| 00:02.59 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 00:05.09 | amazinzay | exit |
| 00:16.33 | *** join/#asterisk [Outcast] (~anonymous@24-183-177-242.dhcp.oxfr.ma.charter.com) |
| 00:27.05 | *** join/#asterisk Montys (~dmartinez@nat/digium/x-uuvvlsnwqbamajxg) |
| 00:27.36 | Montys | Guys, Question, How you enable the Call Pickup feature in Asterisk |
| 00:28.22 | Montys | according to what I read, it seems that you need to set pickupgroup and callgroup in the user |
| 00:28.45 | Montys | but everyting that I dial *8 Asterisk says that the *8 does not exist as valid extention |
| 00:28.47 | Montys | any ideas? |
| 00:31.44 | ChannelZ | is it enabled in features.conf? |
| 00:32.16 | Montys | Yep |
| 00:32.35 | Montys | pickupexten = *8 |
| 00:32.48 | Montys | ^^ ChannelZ |
| 00:33.50 | ChannelZ | And are these sip phones, analog phones? (is the *8 even making it to Asterisk, are you seeing any errors on the console?) |
| 00:34.25 | Montys | iax |
| 00:35.16 | Montys | and this is the error that is showing: NOTICE[10750]: chan_iax2.c:9491 socket_process: Rejected connect attempt from <MyIP>, request '*8@DLPN_DialPlan1' does not exist |
| 00:35.42 | *** join/#asterisk hrhrhr (~c1@unaffiliated/hrhrhr) |
| 00:36.42 | ChannelZ | hmm |
| 00:36.47 | *** join/#asterisk ariel_ (~chatzilla@99-1-236-49.lightspeed.miamfl.sbcglobal.net) |
| 00:38.06 | Montys | Do I need to add something in the dialplan too tell Asterisk that I'm going to use the features? |
| 00:38.24 | *** join/#asterisk Bidik (~bidik@li267-109.members.linode.com) |
| 00:38.35 | *** join/#asterisk niekie (~niek@CAcert/Assurer/niekie) |
| 00:39.26 | *** join/#asterisk mac-mini (~mac-mini@unaffiliated/macmini/x-648924) |
| 00:40.36 | *** join/#asterisk binbash_ (~peter@ip4da5c213.direct-adsl.nl) |
| 00:43.47 | ChannelZ | I dunno this is confusing |
| 00:43.58 | Montys | same for me :( |
| 00:44.41 | Montys | I search on the interwebs for clues about what I doing wrong, but sadly there is not good info |
| 00:45.03 | Montys | I searched * |
| 00:45.43 | ChannelZ | you don't have a catch-all extension or anything defined in extensions.conf that would be matching the *8 ? |
| 00:46.37 | Montys | Nop, |
| 00:46.40 | ChannelZ | cuz I don't even have pickup configured at home and get "Nothing to pick up for xxxxx" when I try. The fact that you're getting more of an unknown extension notice would lead me to believe it's being trapped some other way |
| 00:46.43 | Montys | *CLI> dialplan show *8@DLPN_DialPlan1 |
| 00:46.43 | Montys | There is no existence of *8@DLPN_DialPlan1 extension |
| 00:47.28 | ChannelZ | yes but you don't have any other pattern that could possibly match? |
| 00:48.14 | Montys | to be quite honest, I not sure, I downloaded the latest SVN asterisk GUI version and installed, |
| 00:48.37 | Montys | I dont think that the dialplan would have something that could match *8 |
| 00:49.42 | ChannelZ | I guess the dialplan show would have shown a pattern match. meh |
| 00:50.28 | Montys | Could you please check http://pastebin.com/Z1RKDAP9 , and let me know if I missing a context that could be the reason that call pickup is not working |
| 00:51.03 | ChannelZ | that doesn't mean a lot since I've no idea what are in those contexts |
| 00:51.31 | ChannelZ | is this freepbx or the old asterisk gui? in either case they might have hotwired the *8 in the first place and have their own mechanism for the call pickups |
| 00:51.37 | Montys | Asterisk GUI |
| 00:51.39 | ChannelZ | but I dunno cuz I don't use those things |
| 00:52.17 | *** join/#asterisk niekie (~niek@CAcert/Assurer/niekie) |
| 01:00.40 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 01:04.04 | Montys | ChannelZ: Well, I need to go, Thank you for you assistance :) |
| 01:12.41 | ChannelZ | or lack thereof |
| 01:12.42 | ChannelZ | good luck |
| 01:21.18 | *** join/#asterisk saxa (~sasa@189.26.255.43) |
| 01:21.54 | saxa | ~newbook |
| 01:21.55 | infobot | Asterisk: The Definitive Guide. A preview of the upcoming release is available at http://ofps.oreilly.com/titles/9780596517342/. The book is being released under a Creative Commons license: http://creativecommons.org/licenses/by-nc-nd/3.0/us/. Please consider supporting the authors generosity by purchasing a copy at http://oreilly.com/catalog/9780596517342. |
| 01:22.03 | saxa | thx infobot |
| 01:22.17 | leifmadsen | infobot: thanks! |
| 01:22.17 | infobot | my pleasure, leifmadsen |
| 01:22.49 | saxa | leifmadsen: thx again :) I need to read more stuff in there :) |
| 01:22.56 | leifmadsen | it's a good book :) |
| 01:23.24 | saxa | just trying to access it from another machine |
| 01:23.47 | saxa | leifmadsen: i believe so, thx for writing it |
| 01:24.21 | saxa | it seems my server at home went offline :( |
| 01:25.50 | saxa | leifmadsen: btw, is there a pdf version of the book ? I mean the new one ? |
| 01:26.22 | saxa | sometimes is very handy to read offline |
| 01:26.24 | leifmadsen | saxa: no, PDF will be via purchase this time around |
| 01:26.35 | leifmadsen | only HTML will be available |
| 01:26.47 | saxa | ok, who knows maybe I buy it |
| 01:26.49 | leifmadsen | of course you could download and save the HTML |
| 01:26.55 | leifmadsen | maybe you will |
| 01:27.02 | leifmadsen | maybe I'll write another version if it' |
| 01:27.05 | saxa | and then print it in .pdf :) |
| 01:27.06 | leifmadsen | s worth my time :) |
| 01:27.22 | leifmadsen | remember the authors don't make very much |
| 01:27.31 | leifmadsen | I'm not buying new cars with the royalties :) |
| 01:27.45 | saxa | even that not ? |
| 01:27.58 | leifmadsen | ? |
| 01:28.08 | saxa | you should be able to buy at least one new car :) |
| 01:28.11 | leifmadsen | my wife and I share a car heh |
| 01:28.22 | saxa | heh |
| 01:28.26 | saxa | lie me |
| 01:28.29 | leifmadsen | luckily i work from home an don't need to go anywhere |
| 01:28.29 | saxa | like me |
| 01:28.42 | leifmadsen | anyways, I'm going to drink wine with said wife |
| 01:28.42 | saxa | thats a good part of this job |
| 01:29.07 | saxa | woow, then have a nice cup of wine |
| 01:29.15 | saxa | and make smile your wife |
| 01:29.35 | saxa | I'll go to read * book |
| 01:29.41 | saxa | have a good night |
| 01:30.06 | saxa | exit |
| 01:35.34 | *** join/#asterisk f2Knight (~ben@c-71-236-165-148.hsd1.or.comcast.net) |
| 01:37.38 | f2Knight | Q: Just put up a clean install of 1.8.2.4, modified extconfig to get sip users from mysql. Registed a phone and it connects fine. I am using g722 codec, dialing the 'demo' I hear the audio, but when it tries to connect to digium i get a translation error |
| 01:38.50 | f2Knight | I get Call rejected by 216.207.245.8: Unable to negotiate codec |
| 01:39.20 | pabelanger | f2Knight: over IAX2? |
| 01:39.48 | f2Knight | yes |
| 01:39.53 | f2Knight | pabelanger, yes. |
