10:01.39 | kaldemar | depends on how you want to connect and where. |
| 10:02.06 | sameerynho | kaldemar, do you have any ref for me to begin with |
| 10:02.23 | kaldemar | you already got one. read the book. |
| 10:02.26 | Chainsaw | sameerynho: You will need an actual telephony adapter. Were you planning to save some money by using a soldering iron and a stack of old modems? |
| 10:03.21 | sameerynho | kaldemar, ok thanks |
| 10:03.36 | *** join/#asterisk moodyy (~chatzilla@estrela-adm.nortenet.pt) |
| 10:03.45 | sameerynho | Chainsaw, yeah i want to save money but not for strange stuff |
| 10:04.19 | Chainsaw | sameerynho: First things first. These calls you're wanting to distribute. How do they come in now? |
| 10:04.42 | sameerynho | Chainsaw, phone line |
| 10:04.53 | Chainsaw | sameerynho: Well I guessed that much, yes. What *kind* of phone line? |
| 10:04.54 | *** join/#asterisk Stese (~Stese@adsl.ntsols.com) |
| 10:05.38 | Stese | Is that book the successor to Asterisk ~ the future of technology? |
| 10:05.47 | Stese | *telephony even |
| 10:06.43 | sameerynho | Chainsaw, i don't know about it, how can i found out? |
| 10:06.54 | kaldemar | Stese: yes, written for asterisk 1.8. |
| 10:07.29 | Chainsaw | sameerynho: Well, how many calls can go through that line at once? Just 1, or more than one? Do you have internal extensions? |
| 10:08.41 | sameerynho | Chainsaw, normally just one, but i think there some way to ask for more calls in an instance, internal extensions?\ |
| 10:10.20 | Chainsaw | sameerynho: Yes. Is there more than one phone in the building? |
| 10:11.04 | *** join/#asterisk sourcode (~code@ppp-58-8-126-117.revip2.asianet.co.th) |
| 10:12.11 | sameerynho | Chainsaw, more than one node , and 2 phone line |
| 10:13.10 | Chainsaw | sameerynho: Those lines are analog or digital. Do you know which? |
| 10:13.13 | mandla | irroot, are we still on?? |
| 10:14.23 | sameerynho | Chainsaw, analog i think , |
| 10:14.41 | Chainsaw | sameerynho: I need you to know, not guess. |
| 10:14.49 | Chainsaw | sameerynho: Go forth and investigate. |
| 10:16.04 | *** join/#asterisk Gugge (~gugge@91.208.16.1) |
| 10:16.18 | sameerynho | Chainsaw, they are analog |
| 10:16.59 | Chainsaw | sameerynho: Okay. Do those two lines go into a PBX of some kind before they reach the phones? Or straight in? |
| 10:17.50 | *** join/#asterisk MarKsaitis (~MarKsaiti@host81-137-245-117.in-addr.btopenworld.com) |
| 10:18.35 | sameerynho | Chainsaw, straight into phone |
| 10:19.19 | Chainsaw | sameerynho: Okay. So you need a card with 2 FXO ports to accept those two analog lines. 2 FXS ports if you want to connect those two analog phones and do something intelligent in between. |
| 10:20.17 | Chainsaw | sameerynho: Something like a TDM410 should do nicely, 2 FXO modules, 2 FXS modules. |
| 10:20.42 | sameerynho | Chainsaw, aha thanks a lot my friend |
| 10:21.21 | Chainsaw | sameerynho: Now, the book that was recommended to you... |
| 10:21.31 | Chainsaw | sameerynho: I would recommend that you read it to get more familiar with the terminology in use here. |
| 10:21.35 | sameerynho | Chainsaw, yeah sure |
| 10:22.12 | moodyy | Hi all, is there a way to send/receive ISDN user-to-user info in asterisk? And if exists whats is the best one? |
| 10:23.16 | WIMPy | moodyy: I don't know any. |
| 10:24.44 | moodyy | in asterisk 1.4 there is one reference to that in chan_zap, but in asterisk 1.8 i can't find any info... |
| 10:24.55 | Chainsaw | moodyy: That's because "Zaptel" is now "DAHDI". |
| 10:25.24 | *** join/#asterisk timahvo1 (~rogue@41.223.57.72) |
| 10:29.25 | moodyy | yes i know that, but i can't find any info about this in dahdi |
| 10:35.49 | sameerynho | Chainsaw, TDM410 is $410, is there any thing cheaper, its very expensive for me to buy |
| 10:37.35 | Chainsaw | sameerynho: You could check for an "ATA" with 2 FXO ports and 2 FXS ports, but I doubt there will be many devices that are cheaper. |
| 10:38.35 | sameerynho | Chainsaw, so for running such system i have to spend about 500$ , am i right |
| 10:38.35 | WIMPy | Analog is not only evil, its expensive, too. |
| 10:39.11 | Chainsaw | sameerynho: Sounds about right, yes. |
| 10:39.12 | sameerynho | WIMPy, does the digital lines solution is cheaper |
| 10:39.17 | sameerynho | Chainsaw, :( |
| 10:39.42 | WIMPy | sameerynho: Hardwarewise yes. But that mainly depends on your telco. |
| 10:40.00 | sameerynho | WIMPy, how much it cost? |
| 10:41.45 | WIMPy | sameerynho: A BRI card is around 20 bucks. But if you're in the US you probably don;t want to know the price of the line. |
| 10:42.35 | sameerynho | WIMPy, how can i make sure about my line type ? |
| 10:44.36 | sameerynho | Chainsaw, it seems that we can buy digital lines , |
| 10:44.50 | sameerynho | Chainsaw, not what hardware do you suggest ? |
| 10:45.05 | Chainsaw | sameerynho: I shall transfer this call to WIMPy. |
| 10:45.18 | sameerynho | Chainsaw, thanks for your time man |
| 10:45.23 | Chainsaw | sameerynho: You're welcome. |
| 10:46.39 | WIMPy | sameerynho: You need to check what your telco(s) offer. |
| 10:46.44 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 10:47.33 | sameerynho | WIMPy, i ask them just now, we can buy digital line for low price |
| 10:47.48 | sameerynho | WIMPy, now what hardware do you suggest? we have two line |
| 10:48.35 | WIMPy | sameerynho: A BRI has two channels. You can get that in to an HFC-S card. |
| 10:49.36 | sameerynho | WIMPy, ?? i don't get it, can you be more clear please :P |
| 10:50.13 | WIMPy | What exactely do you want to know? |
| 10:50.54 | sameerynho | WIMPy, just a hardware for two digital line and running the ACD, i want to order the hardware and then read the astrisk book |
| 10:52.16 | WIMPy | As I said: One line has two channels. And for the hardware look for any el cheapo ISDN card, they usually have HFC-S chips that are used by many Asterisk users. |
| 10:53.05 | sameerynho | WIMPy, aha , thanks a lot man |
| 10:54.01 | WIMPy | sameerynho: If I got that right, you want to connect two phones. Just use IP phones for them. |
| 10:55.32 | sameerynho | WIMPy, i want to build a software for auto answering and some intelligence work on those two lines, also i want to transfer calls |
| 10:56.30 | WIMPy | Transfer out the same line again? |
| 10:56.51 | sameerynho | WIMPy, to a local phone somewhere else in building |
| 10:56.59 | WIMPy | ok |
| 10:57.30 | sameerynho | WIMPy, sorry but could you please give me the link of that 20$ card? |
| 10:58.28 | WIMPy | sameerynho: I don't know what suppliers you have in your area. But you can always find them on ebay. |
| 10:59.41 | *** join/#asterisk engrxyz (~fsgsfg@212.23.51.7) |
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| 11:13.25 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 11:27.04 | sameerynho | WIMPy, hmm i mixed up man, i can't found out any suitable hardware |
| 11:28.26 | *** join/#asterisk irroot (~irroot@pbx.distrotech.co.za) |
| 11:29.12 | WIMPy | For example 130536055016 on ebay. But you should be able to find them in any electronics/PC parts store, |
| 11:30.36 | sameerynho | WIMPy, i can not buy from ebay , my country is in ebay black list , how much does it cost from a stock market ? |
| 11:30.40 | WIMPy | They probably won't tell you the chip, but all cheap ones should fit. If in doubt you can look at pictures and see if you can find the Cologne Cathedral on the chip. |
| 11:31.11 | WIMPy | I once bought one here in town for 28 EUR. |
| 11:31.55 | sameerynho | WIMPy, aha thanks |
| 11:32.58 | *** join/#asterisk imox1234 (~imox1234@88.240.98.160) |
| 11:34.13 | sameerynho | WIMPy, i have one question about simple modems, can i use a simple modem for autoanswering ? |
| 11:34.32 | WIMPy | Modem? |
| 11:34.42 | WIMPy | Where does that fit in? |
| 11:35.23 | sameerynho | WIMPy, hmm i don't know, connecting the line to modem, |
| 11:35.54 | WIMPy | What do you want to do with that modem? |
| 11:36.45 | sameerynho | WIMPy, receiving calls and play some sound for them |
| 11:37.14 | WIMPy | Not with Asterisk. |
| 11:37.59 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 11:47.59 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 11:50.34 | *** join/#asterisk bluOxigen (~noreply@unaffiliated/bluOxigen) |
| 11:52.16 | sameerynho | WIMPy, its un related but, is there any thing like astrisk for my goal ( using modems)? |
| 11:52.47 | WIMPy | To just play announcements or to do more? |
