| 00:16.30 | *** join/#asterisk pabelanger_ (~pabelange@CPE001fe2a8fd1d-CM0012254094b2.cpe.net.cable.rogers.com) |
| 00:17.06 | pabelanger-lapto | heh |
| 00:21.47 | *** join/#asterisk x303 (~x303@wifi222-144.meruwifi.fit.edu) |
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| 01:00.47 | *** join/#asterisk rustyclarkson (~rusty@u53.sutus.com) |
| 01:01.23 | rustyclarkson | What's the best way to get holdtime into the CDR? |
| 01:02.17 | rustyclarkson | i can only come up with dumb ideas like an in-queue AGI script to run CDR() |
| 01:10.30 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
| 01:12.06 | pabelanger-lap | rustyclarkson: exten => s,n,Set(CDR(holdtime)=1234) |
| 01:14.33 | rustyclarkson | Thanks pabelanger-lap, I understand that I can set the holdtime value in the CDR, but I'm concerned about being able to get the holdtime value while in Queue(), then being able to put that into the CDR. |
| 01:17.17 | pabelanger-lap | rustyclarkson: core show application Queue |
| 01:17.35 | pabelanger-lap | rustyclarkson: look at the macro option |
| 01:17.39 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 01:19.06 | pabelanger-lap | rustyclarkson: then extract the QEHOLDTIME in the macro |
| 01:24.03 | *** join/#asterisk coppice (~chatzilla@19.176.64.202.dyn.pacific.net.hk) |
| 01:26.18 | rustyclarkson | thanks pabelanger-lap, didn't realize macro's were a possibility |
| 01:26.27 | rustyclarkson | doesn't help that we're operating under 1.4.22 :s |
| 01:28.20 | pabelanger-lap | rustyclarkson: Do you need to access the value, while they are still in the queue? If not, you can simply write to the CDR after they drop |
| 01:29.20 | rustyclarkson | pabelanger-lap: I do not need access to it, I just need it in the CDR. |
| 01:29.57 | pabelanger-lap | rustyclarkson: Then set the value after your Queue comment, next priority |
| 01:30.13 | pabelanger-lap | s/comment/command |
| 01:30.30 | rustyclarkson | if the caller hangs up before the queue agent, will the Set(CDR...) still get executed? |
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| 01:37.47 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 01:38.58 | pabelanger-lap | rustyclarkson: core show application Queue |
| 01:39.02 | pabelanger-lap | rustyclarkson: option c |
| 01:39.59 | rustyclarkson | oh my god |
| 01:40.19 | rustyclarkson | the continuous flaws of being on 1.4 |
| 01:40.29 | rustyclarkson | thanks very much for your help pabelanger-lap |
| 01:40.47 | pabelanger-lap | rustyclarkson: np |
| 01:40.52 | rustyclarkson | I've learnt something new with that "core show application" command |
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| 01:51.56 | Sargun | dudes, set +C |
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| 02:05.34 | krahe | hi do I really have to define trunks and Outgoing Calling Rules before inform USERS (SIP) and DialPlan? I am using the GUI interface, and initially I am using Asterisk as a test only for communications on the internal network, latter I am installing an SPA 8800 |
| 02:06.52 | krahe | I just want to setup all extensions and users and have it working, with mailboxes, fax, answer machines, etc before put it on production mode |
| 02:08.44 | rustyclarkson | pabelanger-lap: I am wanting to add the CDR holdtime to the Master.csv, but it doesn't look like it's possible to add new fields to the Master.csv without patching Asterisk. If it's not possible, I'm fine with patching Asterisk instead of using an existing editable field. |
| 02:09.04 | rustyclarkson | Kind of a question, but I feel I already have the answer :p |
| 02:09.48 | pabelanger-lap | rustyclarkson: cdr_custom.conf |
| 02:09.53 | rustyclarkson | aha! |
| 02:09.55 | rustyclarkson | genious |