| 01:40.33 | pabelanger | f2Knight: Hmm, it maybe possible this is a IAX2 bug on the Digium side. Can you see what version of Asterisk it is running? |
| 01:40.53 | pabelanger | Not sure if that is exposed via IAX2 messages |
| 01:41.04 | f2Knight | pabelanger, this is really a clean install... right out of the box the only mods i have made is to enable realtime sip accounts in a db. |
| 01:41.48 | pabelanger | f2Knight: Yes, if it is the problem I am thinking of, its happening on the farside, not your box |
| 01:42.04 | f2Knight | umm how can i check |
| 01:42.15 | f2Knight | because i have another box in my room that is svn and it works |
| 01:42.37 | f2Knight | pabelanger, in fact, i actually coppied the database file from that box and put it on the other box. |
| 01:43.24 | pabelanger | ~collectdebug |
| 01:43.24 | infobot | hmm... collectdebug is a method of collecting logs allowing others help troubleshoot an issue. Refer to https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 01:43.45 | pabelanger | f2Knight: ^ collect a debuglog of your IAX2 call to 216.207.245.8, then pb it |
| 01:44.03 | pabelanger | besure to 'iax2 set debug on' |
| 01:44.28 | f2Knight | pabelanger, let me do that. |
| 01:46.18 | f2Knight | pabelanger, on the logger would full be too much? |
| 01:46.33 | pabelanger | no |
| 01:50.10 | f2Knight | pabelanger, okay just made the dump, uploading it to pastbin now.. the full dump or do you want just a portion? |
| 01:52.40 | fauxalliance | tries the new caffeinated brew |
| 01:53.12 | pabelanger | f2Knight: full log is fine |
| 01:54.04 | f2Knight | pabelanger, too big for pastebin :( |
| 01:56.06 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 01:57.56 | f2Knight | pabelanger, http://pastebin.com/J3tt4NMk |
| 01:58.14 | f2Knight | pabelanger, start and end of call posted stopped at the CDR write |
| 02:00.22 | pabelanger | f2Knight: k, not sure information exposed. This looks to be the issue I was thinking of. Create a new issue on the tracker and we can triage it from there. It will likely not be before Monday until I can find out which version of Asterisk is running on that host |
| 02:01.41 | f2Knight | pabelanger, sorry not sure how to do the tracker thing or where it even is. |
| 02:01.44 | f2Knight | ~tracker |
| 02:01.52 | pabelanger | https://issues.asterisk.org |
| 02:06.56 | f2Knight | pabelanger, damn, need to make an account and all ughh. |
| 02:11.40 | *** join/#asterisk Elijah_ (~elijah@184-98-245-91.phnx.qwest.net) |
| 02:17.41 | *** join/#asterisk JAMMAN2110 (~JAMMAN211@unaffiliated/jamman2110) |
| 02:19.10 | f2Knight | pabelanger, thanks submitted it |
| 02:40.24 | *** join/#asterisk Mark_ (~mark@unaffiliated/morkbork) |
| 02:41.46 | Mark_ | is there a 'proper' way to bypass a trunk when a user dials a DID on the same asterisk setup |
| 02:42.08 | Mark_ | i.e. user too lazy/stupid to dial extension, dials the external number instead |
| 02:47.39 | Mark_ | what ive tried is creating an iax2 peer to loopback |
| 02:47.48 | Mark_ | and creating an outbound route for the dids |
| 02:47.52 | Mark_ | but that seems convoluted |
| 02:48.16 | Mark_ | and its not working :P |
| 02:56.14 | *** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk) |
| 03:11.34 | *** join/#asterisk docid (~PISSS@69.196.68.146) |
| 03:12.46 | docid | so, im trying to keep an iax2 trunk up over a low bandwith/high packet loss satalite link..... but it keeps going unreachable... how do i increase the latency limit to say, 20000ms? |
| 03:13.07 | docid | (yes thats the right amount of 0's) |
| 03:16.03 | devdvd | why? at that point no call is gonna be usable |
| 03:17.03 | docid | because the whole town is cut off and were trying to cowboy it |
| 03:17.12 | devdvd | ah |
| 03:17.13 | docid | huge blizzard |
| 03:17.19 | docid | gunna be down for a week |
| 03:17.29 | docid | were up in the arctic, remote communities |
| 03:17.34 | devdvd | got ya |
| 03:18.13 | docid | i can get it to stay up and be fully usable till about 2000 |
| 03:18.27 | docid | sounds surprizingly good |
| 03:18.29 | docid | actually |
| 03:19.14 | pabelanger | docid: qualify=20000 > |
| 03:19.20 | pabelanger | s/>/?/ |
| 03:19.48 | docid | ahhh, i tryed that, it just goes unreachable |
| 03:19.49 | devdvd | yea, was about to say that :) |
| 03:19.52 | devdvd | oh |
| 03:19.54 | docid | ok, i thought i was in the right place |
| 03:20.01 | devdvd | try qualify no |
| 03:20.04 | devdvd | see if that helps |
| 03:20.10 | *** join/#asterisk jkroon (~jkroon@dsl-242-9-62.telkomadsl.co.za) |
| 03:20.17 | docid | then i get auth failures on the other side, which is REALLY confusing |
| 03:20.24 | pabelanger | docid: Also check out qualifyfreqok and qualifyfreqnotok |
| 03:20.34 | docid | i was jsut making sure the qualify line is the right place to set that |
| 03:20.44 | docid | thanks :) |
| 03:21.01 | *** join/#asterisk Eldonn (~Eldonn@S0106e0cb4e3ba016.cg.shawcable.net) |
| 03:21.06 | docid | ill dig some more, gotta run, but thanks for the hhelp |
| 03:21.11 | *** part/#asterisk Eldonn (~Eldonn@S0106e0cb4e3ba016.cg.shawcable.net) |
| 03:21.13 | devdvd | btw, im currently reading the asterisk definitive guide on ofps.oreilly...whoever done the drawing on the first page of chapter 2....my hats off to you :) |
| 03:26.26 | devdvd | hey, i am having a big issue with my sip phones (polycom sp 221/321) where dial tones aren't being picked up right by the receiving end in most cases. for example if i press 123123 the remote end may receive that as 122233212 |
| 03:26.41 | devdvd | any thoughts on how to fix that |
| 03:36.15 | *** part/#asterisk greggft (~GReggFT@fw1-p1.lexi.com) |
| 04:06.41 | *** join/#asterisk Mhaddog_Mac (~anonymous@z65-50-116-17.ips.direcpath.com) |
| 04:33.29 | *** join/#asterisk bjhaid (~abejide@41.206.12.19.vgccl.net) |
| 04:53.46 | *** join/#asterisk tyman_ (~tyler@173-12-219-179-Fresno.hfc.comcastbusiness.net) |
| 04:54.31 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 05:09.31 | Mark_ | is there anyway to achieve the behavior the user in this post was looking for |
| 05:09.32 | Mark_ | http://fonality.com/trixbox/node/37866 |
| 05:09.37 | Mark_ | without the loopback nonsense |
| 05:09.50 | Mark_ | (im not using trixbox or anything, the post just describes what im after) |
| 05:10.48 | *** join/#asterisk coppice (~chatzilla@60.157.17.210.dyn.pacific.net.hk) |
| 05:19.13 | p3nguin | mark_: What part are you trying to achieve? |
| 05:19.34 | Mark_ | the part where a lazy/stupid user can dial the external user |
| 05:19.38 | Mark_ | err |
| 05:19.44 | Mark_ | external number/did |
| 05:19.53 | Mark_ | and not use the trunk |
| 05:20.58 | Mark_ | i.e. theres an inbound route for (123) 456-7890 to extension 1234 |
| 05:21.32 | Mark_ | if an internal user dials that number, they just get sent to 1234 instead of going out the trunk and back in |