| 11:53.06 | sameerynho | WIMPy, a little more |
| 11:55.02 | WIMPy | Some IVR menu is possible, but anything beyond probably not. |
| 11:56.35 | sameerynho | WIMPy, IVR? |
| 11:57.07 | WIMPy | A menu with DTMF selection. |
| 11:57.45 | sameerynho | WIMPy, then how to transfer the call after selecting a menu ? |
| 11:59.12 | *** join/#asterisk billmania (~bill@38.98.130.98) |
| 11:59.17 | WIMPy | That's where you leave the easy part. You need to do that via your PBX somehow. But I wouldn't bother the effort. |
| 12:00.01 | sameerynho | WIMPy, thanks a lot man |
| 12:00.12 | *** join/#asterisk fenlander (~fenlander@82.152.81.57) |
| 12:01.03 | WIMPy | If you have the possibility to go the digital way, do it. |
| 12:02.05 | sameerynho | WIMPy, at this point i do not have much money |
| 12:02.18 | sameerynho | WIMPy, those ISDN cards it very expensive in my country |
| 12:02.28 | sameerynho | WIMPy, something like 180$ |
| 12:03.16 | WIMPy | Then take a look at other shops that will ship to your country. Maybe Amazon? |
| 12:03.31 | WIMPy | And 180$ to do exactely what? |
| 12:04.16 | sameerynho | WIMPy, i live in Iran, 180$ to buy the simplest ISDN card |
| 12:04.55 | *** join/#asterisk wwgd (~wwgd@208.79.14.130) |
| 12:05.42 | WIMPy | Ok, pretty remote for most of us, but I'm sure some shop will ship there. But it might indeed be hard to find one. |
| 12:06.27 | sameerynho | WIMPy, yeah , any way thanks for your time man |
| 12:11.42 | *** join/#asterisk pc-m (~pascal@modemcable094.94-70-69.static.videotron.ca) |
| 12:12.28 | WIMPy | sameerynho: Can you buy from Amazon? |
| 12:12.53 | sameerynho | WIMPy, no |
| 12:15.10 | irroot | sameerynho try www.miro.co.za |
| 12:15.38 | irroot | we not so uptight about iran possibly be able to ship it there |
| 12:15.42 | *** join/#asterisk timahvo1 (~rogue@41.223.57.75) |
| 12:16.08 | sameerynho | irroot, thanks man |
| 12:16.49 | irroot | im not @ miro they are a voip supplier in south africa and should be free from the stupid EU/US sanctions |
| 12:18.04 | WIMPy | EU as well? |
| 12:18.24 | sameerynho | hmm its hard to live in sanctions |
| 12:18.29 | WIMPy | hates the US assimilation of the EU. |
| 12:18.31 | sameerynho | with |
| 12:18.39 | *** join/#asterisk fenlander (~fenlander@82.152.81.57) |
| 12:19.19 | irroot | knows how it feels and knows there is always a way arround it |
| 12:22.55 | *** join/#asterisk ariel_ (~chatzilla@pdpc/supporter/active/abatista) |
| 12:24.31 | WIMPy | There doesn't seem to be a ban on export oF IT Stuff. |
| 12:25.13 | *** join/#asterisk Footman (~Footman@gwdev.creape.unilim.fr) |
| 12:25.21 | Footman | hello |
| 12:25.34 | Chainsaw | Good afternoon Footman. |
| 12:25.46 | *** join/#asterisk WiretapSeven (~Wiretap@unaffiliated/wiretap) |
| 12:26.43 | Footman | I installed asterisk-1.8 and asterisk-dahdi on Debian Squeeze (instructions on the wiki : https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages) |
| 12:27.09 | *** join/#asterisk acidfoo (~nib@modemcable094.94-70-69.static.videotron.ca) |
| 12:27.48 | Footman | but dahdi doesn't work : module dahdi not found |
| 12:29.03 | Footman | I tried this : http://cryptwizard.info/?p=691 but dahdi-source doesn't install properly (seems missing files) |
| 12:29.11 | Footman | is it a known bug ? |
| 12:31.02 | beek | Footman: Not enough information. Are you compiling from source or is this a prepackaged installation? |
| 12:32.07 | *** join/#asterisk ParanoyaM (~ParanoyaM@35-18-94-178.pool.ukrtel.net) |
| 12:32.12 | ParanoyaM | hi |
| 12:32.33 | ParanoyaM | Does anybody know is it possible to remove peer from routing if it is offline? |
| 12:32.59 | Footman | beek: I install the packages from asterisk repositories |
| 12:33.14 | *** join/#asterisk pk1r (~robin@D57D44FA.static.ziggozakelijk.nl) |
| 12:34.50 | pk1r | hello there. can i just shoot? |
| 12:34.52 | *** join/#asterisk fhmiv (~fhmiv@c-67-173-205-151.hsd1.ga.comcast.net) |
| 12:35.59 | pk1r | guess so... |
| 12:36.57 | pk1r | anyway, having trouble with IAX in combination with SIP and one way jitter |
| 12:37.31 | pk1r | same boxes using zap, no jitter at all |
| 12:38.25 | beek | Footman: Are you sure that the DAHDI package you installed is for the kernel you are running? |
| 12:39.17 | pk1r | wireshark tells me a 43670.73 ms of jitter... |
| 12:40.54 | Footman | There is no information about kernel in the wiki, just the version of Debian. I use 2.6.32-5-686 |
| 12:41.09 | pk1r | anybody familiar with this kind of problems? |
| 12:42.11 | beek | I'm sorry Footman but I'm not going to be much help. I compile my own Asterisk/DAHDI so I'm sure what I'm looking at. |
| 12:42.56 | beek | You need to use the apt tool to find where it loaded modules for DAHDI and ensure that they are indeed for your running kernel. |
| 12:44.08 | Footman | apt doesn't install modules :/ |
| 12:45.45 | *** join/#asterisk fish-bulb (~qcstewart@nat/digium/x-tjvzkrknrnnfmuxw) |
| 12:46.01 | beek | Footman: You did start DAHDI, right? |
| 12:46.06 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 12:46.09 | beek | /etc/init.d/dahdi start |
| 12:46.29 | beek | It needs to be started prior to starting Asterisk. |
| 12:46.50 | Footman | if it's easier to install from sources, create packages doesn't make sense ;) |
| 12:47.05 | Footman | I think I'll install from tarball |
| 12:47.17 | beek | Well, they're alleged to be easier to maintain. I'm just used to loading from source. |
| 12:47.32 | beek | Did you see my question regarding starting DAHDI? |
| 12:47.45 | pk1r | it only happens with about 8 or more sim calls |
| 12:49.16 | Footman | beek: that's the problem starting DAHDI say "FATAL: Module dahdi not found." |
| 12:50.01 | Footman | that's why I installed dahdi-source package and do make install on it but it doesn't install successfully because some files missing |
| 12:51.12 | puzzled | hi |
| 12:53.08 | pk1r | ok, probably more info is needed... |
| 12:53.45 | pk1r | i can pastebin iax settings if that would help? |
| 13:04.30 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-meeiflrjbidxjwcj) |
| 13:05.41 | beek | Sorry Footman -- I was on the phone. |
| 13:05.54 | beek | So you're installing DAHDI from source? |
| 13:06.24 | Footman | no problem beek ;) |
| 13:07.18 | beek | Footman: I don't want to insult your intelligence.... are you familiar with the process of compiling from source? |
| 13:07.29 | Footman | yes and no : I install the package dahdi-source and I install it from /usr/src |
| 13:08.18 | beek | Footman: When you do the 'make' step, what errors are you seeing? |
| 13:08.53 | Footman | but there is missing files from this source (I just see these files are present on the real source on the site) |
| 13:09.48 | Footman | so, I'll reinstall all from sources :) |
| 13:09.55 | beek | Let's start again... rm -rf /usr/src/dahdi-linux-<version> |
| 13:10.03 | Footman | yes, I know, make, make install, ... |
| 13:10.25 | beek | curl -O http://downloads.asterisk.org/pub/telephony/dahdi-linux/dahdi-linux-current.tar.gz |
| 13:10.39 | beek | That'll ensure that you're getting the right tar file. |
| 13:16.49 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-wpoishlxnltetyuv) |
| 13:19.54 | *** join/#asterisk whirlpbx (~whirlpbx@212.58.196.250) |
| 13:20.13 | Katty | morning |
| 13:20.30 | irroot | hehe one manager wants to know what i did to fix something told him its way too complex for his mind ... but if he brings his pet monkeys into office i can explain it too them ... |
| 13:20.38 | irroot | morning there fair maiden\ |
| 13:22.25 | beek | hugs Katty |
| 13:24.06 | whirlpbx | Hello, my voip provider allows multiple input/output calls from one sip account. Is there a way in * to restrict number of sumultanious calls on sip account to one call? |
| 13:27.10 | *** join/#asterisk scubes13 (~scubes13@cpe-024-088-120-157.sc.res.rr.com) |
| 13:30.28 | l2trace99 | call-limit=1 on peer config |
| 13:31.11 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 13:31.24 | jaytee | Happy Tau Day everyone! |
| 13:31.42 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 13:32.06 | *** join/#asterisk sourcode (~code@ppp-58-8-126-117.revip2.asianet.co.th) |
| 13:32.24 | WIMPy | What's that? |
| 13:32.30 | irroot | http://tauday.com/ |
| 13:32.53 | irroot | We begin repairing the damage wrought by Ï by first understanding the notorious number itself. |
| 13:33.42 | irroot | Buy a Tau Day shirt. Tell the world that the Ï is a lie! |