| 05:22.02 | Mark_ | i tried making a dialplan to capture the numbers that are 'local' to the asterisk system but i dont know what to do with them |
| 05:25.36 | p3nguin | First of all, you're going to need to get on an Asterisk level. An "inbound route" for 1234567890 is nothing more than extension 1234567890 in an arbitrarily-named context which is assigned to the peer that sends calls in to Asterisk. |
| 05:27.01 | Mark_ | so i could just create another context |
| 05:27.08 | Mark_ | and do what i want pretty trivially? |
| 05:27.34 | p3nguin | If you want the people that use phones which make calls through Asterisk to be able to call the full DID number, give the phones access to the context where those extensions exist. |
| 05:27.58 | p3nguin | It's basic dial plan logic. |
| 05:29.16 | p3nguin | On my system, phones have a context of phones, oddly enough. Within the [phones] context, I include other contexts that have useful extensions inside them. I would simply include the inbound context before my outbound context. |
| 05:29.25 | Mark_ | aha |
| 05:30.29 | p3nguin | Make sure you don't include the outbound context in the inbound context, though. That could potentially be an expensive mistake. |
| 05:30.55 | Mark_ | yea ive been reading about a lot of pitfalls |
| 05:31.01 | Mark_ | thankfully this is just a test setup |
| 05:32.58 | p3nguin | But what you wanted to do is as easy as including the inbound context in the phones' context before including the outbound context. Done. Enjoy. |
| 05:33.25 | Mark_ | i was hoping it would be something like that |
| 05:33.27 | Mark_ | thank you very much |
| 05:34.11 | p3nguin | I'm not real sure why that person was complaining about an endless loop. |
| 05:34.29 | p3nguin | I mean, I could create an endless loop, but I don't know why I would want to. |
| 05:41.49 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 06:09.07 | *** join/#asterisk titter (~Justin@c-98-208-153-148.hsd1.fl.comcast.net) |
| 06:10.31 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 06:14.02 | *** join/#asterisk titter (~Justin@c-98-208-153-148.hsd1.fl.comcast.net) |
| 06:24.33 | devdvd | hi p3nguin, i got chan_sccp-b up, got my phone registered to asterisk. I can make calls to other extensions in the phones context, and receive calls as well. However. I cannot receive audio on the cisco phone although it sends just fine. When I dial my echo extension on asterisk, here is what i see on the output |
| 06:24.34 | devdvd | [Feb 25 22:24:17] WARNING[6917]: sccp_pbx.c:786 sccp_pbx_write: SEP001B54E0045F: Can't send 0 type frames with SCCP write on channel SCCP/898-0000000e |
| 06:24.37 | devdvd | thoughts? |
| 06:25.08 | devdvd | the phone is behind a nat but i have it in the DMZ |
| 06:26.52 | devdvd | http://pastebin.com/WG41cQ5T |
| 06:26.56 | devdvd | there's my sccp.conf |
| 06:35.30 | *** join/#asterisk ShadowHntr (~sentinel@wikipedia/Shadowhntr) |
| 06:39.38 | *** join/#asterisk benngard (~mabe@h30n2-g-ml-a11.ias.bredband.telia.com) |
| 06:54.20 | *** join/#asterisk Tim_Toady (~moi@77.49.3.6.dsl.dyn.forthnet.gr) |
| 06:55.07 | raden | p3nguin, what was that program that allowed the mouse and keyboard to jump between windows and linux seemlessly ? |
| 06:56.25 | p3nguin | devdvd: I guess if you knew my thoughts about DMZ, you wouldn't have brought it into the equation. |
| 06:56.30 | p3nguin | raden: synergy |
| 06:58.05 | raden | thanks bro ( does it work with win7 ? ) |
| 06:59.01 | devdvd | p3nguin, you're probably right. I pretty much exhausted all I could think of, DMZ was kinda the last option. |
| 06:59.08 | devdvd | one other bit of info for ya |
| 06:59.47 | devdvd | i called another extension, started talking, i could hear what i said into the sccp phone on the sip phone, but not the other way around, at the end of the call i see tis |
| 06:59.49 | devdvd | this |
| 07:00.07 | devdvd | <PROTECTED> |
| 07:00.38 | p3nguin | I'm not so sure "progress" is a valid option for earlyrtp. I'd comment out that setting. |
| 07:00.45 | devdvd | ok |
| 07:01.00 | devdvd | the sample config says its the default |
| 07:01.38 | p3nguin | Not in my samples. |
| 07:01.38 | devdvd | commented out and reloading phone |
| 07:01.44 | p3nguin | earlyrtp = none ; valid options: none, offhook, dial, ringout. default is none. |
| 07:02.07 | devdvd | earlyrtp = progress ; valid options: none, offhook, dial, ringout and progress. default is progress. |
| 07:02.19 | p3nguin | Which version/build did you get? |
| 07:02.23 | devdvd | is what it says in mine. I got the newest from svn |
| 07:02.30 | devdvd | just today |
| 07:03.03 | devdvd | im gonna see what happens with it commented out then im gonna change it to none see what that does |
| 07:03.50 | benngard | morning folks |
| 07:03.58 | p3nguin | I pulled a sample from the newer version and it's the same as yours. |
| 07:04.26 | p3nguin | I know I already asked, but I forgot the answer... is your Asterisk behind a NAT? |
| 07:04.49 | devdvd | no |
| 07:04.58 | p3nguin | That's what I thought. |
| 07:05.40 | ShadowHntr | hey anyone care to recommend an asterisk gui that i could try out that can install on a standard linux install (something that isn't a seperate appliance, such as elastix, asterisknow, etc.) |
| 07:06.00 | ShadowHntr | i'm considering freepbx, but if anyone's had better luck with others... |
| 07:07.07 | devdvd | another odd thing p3nguin is that even with the ringer set to its highest value |
| 07:07.20 | devdvd | s/ringer/ringer volume/ |
| 07:07.31 | devdvd | the phone won't ring |
| 07:07.33 | devdvd | it will vibrate |
| 07:07.35 | devdvd | but not ring |
| 07:08.14 | p3nguin | I have to get another phone I can take out of the office, then I am going to test chan_sccp-b through two NATs. |
| 07:09.06 | p3nguin | I hope to get it tomorrow. I'd really like to be able to test this fully. |
| 07:09.49 | p3nguin | shadowhntr: You can install the Asterisk GUI if you don't like FreePBX, but support for either one will be lacking in this channel. |
| 07:10.27 | ShadowHntr | p3nguin: not a problem, just thought i'd put out some feelers. i'm pretty new to asterisk, and i'd like the assistance of a gui tool, whether it's a web interface or an application on the host |
| 07:11.48 | p3nguin | shadowhntr: Unless you have some reason to not use AsteriskNOW, it's really a great OS that you should consider. It gives you the choice of FreePBX and Asterisk GUI at install time. |
| 07:12.14 | ShadowHntr | p3nguin: i appreciate the offer, and if i weren't making my server multi-purpose, i'd use asterisknow in a heartbeat. :) |
| 07:12.42 | ShadowHntr | i guess i'll try freepbx and see how that works out |
| 07:13.39 | devdvd | p3nguin, thoughts on the ringing thing? think it might be a defective phone? |
| 07:14.06 | p3nguin | shadowhntr: It'll work out almost exactly like if you used AsteriskNOW, I'd bet. AsteriskNOW simply throws it all together so you don't have to. |