| 13:37.33 | jaytee | personally I like pi but I'd be all for tau if it means I get twice as much pie. Like to have a scoop of vanilla on the side too! :-) |
| 13:39.00 | irroot | vanilla ^ 2 ?? |
| 13:39.58 | beek | Mornin' jaytee |
| 13:40.01 | pigpen | Hi all. Is there any way to strip a leading or tailing character (such as &) from a variable? |
| 13:40.32 | jaytee | mornin beek |
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| 13:41.33 | pigpen | sed in the dialplan sure would be nice |
| 13:41.50 | irroot | s/// |
| 13:41.59 | *** join/#asterisk timahvo1 (~rogue@41.223.57.77) |
| 13:42.16 | pigpen | in the dialplan? |
| 13:42.20 | kaldemar | pigpen: func SHELL |
| 13:42.35 | pigpen | hmm.... |
| 13:43.00 | kaldemar | or functions REPLACE, FILTER... |
| 13:43.10 | pigpen | so I could take a variable, push it through SHELL (sed, whatever) and kick it out to a new variable? |
| 13:43.40 | kaldemar | or the same variable. SHELL returns stdout of a command. |
| 13:43.40 | pigpen | hmmm yeah REPLACE in 1.8 was getting close, but I haven't looked at FILTER |
| 13:43.49 | pigpen | k. |
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| 13:53.31 | *** join/#asterisk mandla (~mandla@168.167.180.161) |
| 13:54.18 | mandla | irroot, thanx for that. |
| 13:54.37 | irroot | mandla cool let me know if you have issues |
| 13:54.49 | mandla | irroot, mmh, where exactly do i specify them in Asterisks and where in X-lite?? |
| 13:55.53 | irroot | mandla sip.conf in x-lite the sip settings in the profile sugest you try x-lite first also make sure stun is enabled can use same ip as stun server |
| 14:00.01 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
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| 14:01.19 | mandla | irroot, so instead of using SIP setting to my asterisk server i use the SIP settings you gave me?? |
| 14:01.37 | irroot | yeah |
| 14:01.39 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-eborprpisdpbnsre) |
| 14:02.27 | ixyd_ | hi guys, can you tell me the reason for 1.8.5 not being released while all according bugs seem to be closed on https://issues.asterisk.org/roadmap_page.php ? |
| 14:03.00 | *** join/#asterisk apten (~apten@carbon.gonicus.de) |
| 14:04.06 | mandla | irroot, so i will be connecting directly to the VOIP provider you gave me bypassing the asterisk server?? |
| 14:04.17 | irroot | Yes |
| 14:05.21 | pabelanger | ixyd_: RCs should be out this week. There were some security issue that needed to be resolved first |
| 14:05.42 | mandla | So that means i dnt really have to do anything to the sip.conf in asterisk... |
| 14:05.58 | ixyd_ | pabelanger: thx! |
| 14:05.59 | irroot | no test with x-lite first |
| 14:06.41 | mandla | ok |
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| 14:13.18 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 14:16.57 | dlublink | I have an inbound SIP call being routed to an another SIP device. The second device returns "404" so asterisk returns "503", I want asterisk to return busy instead. How can I do this ? |
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| 14:23.21 | Katty | hugs beek |
| 14:26.30 | beardy | Katty! |
| 14:28.35 | Footman | beek: I just reinstalled my OS, installed dahdi, libpri and asterisk from source tarball and all is OK now :) thanks for help ! |
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| 14:31.42 | Footman | beek: do you know where I could notify my bug with the debian packages ? |
| 14:34.56 | beardy | Footman: http://www.debian.org/Bugs/ if that's what you mean. Install "reportbug". |
| 14:35.54 | Footman | no, I think bug comes from Asterisk |
| 14:36.20 | Footman | this is Asterisk repositories for Debian and Ubuntu |
| 14:37.16 | beardy | Oh. |
| 14:38.09 | beek | Footman: Glad things worked for you. I don't know where you can report your problems as I use CentOS for my installations. |
| 14:38.37 | Footman | ok, thanks :) |
| 14:43.46 | tzafrir | Footman, what Debian packages specifically? |
| 14:44.15 | tzafrir | ah, those |
| 14:44.59 | *** join/#asterisk timahvo1 (~rogue@41.223.57.76) |
| 14:45.59 | Footman | tzafrir: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages#AsteriskPackages-APT%28Debian%2FUbuntu%29 |
| 14:59.24 | mandla | irroot, hey man, X-lite shows that it has registered with the servers but the problem is that i cant call |
| 14:59.41 | mandla | irroot, tilililili sound. |
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| 15:39.59 | Katty | stretches |
| 15:40.33 | _jdccdevel | Hello! Is it possible, using any combination of ForkCDR, NoCDR, and ResetCDR, to record the CDR at a particular point in the dialplan, and not generate a CDR for anything that happens after? I'm Thinking a ForkCDR(Av) Followed by a NoCDR() ? |
| 15:45.54 | *** join/#asterisk jkroon (~jkroon@dsl-240-137-67.telkomadsl.co.za) |
| 15:47.50 | beardy | Katty: Where have you been hiding? |
| 15:49.15 | *** join/#asterisk usc911 (~ben@78-105-116-233.zone3.bethere.co.uk) |
| 15:49.23 | usc911 | heya guys, I have an odd issue. New install of trixbox I did exactly the same as last time, calling out works fine, but if you call in and I answer it the persona calling in just keeps on the ringing tone |
| 15:49.46 | *** part/#asterisk apten (~apten@carbon.gonicus.de) |
| 15:49.47 | usc911 | I have asked in the #trixbox room but there were more people in here so thought I'd put the question forward |
| 15:49.57 | beardy | usc911: #trixbox |
| 15:50.23 | irroot | usc911 you got backup of configs ?? #trixbox doubt many here use |
| 15:50.37 | usc911 | nah I havnt |
| 15:50.46 | usc911 | its no biggy Ill keep playing about |
| 15:51.34 | beardy | Why not play with something that has more people supporting it instead if you're only playing? |
| 15:52.11 | usc911 | just always used trixbox really, do you have any suggestion of an alternative? |
| 15:52.45 | p3nguin | usc911: You know we don't support trixbox here, right? |
| 15:52.48 | beardy | ... |
| 15:52.59 | usc911 | yea sorry man, just thought id ask |
| 15:53.09 | beardy | usc911: This is #asterisk, Asterisk is suggested. |
| 15:53.26 | usc911 | ok man I get it |
| 15:54.10 | SunTsu | usc911: and, btw. irc doesn't have rooms, it has channels |
| 15:54.13 | p3nguin | This is, after all, an ASTERISK channel... for people who use ASTERISK. |
| 15:55.39 | usc911 | jeez simmer, I just put the question forward on the off chance that someone may have encountered it, a simple no would have sufficed. |
| 15:56.58 | p3nguin | Alright... now that we have that out of the way, do you have any Asterisk questions? |
| 15:57.54 | usc911 | not currently, but this is the right place to come for asterisk questions in future right? |
| 15:58.17 | p3nguin | Yes, this Asterisk channel is the right place to ask questions about Asterisk. |
| 15:58.36 | usc911 | so this chanel for asterisk, trixbox chanel for trixbox? |
| 15:59.14 | p3nguin | I can't say anything about any trixbox channels because I don't know what they do there. |
| 15:59.35 | p3nguin | I've heard they don't do very much of anything there. |
| 15:59.37 | usc911 | hang out talking about trixbox I assume |
| 16:00.45 | usc911 | share lolcats maybe |
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| 16:01.47 | SunTsu | remember, the actual meaning of IRC is "Idler Relay Community" |
| 16:03.05 | ariel_ | Ok quick qestion. In the dial plan we can script the first digit with {EXTEN:1} or pick how many digits we want to read from the beginning like {EXTEN:0:5} But what if we just need to strip the last digit? |
| 16:03.19 | ariel_ | s/qestion/question/ |
| 16:03.43 | p3nguin | Did you try EXTEN:-1 ? |
| 16:03.59 | *** part/#asterisk usc911 (~ben@78-105-116-233.zone3.bethere.co.uk) |
| 16:04.22 | ariel_ | ${foo:offset:length} |
| 16:04.35 | ariel_ | maybe that would work let me see |
| 16:05.09 | ariel_ | would it not be {EXTEN:0:-1} |
| 16:05.27 | p3nguin | Try both ways and report back the results. |
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| 16:31.37 | ariel_ | those do not work |
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| 16:33.18 | p3nguin | Oh, EXTEN:-1 will show you the last character. |
| 16:33.41 | p3nguin | What are you doing that you want to strip off the last character? |
| 16:34.20 | ariel_ | I need to so a database look up for matching numbers |
| 16:34.37 | ssureshot | are DBget and DBput still included ? or have those been removed? |
| 16:34.38 | ariel_ | and sometimes I get 6 digits sometimes only 4 |