| 07:14.14 | *** join/#asterisk joako_ (~joako@opensuse/member/joak0) |
| 07:15.15 | p3nguin | devdvd: The chance of a defective phone is really low (unless you bought a used phone, then it's only slightly higher). It's most likely a problem with the NAT that some testing will eventually uncover. |
| 07:15.31 | devdvd | ok, thats kinda my thought to, was a used phone. |
| 07:16.39 | devdvd | the good news is, there seems to be fairly complete communication between the phone and asterisk |
| 07:16.50 | devdvd | i can send messages to it, reboot it, call it, etc |
| 07:16.54 | devdvd | just cant hear anything :) |
| 07:17.17 | p3nguin | It sounds like sccp works just fine, but rtp is having a problem. |
| 07:17.30 | p3nguin | This is a classic SIP NAT issue. |
| 07:17.38 | joako_ | If it's a bad phone then probably the speakerphone works. LOL I have a grandstrem that does that... worst hardware ever |
| 07:18.58 | devdvd | joako_, I agree with you there :), sadly dont have a speaker phone. Has a headset jack but i dont have a 2.5mm handy |
| 07:20.56 | *** join/#asterisk lfod- (lfod@unaffiliated/lfod-) |
| 07:21.04 | ShadowHntr | well i'm going to head to bed |
| 07:21.09 | ShadowHntr | thanks for the pointers p3nguin |
| 07:21.10 | ShadowHntr | :) |
| 07:22.00 | devdvd | p3nguin, what's odd is i have another sip phone on my network. (i unplugged it cuz i thought it may be causing interference but the same audio issues occurred) |
| 07:22.21 | p3nguin | That supports my thoughts. |
| 07:22.28 | devdvd | about the nat issue? |
| 07:22.36 | p3nguin | yes |
| 07:22.45 | devdvd | now the question becomes what to do about it :) |
| 07:22.51 | p3nguin | Well, maybe I misunderstand what you just said... |
| 07:23.08 | devdvd | which part? |
| 07:23.11 | p3nguin | The SIP phone also exhibits the same behavior? Or is the SIP phone working? |
| 07:23.21 | devdvd | sip phone is working |
| 07:23.58 | p3nguin | Oh, then I misunderstood your statement. Okay, so tell me what you've done to try to get the SIP phone up and running. I have a theory. |
| 07:25.07 | devdvd | nothing beyond the configuring of the phone via ftp and the asterisk configuration, i plugged it in and it just worked |
| 07:26.26 | p3nguin | You didn't play around with network configuration at all? |
| 07:26.34 | devdvd | didn't have to |
| 07:29.11 | p3nguin | Do an rtp debug on Asterisk and make a call using that phone. |
| 07:30.15 | devdvd | actually wait...just remembered something. I didn't do anything in the network config BUT i did this in the sip.cfg for the phone |
| 07:30.16 | devdvd | tcpIpApp.port.rtp.forceSend="10000" tcpIpApp.port.rtp.mediaPortRangeStart="10000" /> |
| 07:30.21 | devdvd | forcing all rtp over 1 port |
| 07:30.38 | p3nguin | and over TCP? |
| 07:30.46 | devdvd | think so yep |
| 07:30.57 | p3nguin | That's strange. |
| 07:31.48 | devdvd | maybe im misinterpreting what i done...but i now remember why i made the change. Was having one way audio issues cuz of the janky nat |
| 07:32.02 | devdvd | so i had a thought of pushing all audio over 1 port |
| 07:33.10 | p3nguin | You've got a Belkin router? |
| 07:33.14 | devdvd | no |
| 07:33.37 | devdvd | its an old moto router with dd-wrt on it |
| 07:33.47 | devdvd | i have the rtp debug running and have an active call going |
| 07:34.10 | p3nguin | Belkin is the top 5 on my "Do not buy if you ever want to use VoIP" list. |
| 07:34.43 | devdvd | Belkin is on my top 5 do not buy if you want your internet to work at all list |
| 07:34.59 | p3nguin | Actually it's the top 5 on my "do not buy" list, but wanting to use VoIP just enforces it. |
| 07:35.04 | devdvd | aye |
| 07:35.19 | p3nguin | Oh man, that reminds me... |
| 07:35.39 | p3nguin | My mother bought a Belkin router for her cable modem service. |
| 07:35.43 | p3nguin | I was so pissed. |
| 07:35.44 | devdvd | eeek |
| 07:35.47 | devdvd | bad mom |
| 07:36.17 | devdvd | i forbid my dad or my wife's parents from buying any technology without my consent for that very reason |
| 07:36.32 | devdvd | for that matter my wife is forbid from it as well |
| 07:36.36 | p3nguin | I wasn't nearly as upset that she bought the Belkin as I was that she didn't ASK ME before she bought it. |
| 07:37.34 | p3nguin | ... also that she bought an HP laptop from Staples. |
| 07:37.39 | *** join/#asterisk doolittlework (~doolitlew@41-134-22-10.dsl.mweb.co.za) |
| 07:38.03 | devdvd | *facepalm* |
| 07:38.10 | p3nguin | I don't have anything against HP, nor against Staples, but what kind of tech person buys an HP from Staples? |
| 07:38.14 | devdvd | let me guess...another purchase you were not consulted about |
| 07:38.39 | p3nguin | Not so much as a phone call. |
| 07:39.07 | doolittlework | i ask if asterisk is a good solution for creating a hosted pabx solution, and then my internet dropped so i dont know what all your thoughts was in this regards, so here goes is asterisk a good solution for creating a hosted pabx? |
| 07:39.08 | p3nguin | Buying a computer from Staples is almost like buying a computer at Best Buy. |
| 07:39.18 | devdvd | thats almost as bad as my dad buying a dell from walmart |
| 07:39.42 | p3nguin | Does Walmart fiddle with their retail computers like Staples and Best Buy? |
| 07:39.49 | joako_ | Staples has been strongly advertising their computer repair services! |
| 07:40.09 | devdvd | p3nguin, yea they do |
| 07:40.14 | p3nguin | doolittlework: Yes, it is a good tool for creating a PBX. |
| 07:40.17 | devdvd | though not as bad |
| 07:40.18 | p3nguin | ~pbx |
| 07:40.18 | infobot | well, pbx is a Private Branch eXchange |
| 07:40.23 | p3nguin | ~asterisk |
| 07:40.23 | infobot | Asterisk is an open source telephony toolkit, or #asterisk on irc.freenode.net, or http://www.asterisk.org/ |
| 07:40.29 | p3nguin | hmm |
| 07:40.36 | devdvd | ~p3nguin |
| 07:40.38 | p3nguin | Those weren't the ones I was looking for. |
| 07:41.17 | devdvd | p3nguin, any thoughts on how i can force rtp over 1 port with sccp? |
| 07:41.27 | p3nguin | I was looking for the one that says that Asterisk is not a PBX, but is a toolkit used to develop a PBX. |
| 07:42.15 | p3nguin | devdvd: Uh, I just saw that you have directrtp on. |
| 07:42.17 | p3nguin | bad |
| 07:42.44 | devdvd | actually that's off now |
| 07:42.57 | devdvd | i thought that might be an issue so i turned it off |
| 07:43.01 | p3nguin | Still not working though? |
| 07:43.15 | p3nguin | directrtp means the phone can reinvite and kick Asterisk out of the media path. That's bad when NAT is involved. |
| 07:43.38 | devdvd | aye |
| 07:45.16 | doolittlework | thanks |
| 07:45.37 | doolittlework | what is latest stable version for astertisk? |
| 07:45.49 | devdvd | ok, wife is nagging me to come to bed |
| 07:45.55 | devdvd | so ill pick this up tomorrow |
| 07:45.58 | p3nguin | There are three branches. Which branch are you referring to? |