| 16:34.45 | p3nguin | phone numbers? |
| 16:35.07 | ariel_ | yes |
| 16:35.08 | p3nguin | ssureshot: Use DB(). |
| 16:35.09 | ariel_ | numbers |
| 16:35.16 | irroot | is listening |
| 16:35.18 | p3nguin | Variable length phone numbers? |
| 16:35.32 | ariel_ | it's not aways the same |
| 16:35.56 | irroot | func_odbc with a strip prefix / match query ?? |
| 16:36.34 | p3nguin | But are the numbers variable length? |
| 16:36.59 | irroot | ELECT prefix||substr('${ARG1}',strip+1) FROM trunkmap WHERE '${ARG1}' ~ match ORDER BY length(prefix) LIMIT 1; |
| 16:37.11 | irroot | s/ELECT/SELECT/ |
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| 16:41.48 | ssureshot | p3nguin: k, so I would make this change http://pastebin.com/SdqPi5A7 ? |
| 16:42.36 | p3nguin | ssureshot: Looks right to me. |
| 16:43.03 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-xngapyuahdglblwm) |
| 16:43.06 | ssureshot | thank you |
| 16:44.04 | *** join/#asterisk skrusty (~ben@93-97-20-22.zone5.bethere.co.uk) |
| 16:44.18 | p3nguin | You put entries with DB() and get them with ${DB()} |
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| 16:48.29 | irroot | cursing size of polycom bl / sip files for all model support |
| 16:53.14 | citywok | okay i have a crazy problem, i use realtime for voicemail, and have mailboxes of varying length. if i use the voicemail app as it stands, asterisk segfaults. if i delete all of the 4 digit mailboxes it doesn't. if i instead delete all of the mailboxes longer than 4 digits, it doesn't. |
| 16:54.53 | irroot | citywok what version ?? can you post a BT ?? |
| 16:55.17 | citywok | 1.6.2.18 (started in .11 after i rebuilt my box, basic debian 64bit) |
| 16:56.48 | citywok | yea, i'll post the bt, hang on |
| 16:57.23 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 16:58.20 | citywok | irroot: http://pastebin.com/SVX6vvq2 |
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| 16:59.04 | citywok | wow that backtrace was useless, sorry. i must have screwed up the gdb command |
| 16:59.15 | irroot | mmmm |
| 16:59.24 | irroot | lol you bt /dev/null :P |
| 16:59.33 | Katty | watches some dr who |
| 16:59.42 | irroot | who katty ?? |
| 16:59.54 | citywok | irroot: http://pastebin.com/d7Vhf0xv |
| 16:59.58 | Katty | yes |
| 17:00.02 | ariel_ | NoOp(${EXTEN:0:$[LEN(${EXTEN})-1]) |
| 17:00.11 | ariel_ | trying to see how I can close this correctly |
| 17:02.10 | citywok | irroot: i lied when i said it crashes when i use the voicemail app. i meant it crashes when i use the directory app. |
| 17:02.32 | citywok | i've spent so much time staring at this voicemail user table which is what the directory is reading from that i confused myself :P |
| 17:04.43 | irroot | ok see its not bt full and that its optomised out so not real helpfull can you rebuild * with dont_optomize |
| 17:05.01 | citywok | irroot: it was built w/ dont opt (but not debug threads) |
| 17:05.17 | irroot | please kind sir |
| 17:05.53 | citywok | i'm pretty sure it was never built without dont_opt, but i'll make clean and re-make it again just to be sure |
| 17:06.02 | citywok | i download 18.1 just to test it for this exact thing :p |
| 17:10.36 | citywok | irroot: when i run gdb it says "warning: Can't read pathname for load map: Input/output error." and in the output i see "No symbol table info available." |
| 17:11.17 | irroot | you may for some libs |
| 17:11.39 | irroot | did you run strip on your /modules path perhaps |
| 17:12.46 | citywok | i just wiped out the modules directory, recompield * from scratch (dont_opt), and then recompiled addons from scratch. |
| 17:13.35 | citywok | is this one any more helpful? http://pastebin.com/x1aCv8Lp |
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| 17:15.46 | irroot | this chilli will make a grown texan cry ... winter food |
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| 17:19.18 | irroot | citywok # gdb -se "asterisk" -ex "bt full" -ex "thread apply all bt" --batch -c core > /tmp/backtrace.txt |
| 17:20.13 | irroot | https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace |
| 17:21.21 | citywok | irroot: yea, that's where i copied it from originally. when it complained about symbols i tried using the path to asterisk instead. http://pastebin.com/RBJs7Efz |
| 17:22.10 | *** join/#asterisk neurosys (~neurosys@69.199.133.154) |
| 17:23.05 | citywok | interestingly in the scrollback of putty it shows running asterisk, and running the backtrace backwards. odd. i double checked the command history to make sure it was run in the right order (and i know it was the only core dumb b/c i rm'd them all before running it) |
| 17:24.41 | Footman | bye ! |
| 17:25.15 | citywok | bye |
| 17:26.36 | citywok | irroot: also, this happens on two different debian boxes (fairly similar debian lenny 5.0 installs), using the same copy of the voicemail database. same w/ *1.6.2.11 & addons 1.6.2.1 or .18.1 & .3 |
| 17:27.13 | irroot | cant see * blowing up in trace ... so hard to say |
| 17:27.44 | citywok | i've wiped pretty much all of the information from the voicemail db (password, fullname, email, tz) all set to the same test info, no dupe mailboxes, all in the same context. pretty much 300 identical rows except for different mailbox #'s |
| 17:42.53 | FLeiXiuS | With NAT'ed sip peers, should I see the external IP of the peers in asterisks' SIP SHOW PEERS? |
| 17:43.14 | p3nguin | If you have configured it correctly, yes. |
| 17:43.28 | FLeiXiuS | p3nguin, meh, apparently I do not. All of mine says UNSPECIFIED |
| 17:43.45 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 17:43.49 | p3nguin | Then the peers are not registered and not known. |
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| 17:44.50 | FLeiXiuS | p3nguin, Ah OK...they were connected but not registering for calls. |
| 17:44.58 | FLeiXiuS | Now to figure out why RTP isnt making it through |
| 17:46.31 | p3nguin | What is "connected" if they are not registered? |
| 17:46.54 | Kevin24 | HI everyone |
| 17:47.48 | FLeiXiuS | p3nguin, SIP will connect them to the server but not register their account/exten. It's for debugging. |
| 17:48.00 | Kevin24 | About SS&, i'm getting some SS7 flap issues, everything i have read and researched says its a problem in the underlying asterisk. $$$$$$ WARNING[31876]: chan_ss7.c:126 process_event: MTP is now DOWN on link 'ostional2'. |
| 17:48.00 | Kevin24 | [Jun 27 00:22:23] NOTICE[31876]: mtp.c:666 mtp_changeover: MTP changeover last_ack=4, last_sent=4, from schannel 31, no INSERVICE schannel found |
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| 17:48.23 | p3nguin | fleixius: That doesn't really make any sense to me since SIP is connectionless. |
| 17:49.46 | *** join/#asterisk vinhdizzo (~vinh@dhcp-053212.ics.uci.edu) |
| 17:50.29 | p3nguin | Anyway, make the phones register to Asterisk and the address will become known. |
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| 18:05.45 | Kevin24 | anyone that might have chan_ss7 knowledge please priv msg me |
| 18:08.13 | p3nguin | Or keep it in channel so everyone can benefit from it. |
| 18:08.27 | Kevin24 | that too |
| 18:08.28 | p3nguin | Since, you know, that's what we do here. |
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| 18:10.16 | Kevin24 | Yeah i understand the idea is to share the information. But i haven't heard anything yet on my issue. FOund some old posts but all seem to be dead ends. |
| 18:12.27 | p3nguin | I know there are some people that float around here who use SS7, but they rarely talk about it. |
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| 18:16.24 | citywok | irroot: https://issues.asterisk.org/jira/browse/ASTERISK-18072 |
| 18:16.25 | CaptainPants | [ASTERISK-18072] [Status: Triage] Segfault when using Directory() - https://issues.asterisk.org/jira/browse/ASTERISK-18072 |
| 18:16.35 | irroot | seen it |
| 18:16.42 | citywok | irroot: i did some digging and figured out where it crashes in app_directory, now i need to figure out where it goes in asterisk-addons |
| 18:16.57 | citywok | it's crashing on a call to ast_load_realtime_multientry |
| 18:17.07 | irroot | please citywok as much info as possiblw |
| 18:17.30 | citywok | i just added the line number and call it's crashing on to the bug report |
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| 18:25.11 | Katty | derkaderkader |
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| 18:56.59 | nny | winces |
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| 18:58.25 | nny | so say I am writing a loop to dial 10 at a time and play a message. I keep getting hung up on the notion that Dial *stops* the context. I wrote some cool sub routines and a GROUP count for the context to limit the channels, but realize every time my loop hits Dial, it stops, making it really only 1 concurrent call. Any thoughts? |
| 18:59.38 | p3nguin | If I need to dial 10 people at once, I dial them in sequence, immediately, opening 10 channels. When Dial() is done, it'll exit. |
| 18:59.42 | _Corey_ | nny: Where is this loop happening? |
| 18:59.55 | p3nguin | without a loop, I should add. |
| 19:00.06 | _Corey_ | yeah, I was thinking along the same lines... the loop thing doesn't make sense |
| 19:00.24 | _Corey_ | unless you're redialing busies or something |
| 19:00.26 | p3nguin | Usually a loop would be used to dial the same number over and over again. |
| 19:00.51 | nny | p3nguin: _Corey_ this is for a simple dialer that calls a list of numbers stored in a db. I can post my code if you think it'll help see what I am working on |
| 19:01.09 | p3nguin | I use a bash script. |
| 19:01.13 | _Corey_ | well, I'd push an asterisk call spool file or use AMI to initiate the calls |
| 19:01.29 | _Corey_ | I do the same as p3nguin for basic situations |
| 19:01.36 | p3nguin | I don't know how to use AMI, so I do everything from bash and asterisk -rx. |
| 19:01.58 | nny | _Corey_: ahh, had forgotten about the spool file |
| 19:02.28 | nny | _Corey_: man, I was just working on that last month, and got tunnel vision with this setup as is |
| 19:02.42 | _Corey_ | just make sure your script has a sleep command or something to batch 10 at a time and then wait an appropriate amount of time before batching another 10 |
| 19:03.53 | nny | _Corey_: yeah was trying to do it all in the dialplan, but stopping at Dial killed me every time. See: http://pastebin.com/MTX16XWa |
| 19:04.24 | p3nguin | I tune my batch size up to the available bandwidth. If I can get 50 calls going at once, I'll do 50 at once. Without much load on Asterisk from basic Dial() and Playback(), the bandwidth becomes my limiting factor. |
| 19:04.45 | p3nguin | Dial() has an option to continue with the rest of the dial plan. |
| 19:04.47 | *** join/#asterisk bluOxigen (~noreply@unaffiliated/bluOxigen) |
| 19:04.48 | nny | p3nguin: yeah this is gonna have a challenge response with a transfer too. |
| 19:04.51 | nny | p3nguin: g |
| 19:05.17 | nny | p3nguin: but the issue I found was that one occourance of the Dial (when really I was looping through the context with a GROUP 10 set) still stops it |
| 19:05.18 | p3nguin | My biggest issue was legality. |
| 19:05.25 | nny | check the paste bin I posted http://pastebin.com/MTX16XWa |
| 19:05.32 | nny | my idea is there, but Dial is my hurdle |
| 19:05.48 | _Corey_ | I guess as long as you're not calling people to renew their car warranties :) |
| 19:05.49 | nny | p3nguin: this is a subscribed list, so none there |
| 19:06.07 | nny | nah, subscribed list for a client, lol car warranties |
| 19:06.34 | nny | i have a 67 vw ghia, had a call once. Was like.. um.. sure! DO you know what car I drive? |
| 19:06.41 | p3nguin | For my situation, it is not legal to auto-dial a bunch of numbers with a sound file being played. It's legal to auto-dial, but it is required that a person be connected to the recipient rather than a pre-recorded message. |
| 19:06.58 | _Corey_ | ouch |
| 19:07.22 | p3nguin | FCC sucks sometimes. |
| 19:07.24 | nny | p3nguin: yeah that part is out of my hands, just assisting with the dial plan. Should be all on the up and up |
| 19:07.37 | p3nguin | It makes sense to have rules like that, but it impedes me. |
| 19:08.15 | _Corey_ | I had a customer decide to do one of these at 2am to 100,000 numbers and leave a voicemail |
| 19:08.25 | _Corey_ | they got fined by the FCC :) |
| 19:08.31 | nny | yeah gonna put a time check on this one hahahaha |
| 19:08.37 | p3nguin | I don't have 50 reps to connect to callees, so that's a problem for me. |
| 19:09.05 | nny | p3nguin: so your setup is in the dialplan? How do you manage the call rate? |
| 19:10.26 | _Corey_ | nny: I'm looking at your dialplan... explain the group count's purpose? |
| 19:11.23 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 19:11.35 | p3nguin | I have a list of numbers (a text file), and a shell script. The shell script has the number of calls to spawn as a batch all at once. |
| 19:12.43 | nny | _Corey_: http://www.voip-info.org/wiki/view/Asterisk+func+group essentially it was meant to limit the number of times channels that could be used for that context, if it exceeded 10, it went into a wait and checked again |
| 19:13.04 | nny | number of channels* |
| 19:13.28 | _Corey_ | yeah, I don't think the approach here is going to work |
| 19:13.31 | *** join/#asterisk gmaruzz (~gmaruzz@static-217-133-80-112.clienti.tiscali.it) |
| 19:13.36 | nny | p3nguin: interesting yeah looking into external triggers as well, (php, bash etc). |
| 19:14.13 | nny | _Corey_: yeah was good in theory, but the Dial statement stops all, so the approach needs to be rethought |
| 19:14.35 | nny | may do php scripting and AMI |
| 19:14.36 | _Corey_ | Yeah, the problem here is Dial isn't meant to be used the way you're doing it |
| 19:14.45 | nny | yeah |
| 19:15.04 | _Corey_ | You've got to use a call file or other method to Dial and then push the call into your dialplan |
| 19:15.30 | nny | yeah gonna make a script that populates the spool directory and follows the rules there in |
| 19:15.39 | nny | that would work though if I understand it correctly, no? |
| 19:15.52 | _Corey_ | I'd go with p3nguin's suggestion and use an external script |
| 19:16.33 | _Corey_ | yeah, you manage the concurrency there though |
| 19:16.45 | nny | _Corey_: yeah, can combine the two, i'll look into the spool options more. Thanks for the wall to bounce the idea off of, back to the drawing board |
| 19:16.55 | nny | p3nguin: thanks as well |
| 19:17.44 | _Corey_ | No problem, if you can wait until October I'm doing a session on this stuff @ Astricon |
| 19:17.46 | _Corey_ | :) |
| 19:17.52 | p3nguin | You could use something like PHP to write call files, but it's not something I would bother with. The bash script can do all the work of the insertion of phone numbers and then asterisk dial plan handles the rest. |
| 19:18.46 | nny | _Corey_: man, i'll try to be there, been dying to go |
| 19:19.41 | _Corey_ | Looks like a good program overall this year... should be fun |
| 19:19.56 | nny | _Corey_: where is it this year? |
| 19:20.02 | _Corey_ | Denver |
| 19:20.34 | nny | p3nguin: so your bash script is generating call files in spool? |
| 19:20.41 | nny | p3nguin: yeah bash over php seems better |
| 19:20.42 | p3nguin | nope |
| 19:21.01 | nny | p3nguin: is it executing commands in asterisk via -rx? |
| 19:21.20 | p3nguin | Bash pulls the phone numbers from the text file and originates with a local channel, which sticks the call right into the dial plan. |
| 19:21.36 | *** join/#asterisk Cadey (~x@host81-135-124-78.range81-135.btcentralplus.com) |
| 19:22.04 | Cadey | Hi guys, im looking for someone that help help me debug a sip retransmition issue |
| 19:22.09 | *** join/#asterisk nightrid3r (~kvirc@91.176.223.162) |
| 19:22.12 | Katty | hi Cadey how are you |
| 19:22.44 | Cadey | we get an incomming call but once the person picks up we get no audio and see retransmition messages like it cant negotiage which ports to use for media or somthing |
| 19:23.15 | p3nguin | All signs point to NAT misconfiguration. |
| 19:23.27 | Cadey | p3nguin, we dont use nat |
| 19:23.52 | p3nguin | It sounds like someone does. |
| 19:24.07 | Cadey | 172.16.100.x <-- (Local net) asterisk (internet ip) --> |
| 19:24.09 | nny | p3nguin: I think I see. Going to do some script writing and a little testing, mind i f I ask you questions later if I hit any walls? |
| 19:24.16 | Cadey | there is no nat what so ever dude |
| 19:24.37 | p3nguin | 172.16.100.x/24 is a private IP address, which requires you to configure NAT. |
| 19:24.40 | Cadey | the asterisk box as a internet facing nic which all calls go to |
| 19:24.47 | Cadey | we proxy all media through asterisk |
| 19:25.07 | *** join/#asterisk whirlpbx (~whirlpbx@109.124.76.159) |
| 19:25.37 | Cadey | so the 172 range has no need to see the internet range |
| 19:25.43 | Cadey | hence no nat for cals |