| 07:46.01 | devdvd | thank you again for your help p3nguin |
| 07:46.15 | p3nguin | I'll test sccp/nat as soon as I get a chance. |
| 07:46.26 | p3nguin | hopefully tomorrow |
| 08:22.07 | *** join/#asterisk emora (~emora@213.37.32.74.static.user.ono.com) |
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| 08:27.31 | *** join/#asterisk eject_ck (~eject_ck@83-218-246-188.dynamic.vega-ua.net) |
| 08:27.58 | eject_ck | there is any console sip clients ? |
| 08:28.07 | eject_ck | there are any console sip clients ? |
| 09:01.59 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
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| 11:24.18 | *** join/#asterisk daxt (~daxt@112.135.72.127) |
| 11:24.49 | daxt | guys i am having an error in my asterisk - "because extension not found in context 'default'" <-- how do i fix this ? |
| 11:24.55 | *** join/#asterisk Blaguvest (~Blaguvest@unaffiliated/blaguvest) |
| 11:25.17 | ChannelZ | make the correct extension in that context or set your phones to the right context |
| 11:25.49 | *** part/#asterisk Blaguvest (~Blaguvest@unaffiliated/blaguvest) |
| 11:26.58 | ChannelZ | IE a SIP device for instance is set to context=default - when you pick it up and dial 555 (or whatver, you didn't say) Asterisk tries to find a matching extension in the "default" context of extensions.conf |
| 11:27.58 | ChannelZ | It's complaining that extension doesn't exist in that context. |
| 11:27.59 | daxt | in my extensions.conf only [test] context is there , and in sip.conf 'context=test' in there too. |
| 11:28.40 | daxt | there is no context or even word of "default" in both sip.conf and extensions.conf :) |
| 11:28.52 | ChannelZ | reload or restart asterisk, if you've changed your sip.conf while it was running |
| 11:29.04 | daxt | i use reload in cli |
| 11:29.11 | daxt | isnt it enough ? |
| 11:29.44 | ChannelZ | should be. |
| 11:30.19 | ChannelZ | What exactly are you doing? picking up the phone and dialing an extension and getting that error? |
| 11:30.41 | ChannelZ | you didn't paste the entire line to show what the error was a result of |
| 11:31.30 | daxt | http://pastebin.com/CmrCeB9A |
| 11:32.08 | daxt | "chan_sip.c:20314 handle_request_invite: Call from '' to extension '1000' rejected because extension not found in context 'default'." |
| 11:33.30 | fenrus | did you reload? |
| 11:33.34 | ChannelZ | Ah. So whatever you're dialing from is not matching your peer 1000 |
| 11:34.24 | daxt | when i change "test" to "default" dial plan works and i can hear "helloworld" :) |
| 11:34.41 | ChannelZ | has that device registered to asterisk successfully? (what does 'sip show peers' show?) |
| 11:35.13 | daxt | 1000/dan (Unspecified) D 5060 Unmonitored |
| 11:35.19 | ChannelZ | see |
| 11:36.09 | daxt | what is wrong here ? |
| 11:36.14 | ChannelZ | * doesn't know 1000's IP address and isn't matching that peer from sip.conf when you make a call |
| 11:36.19 | *** join/#asterisk Sertys (~sertys@89.252.247.42) |
| 11:36.38 | daxt | host=dynamic |
| 11:36.48 | ChannelZ | In which case it's an "'anonymous" SIP call which defaults to the 'default' context (unless you turn it off or change it in the [general] section of sip.conf) |
| 11:37.10 | daxt | ahaa |
| 11:37.15 | daxt | now we are getting there |
| 11:37.47 | ChannelZ | yes you set host=dynamic but your device isn't registering (thus 'Unspecified' for the IP) |
| 11:38.11 | daxt | what shud i add to [general] ? |
| 11:38.37 | ChannelZ | nothing necessarily |
| 11:38.42 | ChannelZ | that isn't your problem |
| 11:38.50 | ChannelZ | your immediate one anyway. |
| 11:40.24 | daxt | i want to know why * is so adamant with default contexts :) |
| 11:40.51 | ChannelZ | because you misconfigured your sip peer |
| 11:41.09 | daxt | how shud i properly config the sip peer ? |
| 11:41.24 | daxt | arent my sip.conf right ? |
| 11:41.29 | ChannelZ | the entry [1000] in sip.conf is doing nothing because whatever you're actually dialing from is not matching that peer. |
| 11:41.43 | daxt | so i need another peer ? |
| 11:41.45 | daxt | like 1001 |
| 11:41.59 | ChannelZ | Because you told Asterisk it has a dynamic IP, but the phone has never registered with Asterisk for it to know what its IP actually is. |
| 11:42.14 | daxt | still confused |
| 11:42.51 | ChannelZ | You've created a user, but nobody is logging in as it. |
| 11:42.59 | ChannelZ | to make a comparison. |
| 11:43.09 | daxt | hmm |
| 11:43.24 | daxt | weird |
| 11:43.39 | daxt | coz when i make the call user logs in right ? |
| 11:44.08 | ChannelZ | When Asterisk gets an incoming SIP call it tries to match it to a sip peer (in sip.conf) based on a few different methods. In your case it's matching none of them |
| 11:44.41 | ChannelZ | You phone isn't registering with Asterisk at all. Whether it's because you've told it not to, or because the phone is misconfigured I don't know |
| 11:45.02 | daxt | i am using linphone |
| 11:45.18 | daxt | sip:dan@190.120.226.84 |
| 11:45.30 | daxt | this is what added to linphone's config |
| 11:45.40 | ChannelZ | so you've placed an 'anonymous' call to Asterisk as I said |
| 11:45.51 | *** join/#asterisk DomeDan (~DomeDan@78-73-41-163-no184.tbcn.telia.com) |
| 11:45.58 | ChannelZ | it's unauthenticated. |
| 11:46.10 | daxt | in sip.conf i have set username=dan as well |
| 11:46.14 | daxt | but no password |
| 11:46.26 | daxt | if i add a password , then it shud work right ? |
| 11:47.02 | ChannelZ | I don't know, never used linphone so I have no idea what it's config is like |
| 11:47.50 | ChannelZ | you should tell it to register as '1000' with no password, the way you're configured now |
| 11:49.16 | *** join/#asterisk zenobix (~zenobix@p578EDC7F.dip.t-dialin.net) |
| 11:49.50 | daxt | you are right ChannelZ - i get this error "chan_sip.c:21819 handle_request_register: Registration from '<sip:dan@190.120.226.84>' failed for '112.135.72.127' - No matching peer found" |
| 11:50.02 | daxt | i tried another softphone |
| 11:50.22 | *** join/#asterisk esterpader (~esterpade@58.69.18.6) |
| 11:50.25 | ChannelZ | so either change your peer name to [dan] or make the phone use '1000' |
| 11:52.22 | daxt | oh ya |
| 11:52.23 | daxt | now it works |
| 11:52.35 | daxt | i changed [1000] to [dan] in sip.conf |
| 11:52.50 | daxt | now * behaves the way i thought :) |
| 11:52.54 | daxt | thanks ChannelZ |
| 11:54.20 | ChannelZ | one of the worst parts (IMHO) of * is peer matching and the difference between device names and extensions, a lot tutorials and such blur the two and don't really explain that they have nothing to do with each other |
| 11:54.35 | daxt | exactly |
| 11:54.53 | daxt | i had severe pains in getting Asterisk running |
| 11:55.05 | daxt | coz documentation was not spot on as it shud be |
| 11:55.58 | ChannelZ | bed time. have fun |
| 11:56.05 | daxt | ur from ? |
| 11:56.18 | daxt | u too |
| 11:56.19 | ChannelZ | CO USA |