| 19:25.44 | p3nguin | If Asterisk has a private IP address and a public IP address, NAT settings need to be configured on Asterisk. |
| 19:26.02 | Cadey | p3nguin : its not all calls |
| 19:26.11 | Cadey | p3nguin : it happens speradicly |
| 19:26.50 | Cadey | p3nguin : i have a sip trace and TCP dump for the internet facing nic if you can help |
| 19:27.13 | nny | Cadey: reinvites on or off? |
| 19:27.28 | Cadey | I belive they are local only |
| 19:27.30 | Cadey | let me check |
| 19:28.06 | nny | shouldn't matter if the sip.conf has the proper local network etc, but that was my issue (plus a bad sip.conf nat setup) at one time |
| 19:28.25 | Cadey | ok in sip |
| 19:28.32 | Cadey | localnet=172.16.100.0/255.255.255.0 |
| 19:28.33 | Cadey | localnet=10.1.0.0/255.255.0.0 |
| 19:28.33 | Cadey | localnet=10.2.0.0/255.255.0.0 |
| 19:28.40 | Cadey | as we have 3 local nets it can use |
| 19:28.47 | Cadey | canreinvite=yes <-- |
| 19:28.56 | Cadey | i can change that to no |
| 19:30.44 | Cadey | I guess it could be the reinvite, but why? |
| 19:31.47 | Cadey | saying that, can you have more than one setting for localnet or does the last one take presidence? |
| 19:31.56 | Cadey | as the PBX has multiple nics |
| 19:32.29 | nny | assuming the reinvite (hazy here) is telling your provider to communicate directly with a phone |
| 19:32.41 | nny | not sure if asterisk forbids that in your situation or not |
| 19:32.45 | p3nguin | You can have multiple localnet entries, and should have one for each localnet you have. |
| 19:33.16 | Qwell | wonders why Asterisk doesn't automatically know that rfc1918 == localnet |
| 19:33.16 | Cadey | well the phones are forced to proxy media through the asterisk box |
| 19:33.39 | Cadey | so any reinvite would also be forced to the asterisk box |
| 19:33.43 | Cadey | but its worth a try |
| 19:34.08 | Cadey | p3nguin : yeah that is what I was thinking, so its correct |
| 19:34.22 | nny | Qwell: heh um |
| 19:34.43 | nny | Qwell: I have a provider that uses a PFC as it's outward facing network interface, when talking to a public interface |
| 19:34.54 | p3nguin | its |
| 19:34.55 | nny | Qwell: it's stupid and odd, but it works |
| 19:34.57 | nny | its* |
| 19:35.04 | nny | gah HERE COMES AN S! |
| 19:35.08 | Qwell | PFC? |
| 19:35.10 | p3nguin | hahaha |
| 19:35.53 | Katty | Qwell: come watch dr who |
| 19:35.58 | Qwell | k |
| 19:36.01 | nny | Qwell: er acronym fail, but an rfc1918 compliant address |
| 19:36.40 | Qwell | They aren't routable. It isn't coming from that address. It might be in the headers (in which case, they're dumb and broken), but that's about it. |
| 19:37.29 | p3nguin | I wouldn't expect to see very many large providers using private addressing on public phone equipment. |
| 19:38.02 | nny | Qwell: p3nguin one sec pastebin. I didn't set this up, and I can't get a straight answer as to why it is. |
| 19:38.23 | nny | http://pastebin.com/BCEexEfw |
| 19:38.40 | nny | but I can access the local IP on the nix box I did that from |
| 19:38.51 | Qwell | Do you have a VPN to them? |
| 19:38.57 | nny | no |
| 19:39.00 | nny | not in our equipment |
| 19:39.03 | nny | maybe in there |
| 19:39.05 | nny | theres* |
| 19:39.07 | nny | theirs* |
| 19:39.10 | nny | meh i need lunch |
| 19:39.54 | Qwell | Your internets are broken. |
| 19:40.00 | nny | lol actually it works |
| 19:40.10 | nny | that's the main pipe, just doesn't respond to ping past that addy |
| 19:40.16 | nny | i dunno, all their stuff |
| 19:40.40 | nny | always a pain in the ass to diagnose anything wrong with it too |
| 19:41.23 | *** join/#asterisk timahvo1 (~rogue@41.223.57.78) |
| 19:41.28 | *** join/#asterisk The_REV (~The_REV@76-217-184-220.lightspeed.frsnca.sbcglobal.net) |
| 19:41.44 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-148.cablep.bezeqint.net) |
| 19:43.31 | nny | if i tracert it from my desk here on another network etc, the last 3 legs prior to the public interface time out.. i dunno. I am pretty savvy with networking, but this setup has always made me scratch my head |
| 19:44.05 | nny | it's almost like they are doing public ips --> private ips --> public ips |
| 19:44.53 | wdoekes2 | that's quite possible.. there's no need for anyone else to reach those routers |
| 19:45.01 | wdoekes2 | so they're free to use private ip's |
| 19:45.05 | nny | wdoekes2: yeah |
| 19:45.15 | nny | wdoekes2: but it's hardly sane imho |
| 19:45.57 | p3nguin | So you're trying to traceroute a public IP address... from what? |
| 19:46.05 | p3nguin | correction... |
| 19:46.11 | p3nguin | So you're trying to traceroute a private IP address... from what? |
| 19:46.17 | nny | nah the ip is public |
| 19:46.27 | nny | but it transverses what appears to be a private network |
| 19:46.27 | p3nguin | I see you've run a traceroute to a private adddress. |
| 19:46.39 | nny | er that's out to their media gateway |
| 19:46.40 | nny | this is in |
| 19:46.54 | p3nguin | 10.10.212.37 |
| 19:47.05 | p3nguin | Where are you running the traceroute from? |
| 19:47.22 | nny | p3nguin: thats from the * box to the sip provider |
| 19:47.27 | nny | p3nguin: http://pastebin.com/6kFwgmSR is to that ip remotely |
| 19:47.53 | atheos | nny - do you have the same no-audio issues with inner office SIP to SIP calls? |
| 19:47.59 | nny | atheos: no audio here |
| 19:48.06 | nny | sorry this is a waste of channel space |
| 19:48.11 | p3nguin | I don't see any problem with the traceroute. |
| 19:48.11 | nny | no audio issues***! |
| 19:48.40 | nny | i am just discussing it, i'll stop, not important for the channel, more people need help.. i was just making useless conversation |
| 19:49.03 | p3nguin | In the traceroute where you try tracing a public IP address, the trace appears to reach the destination. |
| 19:49.34 | atheos | sorry, meant p3nguin - not nny. |
| 19:49.48 | p3nguin | In the traceroute where you attempt to trace a private IP address, it's a little confusing since I don't know where your trace is originating. |
| 19:49.59 | p3nguin | atheos: I also do not have any audio issues. |
| 19:50.24 | Qwell | ...I think ISPs are advertising that route. |
| 19:50.44 | atheos | time for me to go home and have a beer. disregard me |
| 19:51.24 | *** join/#asterisk moy (~moy@CPE002719f00364-CM0026f3a1701d.cpe.net.cable.rogers.com) |
| 19:51.24 | Qwell | wtf is so special about 10.10? This is bizarre. |
| 19:51.27 | p3nguin | I just don't see a "problem" in the traceroutes. |
| 19:51.37 | Qwell | p3nguin: It's leaving his network. |
| 19:51.43 | Qwell | His ISP is routing it. |
| 19:51.54 | Qwell | (...as is ours, apparently) |
| 19:52.21 | p3nguin | It's probably reflecting off the edge router when it is trying to go out. The anti-spoof mechanism is on the external side. |
| 19:52.36 | p3nguin | It's not really that uncommon. |
| 19:54.15 | Qwell | What I don't understand is how that 64. address is hop 2. |
| 19:54.31 | Qwell | well, 1, since he's doing it from the gateway or something |
| 19:54.47 | p3nguin | I'd venture to say that is is probably seen more often where vlans try pushing traffic out a single router. The router then says, "Hey, wait! That's a private address; send that sucker back inbound where it belongs." |
| 19:55.26 | Qwell | In my case, this ping is getting all the way to San Jose, from AL... |
| 19:55.55 | p3nguin | I also see it on cable networks. |
| 19:59.22 | nny | Qwell: p3nguin sorry had to step away for a sec, that address is a nic on a linux box (the pbx) |
| 19:59.34 | Qwell | nny: what address? |
| 19:59.41 | nny | Qwell: p3nguin er the tracert |
| 20:01.04 | p3nguin | I still have no idea where that traceroute is originating. |
| 20:01.43 | nny | p3nguin: you mean the IP or physically? |
| 20:03.14 | p3nguin | physically, logically, virtually... I have no idea where that traceroute is being run. |
| 20:04.08 | nny | p3nguin: the first one is a pbx attached to an adtran aggregating 3 t1s. the ip on the pbx is a public ip, (64.194.212.250) |
| 20:04.34 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 20:04.41 | wdoekes2 | nny: the fact that you tracert a private ip, and not a public one from that 64. confuses people.. |
| 20:04.58 | p3nguin | What is the .249 address on? |
| 20:05.44 | nny | p3nguin: the adtran |
| 20:05.53 | *** join/#asterisk fhmiv (~fhmiv@c-67-173-205-151.hsd1.ga.comcast.net) |
| 20:05.53 | nny | wdoekes2: yeah that's the odd part for me too. |
| 20:06.20 | p3nguin | And the PBX has the .250 public address on it AND a private address as well? |
| 20:06.24 | nny | p3nguin: the private ip is the sip peer registration address as defined in sip.conf |
| 20:06.26 | nny | p3nguin: no |
| 20:06.35 | nny | p3nguin: well not in that range |
| 20:06.55 | nny | p3nguin: it has a pirvate ip, but it's different. That's just the trunk for the sip provider |
| 20:07.15 | p3nguin | Is the Adtran the gateway between that private range and its own public IP address? |
| 20:07.21 | nny | p3nguin: yes |
| 20:07.57 | p3nguin | Sounds perfectly reasonable to me, then. |
| 20:08.08 | wdoekes2 | nny: I was assuming your first pb would also show private IPs if you tracert a non-rfc1918 ip, does it? |
| 20:08.30 | nny | wdoekes2: no the two device in between don't respond to ping |
| 20:08.42 | nny | wdoekes2: er one sec |
| 20:08.50 | nny | wdoekes2: let me test |
| 20:09.14 | p3nguin | You can't directly traceroute a range that you don't have on your system. You have to go through a gateway of some sort that does the routing. In this case, it seems apparent that the Adtran is doing the routing. |
| 20:09.21 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 20:09.44 | *** join/#asterisk fhmiv (~fhmiv@c-67-173-205-151.hsd1.ga.comcast.net) |
| 20:10.27 | nny | p3nguin: I need to check, you notice the latency jumps up to 25 ms after the first hop, must be something upstream |
| 20:10.48 | p3nguin | In which traceroute? |
| 20:10.51 | nny | and yeah, if I tracert a public IP , it goes out another interface |
| 20:10.59 | nny | http://pastebin.com/BCEexEfw |
| 20:11.18 | nny | since I have route statement in the pbx sending all packets to the private ip over that interface |
| 20:11.42 | p3nguin | That one does not go upstream. It hits the Adtran, which acts as a gateway, and then heads down the route to the destination. |
| 20:12.09 | p3nguin | Seems like normal operation to me. |
| 20:12.37 | p3nguin | Check the routing table on the Adtran. |
| 20:13.31 | *** join/#asterisk nightrid3r (~kvirc@91.176.9.195) |
| 20:13.44 | p3nguin | Also check the quality of the routes that it lists. |
| 20:17.14 | *** join/#asterisk seraphie (~erin@207.98.195.107) |
| 20:21.57 | *** join/#asterisk CoderForLife (~Miranda@unaffiliated/coderforlife) |
| 20:22.31 | *** join/#asterisk esaym153 (~esaym153@cpe-72-181-109-14.satx.res.rr.com) |
| 20:22.31 | *** part/#asterisk Kevin24 (~kevin@dsl-emcali-190.99.201.165.emcali.net.co) |
| 20:24.13 | esaym153 | anyone using asterisk with gtalk? I had jabber connecting with google earlier today, but I restarted asterisk and no jabber just keeps showing "res_jabber.c: JABBER ERROR: No Connection |
| 20:25.13 | esaym153 | looking with wireshark just shows asterisk opening the connection to the google ip and then killing it. No data is ever sent either way |
| 20:26.13 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 20:27.24 | Katty | dumb de dumb |
| 20:28.37 | *** join/#asterisk WiretapWork (~Wiretap@unaffiliated/wiretap) |
| 20:30.34 | *** join/#asterisk [sr] (~kvirc@pal-213-228-140-150.netvisao.pt) |
| 20:30.36 | [sr] | hi WIMPy |
| 20:30.41 | [sr] | WIMPy: solved the case |
| 20:31.38 | beek | Okay gang -- I'm getting this error on IMAP VM storage on both 1.6 and 1.8: app_voicemail.c:2784 mm_log: IMAP Warning: Unknown message data: 636 FETCH |
| 20:32.13 | beek | Googling lists tons of this error, but usually related to another package. |
| 20:32.23 | beek | What is this thing? Anyone? |
| 20:35.04 | nny | is dtmf=inband suppose to work with G729? just curious, my worm addled brain thought it wasn't |
| 20:36.29 | nny | p3nguin: thanks again, yeah just seemed odd, but if it works so be it |
| 20:37.01 | *** join/#asterisk jc319 (~jc318@78-86-169-203.dsl.cnl.uk.net) |
| 20:38.25 | zerohalo | dtmf does not work with g729 |
| 20:38.30 | zerohalo | err, in band that is |
| 20:44.19 | Scorpio2007 | if an incomming call negotiats gsm .. how can i control the outgoing to gsm also ? it seems like incomming negotiates gsm but the outgoing insists on using ulaw and transcode .. |
| 20:44.33 | Scorpio2007 | in both incomming and outgoing i have stated Codec Order : (ulaw:20,gsm:20) |
| 20:49.58 | *** join/#asterisk The_REV_ (~The_REV@184-194-67-226.pools.spcsdns.net) |
| 20:50.35 | *** join/#asterisk jbeez (jbeez@pool-96-245-221-98.phlapa.fios.verizon.net) |
| 20:50.56 | *** topic/#asterisk by leifmadsen -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.4.4 (2011/06/28), 1.6.2.18.2 (2011/06/28), 1.4.41.2 (2011/06/28), *-Addons 1.6.2.3, 1.4.13 (2010/01/14), dahdi-linux 2.4.1.2 (2011/04/11), dahdi-tools 2.4.1 (2011/03/03), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 20:51.28 | jbeez | hey guys Im trying to configure linphone on android, does anyone know how to format the settings to fit authorization username? most clients have a spot for it but I don't see it here, wondering if i can tack it onto another setting somehow with a : or an @ or a \ |
| 20:51.29 | leifmadsen | Asterisk security releases for branches 1.4, 1.6.2, and 1.8 are now available: http://www.asterisk.org/node/51653 |
| 20:51.34 | *** join/#asterisk CaptainPants (~CaptainPa@nat/digium/x-txhmfgqvssjbflpt) |
| 20:53.04 | citywok | leifmadsen: when is the next release of addons? bugfix 1151 is critical for me, prevents a segfault. |
| 20:53.17 | leifmadsen | what do you mean? |
| 20:53.19 | leifmadsen | just use the latest branch |
| 20:53.25 | citywok | yea, that's what i'm doing now in testing |
| 20:53.55 | leifmadsen | it'll happen at the same time as the final 1.4 and 1.6.2 releases |
| 20:54.19 | citywok | i posted a bug report for it, and in going through and trying to figure out where the problem came from figured out it had been fixed a couple weeks ago https://issues.asterisk.org/jira/browse/ASTERISK-18072 |
| 20:54.20 | CaptainPants | [ASTERISK-18072] [Status: Triage] Segfault when using Directory() - https://issues.asterisk.org/jira/browse/ASTERISK-18072 |
| 20:54.50 | citywok | before posting the bug i didn't try the svn version out of 1.6.2 |
| 20:55.43 | leifmadsen | wait are you talking about addons or the 1.6.2 branch? |
| 20:56.27 | citywok | yea |
| 20:56.49 | citywok | 1.6.2.3 has the bug, but the 1.6.2 branch doesn't. |
| 20:56.59 | leifmadsen | you don't mean 'addons' then |
| 20:57.14 | citywok | sorry. the 1.6.2 version under asterisk-addons |
| 20:57.24 | leifmadsen | ok well 1.6.2.19-rc1 is already out... |
| 20:57.31 | leifmadsen | it probably has the fix you speak of |
| 20:57.37 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 20:58.00 | citywok | the fix is in http://svn.digium.com/svn/asterisk-addons/branches/1.6.2/ -- it's the res_config_mysql.c class for realtime*. |
| 20:58.15 | leifmadsen | and is it in 1.6.2.19-rc1 which I just made.... on Friday? |
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| 21:01.11 | citywok | leifmadsen: wouldn't it have to be in asterisk-addons-1.6.2.[4?]-rc since it's a class in the -addons ? |
| 21:01.24 | citywok | sorry, i think i've just managed to confuse you by not using the right terms :p |
| 21:01.24 | leifmadsen | I don't understand your question |
| 21:01.36 | leifmadsen | the 1.6.2 branch contains -addons |
| 21:02.03 | citywok | but -addons aren't included in 1.6.2 releases, right? so 1.6.2.19-rc1 won't include the -addons |
| 21:02.09 | citywok | addons are only included in 1.8? |
| 21:02.09 | leifmadsen | I think you are wrong |
| 21:02.16 | leifmadsen | I may be wrong |
| 21:02.36 | leifmadsen | anyways, if the changes are in -addons, then my statement of, "it will be released with the final 1.6.2 release" stands |
| 21:02.38 | citywok | if i navigate to http://svn.asterisk.org/svn/asterisk/tags/1.6.2.19-rc1/res/ i don't see res_config_mysql.c |
| 21:02.57 | citywok | i'm assuming b/c the addons are packaged seperately in the asterisk 1.6.2 releases |
| 21:03.01 | leifmadsen | yes |
| 21:03.06 | leifmadsen | and changes are infrequent |
| 21:03.10 | leifmadsen | so I don't do those every month |
| 21:03.33 | nny | good program for testing sip on a low level (like a sip ping of some kind) any advice |
| 21:03.35 | leifmadsen | the latest 1.6.2-addons branch will have a release done at the same time as the final 1.6.2 release |
| 21:03.43 | leifmadsen | sipp? |
| 21:03.49 | leifmadsen | sipsak? |
| 21:03.57 | leifmadsen | try those |
| 21:04.16 | nny | leifmadsen: ty |