| 11:56.25 | daxt | i am from Sri Lanka |
| 11:56.39 | ChannelZ | almost 5am here, should have been in bed hours ago |
| 11:57.33 | daxt | yup |
| 11:57.45 | daxt | sleeping in morning is not good for your veins |
| 11:58.01 | esterpader | Good day guys, has anyone from you has an issue of transcoding ulaw to g.729?, the server is power enough to do the transcoding but it always have a clip |
| 11:58.39 | esterpader | but setting g.729 all the way from end to end fix the issue. |
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| 12:51.21 | *** join/#asterisk jsjc (~jsjc@226.Red-80-33-236.staticIP.rima-tde.net) |
| 12:53.33 | jsjc | I am having a little issue, well perhaps big.... I have a phone conneced trough my dahdi card and after a reboot of the machine where the asterisk goes, hungs up when I dial a number. |
| 12:53.59 | jsjc | I open the asterisk shell to check any log but when pressing the number t call just hangs up. |
| 12:54.10 | jsjc | what coudl eb causing this? how could I troubelshoot? |
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| 13:18.36 | *** join/#asterisk jan_bangna (~jandetlef@ppp-124-120-170-187.revip2.asianet.co.th) |
| 13:21.37 | jan_bangna | Hello, i was reading on running Asterisk in the cloud, i.e. on Amazon EC2 but haven't seen details on how to get actual phone lines (aka numbers to call) connected to it. Can anyone recommend resources to read that explain it? |
| 13:36.16 | esterpader | jsjc, what does dmesg show? |
| 13:38.02 | russellb | jan_bangna: you sign up for an account with an ITSP (Internet Telephony Service Provider), and all of your calls to and from the telephone network are just over IP. |
| 13:40.33 | jan_bangna | russellb thank you that makes sense now |
| 13:40.48 | russellb | you're welcome |
| 13:41.14 | russellb | There are many ITSP choices available. |
| 13:46.26 | jan_bangna | do you know how many phonecalls one amazon EC2 instance can switch? trying to think of some solutions for a callcenter with 300 lines. |
| 13:51.42 | russellb | a bunch? I'm sure the power you get is something you can choose |
| 13:52.03 | russellb | and of course, the nice thing about deploying to something like EC2 is how trivial it is to add additional capicity (more servers) if needed |
| 13:52.11 | leifmadsen | isnt' EC2 based on CPU consumption and bandwidth? |
| 13:52.20 | leifmadsen | i.e. the more you use, the more you pay |
| 13:52.54 | leifmadsen | which I presume translates to pretty much as much capacity as you want to pay for -- pretty sure that's the whole point of cloud infrastructure :) |
| 13:53.32 | russellb | THE CLOUD |
| 13:54.34 | jan_bangna | cool, that makes sense, thanks for the replies. |
| 13:55.02 | leifmadsen | russellb: :) |
| 14:00.12 | *** join/#asterisk espiceland (~erin@nat/digium/x-myzwuoczwfntsisu) |
| 14:08.43 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 14:27.55 | jsjc | esterpader what you mean by dmesg? |
| 14:27.59 | jsjc | why shoudl dmesg show anything? |
| 14:28.42 | jsjc | dmest shows nothing.... |
| 14:50.34 | *** join/#asterisk blitzrage (~blitzrage@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:50.35 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 14:55.25 | niekie | Anyone know a test iNum I can reach? |
| 14:58.48 | niekie | Never mind :) |
| 14:59.56 | *** join/#asterisk Olivier_54 (~Olivier_5@vaillant.famille-fontes.net) |
| 15:02.45 | *** join/#asterisk Denial (Denial@drgi.co.uk) |
| 15:03.13 | esterpader | jsjc, that's a kernel message |
| 15:03.51 | esterpader | jsjc, i happen me once on a non digium card, it has something to do with PCI abort master something of that sort |
| 15:04.46 | esterpader | jsjc, restart dahdi then check for dmesg |
| 15:11.36 | zenobix | hi, we had an weird misdn issue last night...misdn deadlock. |
| 15:12.23 | zenobix | set a callerid(num) to string and playback results in deadlock, we solved it, to just to set the callerid(name). |
| 15:26.58 | *** join/#asterisk awclin_ (~alinford@g0962184.demon.co.uk) |
| 15:30.10 | jsjc | esterpader no issues with dmesg, I am using an non digium card as well but was all working nicely til machine was rebooted. It is funny that some extensions are working it is just other that is configured in another context, much longer cable to the phone so I dont know where to start... |
| 15:32.06 | jsjc | when I call another extension that has very short numer such us 101 works |
| 15:32.13 | jsjc | but as soon as I try to diual something longer |
| 15:32.24 | jsjc | hangs on the way. |
| 15:33.12 | jsjc | now not even 101.. dont know where to start troubleshooting this.. |
| 15:35.24 | zenobix | jsjc: do you already tried overlapdial or waitfordigits |
| 15:36.00 | *** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk) |
| 15:41.57 | jsjc | zenobix I am going to check that. |
| 15:45.23 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 15:45.31 | jsjc | zenobix as soon as I try to press any number I hear a hangup tone. all I get from asterisk is this -- Starting simple switch on 'DAHDI/6-1' -- Hungup 'DAHDI/6-1'. And All I have done in between is dial a number. |
| 15:45.38 | jsjc | i am with verbose 7 |
| 15:46.44 | jsjc | cannot understand because is an outgoing call. So should at least give me something... |
| 15:47.07 | jsjc | just seems like asterisk is doing all good but I dont know where the error could be because this stuff was working before last reboot. |
| 15:47.26 | zenobix | do you have in dadhi.conf overlapdial=yes ? |
| 15:48.51 | *** join/#asterisk sourcode (~code@ppp-58-8-163-126.revip2.asianet.co.th) |
| 15:49.32 | jsjc | dahdi.conf? chan_dahdi? or /etc/dahdi/system.conf ? |
| 15:49.59 | jsjc | cannot see overlapdial |
| 15:54.13 | *** join/#asterisk cerberus_za (~coert@196-215-44-48.dynamic.isadsl.co.za) |
| 15:56.35 | cerberus_za | hello all, |
| 15:56.44 | cerberus_za | i am an asterisk newbie |
| 15:57.01 | cerberus_za | i have the voip part running with ekiga and a grandstream phone |
| 15:57.55 | cerberus_za | now I would like to add my TDM410P to the mix. I have one FXO module and one FXS |
| 15:58.17 | cerberus_za | I have configured the dahdi system.conf |
| 15:58.36 | cerberus_za | and also /etc/asterisk/dahdi_channels.conf and chan_dahdi.conf |
| 15:59.05 | cerberus_za | could anyone point me in the direction of more manuals on the setup of asterisk for those channels? |
| 15:59.20 | cerberus_za | google is not being tremendously helpful currently |
| 16:04.46 | cerberus_za | I am running gentoo with asterisk 1.6.2.16.2-r1 |
| 16:05.06 | cerberus_za | and dahdi 2.4.0-r1 |
| 16:07.27 | *** join/#asterisk spas (~spascoe@cpe-24-166-5-225.indy.res.rr.com) |
| 16:19.40 | *** join/#asterisk endemic (~endemic@2001:49f0:400e::5) |
| 16:23.36 | jsjc | is there anyway to get in aserisk shell print of any numer dialed by a phone at realtime? si I can see if there is any issue with dtmf. |