| 21:04.31 | citywok | with just 2 minor changes is it still wise to use the entire version from svn, instead of just patching my 1.6.2.3 install? |
| 21:04.55 | citywok | so far i'm just testing it with the patch applied to 1.6.2.3 |
| 21:05.20 | leifmadsen | that is a local implementation question |
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| 21:05.53 | citywok | kk, i don't really see any reason why it would cause any problems since the changes are local to the res_config_mysql.c class |
| 21:06.11 | citywok | so i'll continue testing it this way, thx |
| 21:06.30 | *** join/#asterisk jeffspeff (~jeffspeff@173-11-144-149-houston.txt.hfc.comcastbusiness.net) |
| 21:07.58 | jeffspeff | i'm having trouble using pickup groups. I have defined one user in a callgroup and another user in a corresponding pickupgroup. when the callgroup user is recieving a call and the pickupgroup user dials *8 to pickup the call i get an erro that it was rejected because the extension wasn't found in the context. any help would be awesome! |
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| 21:18.23 | citywok | leifmadsen: also, when you go to post a bug on jira it won't let you add a version for asterisk-addons. has the label there, but no textbox. |
| 21:19.36 | leifmadsen | you'll have to speak with the maintainer of jira :) |
| 21:19.40 | leifmadsen | espiceland |
| 21:19.45 | leifmadsen | serafie: << |
| 21:20.46 | citywok | will do :) |
| 21:21.39 | beek | downloading 1.8.4.4 now |
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| 21:52.45 | WIMPy | [sr]: Hi. How? |
| 21:53.09 | [sr] | WIMPy: just setting a CID on the route, maybe asterisk 1.8.4.3 or the new libpri needs it |
| 21:53.54 | WIMPy | "on the route"? |
| 21:53.59 | [sr] | si |
| 21:54.02 | [sr] | the outbound route |
| 21:54.11 | WIMPy | i.e. for all calls going out via dahdi? |
| 21:54.30 | [sr] | yes, for the route that uses the ISDN card |
| 21:54.39 | [sr] | on the FXO card it doesn't complain |
| 21:54.40 | WIMPy | I'd call it a bug in libpri. |
| 21:55.30 | [sr] | WIMPy: or asterisk 1.8.4.3 |
| 21:55.49 | [sr] | i say this 'cause yesterday i tested with the libpri i had before the upgrade to .5 |
| 21:55.55 | [sr] | and the problem was the same |
| 21:55.59 | WIMPy | No. Unless it's an explicit configuration option. |
| 21:57.43 | [sr] | or i'll try asterisk 1.8.4.4 to see if there's any change in behavior |
| 21:59.14 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 21:59.52 | [sr] | only see one change for 1.8.4.4 |
| 21:59.59 | [sr] | have to see the full description |
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| 22:03.16 | WIMPy | can't find anything obvious in Asterisk changelog. |
| 22:05.09 | *** join/#asterisk amoR (~amoR@ip50-118.245.80.crimea.com) |
| 22:05.56 | WIMPy | Has downloads.asterisk.org just died or is it my ISP? |
| 22:05.58 | *** join/#asterisk WiretapWork_ (~Wiretap@unaffiliated/wiretap) |
| 22:06.21 | _Corey_ | working for me |
| 22:06.42 | WIMPy | Hmmm. |
| 22:06.54 | WIMPy | can ping it, but otherwise getting connection timed out. |
| 22:07.27 | WIMPy | Got it again. |
| 22:07.28 | hardwire | pings WIMPy |
| 22:09.35 | WIMPy | Nothing obvious in libpri changelog, either. |
| 22:11.52 | [sr] | WIMPy: it can be something with freepbx also |
| 22:13.05 | WIMPy | Sure, but it must not happen, whatever you configured. |
| 22:13.33 | WIMPy | Unless there's an option to explicitely do that in spite of the fact that it's not allowed. |
| 22:16.37 | [sr] | WIMPy: well for now its ok |
| 22:17.51 | [sr] | well |
| 22:17.55 | [sr] | now im going to sleep!! |
| 22:17.58 | [sr] | chat later |
| 22:18.23 | WIMPy | o/~ |
| 22:21.30 | *** join/#asterisk ariel_ (~chatzilla@pdpc/supporter/active/abatista) |
| 22:23.36 | WIMPy | A missing interface in chan_capi seems to prevent Asterisk from starting. |
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| 22:33.51 | puzzled | WIMPy: what's the error? |
| 22:34.49 | WIMPy | It sends the caller name as display IE in the setup in TE mode. |
| 22:35.20 | jeffspeff | i'm having troubles with using pickup(). the console shows that the call is answered by the pickupgroup user but after that you can't hear or talk and asterisk won't make or recieve any type of call and has to be restarted. |
| 22:35.36 | WIMPy | Interestingly his telco only objected when the name was too long, but that should not happen. |
| 22:35.42 | puzzled | WIMPy: how is that related to * not starting? sounds like it started fine :) |
| 22:36.05 | WIMPy | Oh, you mean the capi thing. |
| 22:36.09 | puzzled | yup |
| 22:36.15 | WIMPy | That didn't give an error. |
| 22:36.44 | WIMPy | Let me put that config back. |
| 22:38.22 | WIMPy | With enough debug it gives "WARNING: chan_capi.c: cc_init_capi: CAPI not installed, chan_capi disabled! |
| 22:38.29 | WIMPy | Exit 1 |
| 22:39.00 | WIMPy | " |
| 22:40.54 | puzzled | seems the capi drivers are not installed or if they are and are properly detecting the card then perhaps the /dev/capi* udev rules are not in place/wrong |
| 22:41.13 | WIMPy | Correct. I didn't load capi. |
| 22:42.22 | WIMPy | But I expected what it said, not that it prevents Asterisk from starting. |
| 22:43.18 | _jdccdevel | Quick question about priority labels. Is it OK to re-use a label within a context, but with different extensions? i.e. exten => 1000,n(sayhi),Noop(Hi 1000) and exten =>1001,n(sayhi),Noop(Hi 1001) ? |
| 22:43.35 | WIMPy | _jdccdevel: Sure |
| 22:43.39 | Qwell | _jdccdevel: They are priority labels. |
| 22:43.57 | _jdccdevel | Excellent. |
| 22:44.36 | jeffspeff | i'm having troubles with using pickup(). the console shows that the call is answered by the pickupgroup user but after that you can't hear or talk and asterisk won't make or recieve any type of call and has to be restarted. |
| 22:48.32 | jeffspeff | here's the sip debug of my problem starting from initiating the incoming call http://pastebin.com/5fvX8iCQ any help is greatly appreciated |
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| 23:00.45 | *** join/#asterisk obelix (~obelix@unaffiliated/pavlz) |
| 23:00.48 | obelix | hello |
| 23:00.59 | obelix | i got one query for you |
| 23:01.24 | obelix | actually i am using a phone siemens gigaset |
| 23:01.51 | obelix | and my cousin created to me an account on voipstunt |
| 23:01.53 | WIMPy | A SIP one? |
| 23:02.13 | *** join/#asterisk nightrid3r (~kvirc@91.176.58.161) |
| 23:02.14 | obelix | the fact is that if i finish the credit on cheapnet and voipstunt |
| 23:02.36 | obelix | then it not anyomore to phone (and to receive) |
| 23:02.55 | obelix | is there a way to bypass sip |
| 23:03.33 | obelix | it is possble to phone using the ADSL without pay to cheapnet and voipstunt ? |
| 23:03.45 | obelix | can you help me ? |
| 23:03.53 | WIMPy | Use another ITSP. |
| 23:04.00 | obelix | ITSP ? |
| 23:04.08 | WIMPy | But that's not an Asterisk question. |
| 23:04.12 | WIMPy | ~itsp |
| 23:04.13 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
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| 23:05.13 | obelix | i wrote on asterisk, because i thinked: better of them, don't know others |
| 23:05.29 | obelix | the fact is that i am in Italy |
| 23:06.20 | obelix | and i got the ADSL with TISCALI, tiscali wants cent of euro for minute + 15 cent of euro for any beginning call |
| 23:07.33 | obelix | pratically there is not other ITSP that offer a service similar to voipstunt and cheapnet, they offer you the possibility to phone, having 45 minutes per day |
| 23:08.52 | obelix | you can decide to phone for a time more long than 45 minutes, but then your credit is low |
| 23:09.49 | obelix | pratically you are charging to have the service, and you pay only for the cost of credit you charged on |
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| 23:34.43 | *** join/#asterisk \DSAFEW\ (~\DSAFEW\@ip68-3-53-165.ph.ph.cox.net) |
| 23:34.56 | \DSAFEW\ | ~book |
| 23:34.56 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 23:35.38 | \DSAFEW\ | ~newbook |
| 23:35.38 | infobot | Please see ~thebook for more information about Asterisk: The Definitive Guide |
| 23:35.53 | \DSAFEW\ | Is this the most updated one? |
| 23:36.21 | WIMPy | yes |
| 23:36.32 | \DSAFEW\ | oh, the color scheme seemed off |
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| 23:51.31 | megalomano | hi |
| 23:52.53 | megalomano | pls, some one knows how is that incoming calls works between users a2billing? |
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