| 16:25.04 | *** join/#asterisk Tim_Toady (~moi@77.49.3.6.dsl.dyn.forthnet.gr) |
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| 16:42.23 | zenobix | jsjc: set the debug (level) of the channel on |
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| 16:46.56 | *** part/#asterisk jmordica (~jmordica@c-68-63-214-171.hsd1.ms.comcast.net) |
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| 16:57.55 | retentiveboy | I'm replacing a circa-2005 * machine with a new 1.8 one today. Having trouble with the SIP carrier. |
| 16:58.20 | retentiveboy | Registers ok but rejects inbound calls. |
| 16:58.39 | retentiveboy | Appears carrier isn't retrying with digest auth. |
| 16:58.57 | retentiveboy | Can I allow non-digest auth? |
| 17:00.10 | *** join/#asterisk awclin (~alinford@g0962184.demon.co.uk) |
| 17:06.09 | retentiveboy | found it. "insecure=invite" |
| 17:06.15 | *** part/#asterisk retentiveboy (~retentive@atl.pra-corp.com) |
| 17:06.29 | cerberus_za | think i found it: https://wiki.asterisk.org/wiki/display/AST/SLA+Trunk+Configuration |
| 17:11.39 | *** join/#asterisk Mhaddog (~Mhaddog@adsl-074-228-037-158.sip.bct.bellsouth.net) |
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| 17:18.09 | *** part/#asterisk zenobix (~zenobix@p578EDC7F.dip.t-dialin.net) |
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| 17:45.30 | daxt | anyone use mydivert here ? |
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| 18:35.47 | *** join/#asterisk as001 (~uros@cable-94-189-241-201.dynamic.sbb.rs) |
| 18:36.34 | as001 | Hi I have problems with Music on Hold from Queue application. CLI says starting music on hold but I can not hear anything |
| 18:36.57 | as001 | music class is ok. When I use it in AgentLogin I can hear music. |
| 18:38.36 | as001 | I also hav enothing on rtp set debug on after Queue.. |
| 18:38.55 | as001 | while in AgentLogin I have rtp traffic.. |
| 18:41.15 | as001 | I am using Asterisk 1.6.2.16.2 |
| 18:43.13 | as001 | and it was the same on 1.6.2.10 so I hoped after recompiling with new version I will get rid of this problem.. |
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| 19:14.36 | as001 | why does not asterisk sends rtp packets for musiconhold in Queue application ? |
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| 20:34.04 | guifort | Hello all |
| 20:36.12 | guifort | I need help for the config of asterisk for the Incoming call the 2 lines come from the same sip provider : http://pastebin.com/wcx7mnjs |
| 20:36.29 | *** join/#asterisk chopp (~chopp@unaffiliated/chopp) |
| 20:43.24 | *** join/#asterisk KingDavidNYC (~Chris1232@pool-71-191-150-2.washdc.fios.verizon.net) |
| 20:43.53 | blitzrage | guifort: I find it easier to just add /line1 and /line2 at the end of each of the registrations and handle them as exten => line1 and exten => line2 in the dialplan |
| 20:44.37 | guifort | Thanks for your answer, I try that |
| 20:45.37 | guifort | For example I put : register => user1:pwd1@ippi.fr/1010 (The number of the line 1 ?) |
| 20:46.39 | *** join/#asterisk myke (myke@block-66.135.80.250.montanasat.net) |
| 20:48.04 | KingDavidNYC | I am trying to integrate SER with Asterisk, but I dont understand if the voice packets aare nnot send to asterisk, how will asterisk know how to capture the dtmf in an ivr? |
| 20:49.37 | Gugge | if dtmf is inband the voice needs to goto asterisk for it to see dtmf |
| 20:50.09 | *** part/#asterisk seanjohn (~jerralegA@gateways.sheltoncomputers.com) |
| 20:51.01 | KingDavidNYC | correct, certainly dtmf is inband, like in any other asterisk call |
| 20:52.02 | Gugge | dtmf does not need to be inband |
| 20:52.16 | KingDavidNYC | Gugge: in asterisk? |
| 20:52.56 | Gugge | asterisk supports rfcsomething and info too |
| 20:53.17 | KingDavidNYC | what us rfcsomething? |
| 20:53.34 | Gugge | go read the docs on dtmfmode in sip.conf |
| 20:53.50 | Gugge | im to lazy to look it up :) |
| 20:54.39 | KingDavidNYC | ok, but you mean that I can have autoattendants in asterisk and not necessarily handle the dtmf inband? |
| 20:55.39 | Gugge | i mean you need to send the audio to asterisk if you use inband dtmf |
| 20:56.54 | KingDavidNYC | I get that. but are you saying that I can have autoattendants in asterisk that handle dtmf tones out of band? |
| 20:57.00 | Gugge | no |
| 20:57.18 | Gugge | im saying that if you use inband dtmf asterisk needs the audio to get the dtmf :) |
| 20:57.19 | Gugge | nothing more |
| 20:57.23 | Gugge | nothing less |
| 20:58.17 | KingDavidNYC | but in other words, all autoattendants in asterisk need to have dtmf inband right? |
| 20:58.23 | Gugge | no |
| 20:58.32 | Gugge | go read the docs on dtmfmode in sip.conf |
| 21:00.27 | KingDavidNYC | ok my friend, but last question: can I have SER send the vocie packets out of asterisk and at the same tiem have asterisk handle autoattendandts with outofband dtmf signals? |
| 21:00.41 | Gugge | not if you use inband dtmf |
| 21:00.43 | Gugge | maybe if you dont |
| 21:01.23 | *** join/#asterisk awclin (~alinford@g0962184.demon.co.uk) |
| 21:01.34 | Gugge | ive never tried reading outofband dtmf tones without having the rtp through asterisk too |
| 21:01.45 | Gugge | but i _know_ you cant read the inband without rtp |
| 21:02.24 | KingDavidNYC | Gugge:mmm... really vague my friend, but I thank you for your answer |
| 21:02.26 | blitzrage | KingDavidNYC: you can handle DTMF if you use out of band DTMF like dtmfmode=info |
| 21:02.46 | blitzrage | if it is inband, then the audio has to go through asterisk to "hear" the dtmf obviously |
| 21:03.18 | KingDavidNYC | blitzrate: I can send and handle dtmf auto of band if I set dtmf=info? |
| 21:03.50 | Gugge | remember the other end of the call needs to support that dtmf mode too |
| 21:04.06 | KingDavidNYC | Gugge: oops |
| 21:04.40 | KingDavidNYC | this all sounds like I am gonna have to send audio to asterisk all the time |
| 21:05.00 | Gugge | if your sip calls use inband audio yes |
| 21:05.29 | KingDavidNYC | I am concern with the auttoattendant calls |
| 21:06.42 | KingDavidNYC | it looks like I am gonna have to tell my customer "look, I will have to keep audio in asterisk becasue otherwise your autoattendant calls are not going to work!" |
| 21:17.45 | *** join/#asterisk as001 (~uros@cable-94-189-241-201.dynamic.sbb.rs) |
| 21:18.28 | as001 | Hi I wonder if someone know do I need sound card on asterisk server to be able to play music on hold in Queue application ? |
| 21:18.42 | WIMPy | no |
| 21:18.44 | as001 | Moh works in Playback Background.. |
| 21:18.49 | as001 | but not in QUeue |
| 21:19.22 | as001 | worksin AgnetLogin too.. I can not figure out why I can not hear music on hold in Queue |
| 21:19.30 | WIMPy | You only need a sound card if you want to use the integrated console phone. i.e. only for testing. |
| 21:19.49 | WIMPy | Did you perhaps set a wrong moh class somewhere? |
| 21:19.58 | as001 | ok.. Thanks. No I use default class |
| 21:20.10 | as001 | I tried with same class AgentLogin and it works.. |
| 21:20.38 | as001 | CLI says Started music on hold, class 'default2', on SIP/55-00015428 |
| 21:20.55 | as001 | but there is no RTP stream on rtp set debug on |
| 21:21.14 | WIMPy | default_2_? |
| 21:21.21 | as001 | yes default2 |
| 21:21.30 | WIMPy | ok |
| 21:21.31 | as001 | with files from default |
| 21:22.19 | as001 | moh show files gives me 5 files in default2 like File: /var/lib/asterisk/moh/macroform-cold_day |
| 21:23.56 | as001 | It is strange Playback can playback files, Background can do it too... AgentLogin too.. just Queue can not |
| 21:25.12 | as001 | I also noticed that on my system I have no /dev/snd ... lspci does not show Audio Multimedia Controler |
| 21:25.50 | WIMPy | That doesn't matter unless you want to connect a headset and try to place calls there. |
| 21:26.14 | as001 | ok.. But why there is no RTP stream in Queue ? |
| 21:26.52 | as001 | I tried with sip and iax and I could not get Moh working |
| 21:27.11 | WIMPy | doesn't know specific things about queues. I think they are a good feature, but nobody wanted that so far. |
| 21:28.16 | as001 | it is nice to play music to clients in Queue who wait for operator. |
| 21:28.29 | WIMPy | sure |
| 21:29.58 | as001 | It works out of the box in earlier versions of asterisk. I never had problems with it until now.. |
| 21:30.41 | as001 | I might miss something very big on my Debian server. |
| 21:33.40 | *** join/#asterisk Olivier_54 (~Olivier_5@vaillant.famille-fontes.net) |
| 21:42.47 | as001 | I just figured out |
| 21:42.56 | as001 | in case anyone have similar problems |
| 21:43.08 | as001 | if you want MOH in queue you must answer first |
| 21:43.25 | as001 | it does not work otherwise now like it did before.. |
| 21:43.36 | WIMPy | That makes sense. |
| 21:44.08 | as001 | :) now i am happy.. |
| 21:44.18 | as001 | thanks for your help WIMPy good night |
| 22:23.05 | phix | ok, what is the offical site for configuring dahdi? I have read a few things but they seem to skip or are vague in some areas |
| 22:23.24 | WIMPy | The sample config. |
| 22:23.32 | phix | I have dahdi picking up my card, however I need to load the module manually and if I restart dahdi it unloads the module but doesn't reload it |
| 22:24.16 | phix | the sample config tells me what I need in a particular conf, it doesnt tell me how it all works together, which files are required and the name of the channels when you refer to them in the extensions.conf file in asterisk |
| 22:25.03 | phix | in asterisk dahdi show status lists my card but dahdi show channels lists nothing |
| 22:25.14 | WIMPy | They do after you use dahdi_genconf. |
| 22:25.27 | phix | which I ran |
| 22:25.46 | phix | still doesnt tell me how to refer to the channels in asterisk |
| 22:25.54 | WIMPy | Then it seems to have gone wrong. |
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| 22:25.59 | phix | I was using zaptel before this |
| 22:26.10 | WIMPy | The way you configured them in chan_dahdi.conf. |
| 22:26.38 | phix | where should this file be? /etc/dahdi? /etc/asterisk ? |
| 22:26.40 | WIMPy | The configuration is pretty much identical. |
| 22:26.54 | WIMPy | except for updated since then. |
| 22:26.54 | phix | pretty much? |
| 22:27.22 | WIMPy | /etc/dahdi/system.conf and /etc/asterisk/chan_dahdi.conf. |
| 22:28.10 | WIMPy | Development didn't stop after it was renamed to dahdi. |
| 22:30.40 | phix | ok yay I now have the file in the correct spot and it picked it up |
| 22:30.59 | phix | now, how do I refer to it in extensions.conf? |
| 22:31.06 | phix | I usedto use ZAP/${CHANNEL} |
| 22:31.16 | phix | what isi tn ow? |
| 22:31.25 | WIMPy | I give you one guess. |
| 22:32.17 | p3nguin | If Zap changed to Dahdi and everything remained the same within the channel technology, that shouldn't be too hard to figure out. |
| 22:33.15 | phix | dahdi/$channel} ? |
| 22:33.24 | WIMPy | bingo! |
| 22:35.42 | phix | <3 |
| 22:49.37 | phix | No hardware timing source found in /proc/dahdi, loading dahdi_dummy |
| 22:50.00 | phix | I have a TDM400p card though :/ shouldn't it getting timing info from that? |
| 22:51.36 | WIMPy | I've got no idea. At least it can't retrieve timing from the network. |
| 22:52.02 | phix | <PROTECTED> |
| 22:52.06 | phix | [Feb 27 09:42:57] WARNING[12809]: app_voicemail.c:5260 leave_voicemail: No entry in voicemail config file for 'u99' |
| 22:52.32 | phix | I am guessing the syntax has changed from asterisk 1.4 to 1.6 ? |
| 22:53.58 | WIMPy | I don't think so. |
| 22:54.13 | WIMPy | But u99 does not look like a valid mailbox number. |
| 22:54.25 | WIMPy | Did you mean 99,u? |
| 23:00.24 | p3nguin | u99 is bad. If the mailbox number is 99, use 99@yourvmcontext. |
| 23:00.45 | p3nguin | u99 would have been 1.2 or earlier syntax. |
| 23:01.31 | *** join/#asterisk thews (~thews@24-119-172-234.cpe.cableone.net) |
| 23:02.07 | WIMPy | That's why I started with a beta version in production that time. I didn't want to involve in obsolete syntax. |
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| 23:15.43 | phix | p3nguin: 1.4 uses that syntax too, that is whtat I upgraded from |
| 23:15.58 | phix | 1.4 -> 1.6 |
| 23:16.15 | phix | ok so I dont have a /proc/dahdi why not? |
| 23:16.21 | phix | do I need one? |
| 23:17.43 | p3nguin | phix: No, 1.4 doesn't use that syntax. If 1.4 _allows_ that syntax, it's to retain the behavior carried over from 1.2 because some people, like yourself, don't bother checking to see if syntax changes. |
| 23:17.58 | p3nguin | 1.4 uses VoiceMail(mailbox[@context][&mailbox[@context]][...][|options]) |
| 23:18.13 | p3nguin | VoiceMail(99@context,u) |
| 23:20.44 | p3nguin | When syntax is deprecated, developers often leave the capability for legacy configurations. When they decide to obsolete it, you'll be forced to change to the new syntax, which you would have had plenty of time to correct in the previous version. |
| 23:21.17 | WIMPy | versionS |
| 23:28.27 | *** join/#asterisk fauxalliance (~fauxallia@142.162.117.116) |
| 23:29.41 | p3nguin | yeah, versions |
| 23:30.03 | harold00 | Hello, first time on a IRC. Is it the right place and method to try to find a solution about my asterisk problem ? Thanks |
| 23:30.44 | WIMPy | ~ask |
| 23:30.45 | 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. |
| 23:33.04 | harold00 | I have a fresh asterisk with GUI, everythings fine inside LAN. trunk register is ok, outbound ok but for inbound invite asterisk allways end 401 not authorized |
| 23:33.20 | harold00 | run out of idea what error to look for |
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| 23:42.23 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 23:54.38 | Tozz_ | drop the GUI. Asterisk GUI's all suck |
| 23:54.43 | Tozz_ | oh, okay! |