| 02:10.15 | krahe | Hi rusty, seems you are away ahead of me on Asterisk, do you have a spare time to give me a hand with my setup? |
| 02:10.41 | pabelanger-lap | krahe: Yes, you need to create dialplans |
| 02:10.51 | rustyclarkson | I can, but I haven't ever seen the GUI so I'm unsure how helpful |
| 02:10.54 | pabelanger-lap | ~book |
| 02:11.05 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 02:11.07 | pabelanger-lap | krahe: ^^^ |
| 02:11.24 | rustyclarkson | ^^ very useful book |
| 02:13.07 | *** part/#asterisk lanning (~lanning@208.87.235.224) |
| 02:13.11 | krahe | Thanks rusty, I really appreciate an answer even if you don't know, and pabelanger-lap, yes I am reading the book from asterisk, this free one. On the book they say to create the extension and the users. I did it. but the Asterisk-GUI complains about Trunks and Outgoing Rulles |
| 02:14.16 | pabelanger-lap | krahe: there is no book for the GUI, I'd suggest dropping it and creating everything by hand. A great way to learn |
| 02:14.53 | rustyclarkson | krahe: perhaps you can just enter fake trunk data? |
| 02:14.58 | pabelanger-lap | btw: which GUI |
| 02:16.08 | krahe | I just checked the GUI because I was doing by hand, I even get to record the extensions and users on mysql table. but what happens is - simple steup, 2 users on sip.conf [a] and [b] - if A dial B and B answer they can talk, but if B calls A and answer it is alway mute |
| 02:17.23 | rustyclarkson | clearly seems like a configuration issue |
| 02:17.35 | pabelanger-lap | krahe: both on local LAN? |
| 02:17.45 | krahe | sorry the GUI is: asteriskGUI from asterisk. I didn't want to install tribox, freepbx and others as they mess up with the rest of the sistem, I really don't like the idea of having apache running as other user |
| 02:18.56 | krahe | everything is on the local lan, I am not having it going outside yet, I also checked the firewall just in case. |
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| 02:21.25 | rustyclarkson | krahe: both endpoints are same type of phone? |
| 02:22.13 | krahe | both endpoints are X-Lite, on running on snow leopard other on bsd |
| 02:24.12 | krahe | as it is an initial system, I am using softphones, once everything is working I intend to put some cisco IP phones |
| 02:24.58 | rustyclarkson | krahe: so if B calls A and answers, B is muted? |
| 02:25.09 | krahe | yes |
| 02:26.04 | *** join/#asterisk ideaman (~ihaveapla@74-81-241-158.static.sdyl005.digis.net) |
| 02:26.10 | pabelanger-lap | rustyclarkson: canreinvite=yes? |
| 02:26.20 | pabelanger-lap | opps |
| 02:26.25 | rustyclarkson | :p |
| 02:26.26 | krahe | and must be something on the setup because if I change the extension 1000 to be user B not user A, and them 1001 to be A and not B, then B can call A and the voice is ok, but if A calls B than it is mute call |
| 02:26.44 | pabelanger-lap | krahe: canreinvite=yes for your end points? |
| 02:26.53 | rustyclarkson | yea, canreinvite could be helpful |
| 02:27.09 | krahe | not sure about this one, just a second |
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| 02:28.41 | krahe | this were the only fields I've inserted on the sip.conf under each sipuser: type=friend |
| 02:28.41 | krahe | ;context=phones |
| 02:28.41 | krahe | ;host=dynamic |
| 02:28.41 | krahe | ;secret=123456 ; to be changed by the users latter |
| 02:29.17 | krahe | without the starting ';' on the users, this is only the model that I use |
| 02:30.09 | ideaman | so I'm new and i've setup a simple box for testing, using flowroute as my provider, I have 1 Sip phone on my network and 1 peer in my sip.conf. Outbound works fine, but inbound 90% of the time will ignore my timeout, ring once on the Sip phone and never connect from the calling phone. No firewall. Any tips? |
| 02:30.57 | rustyclarkson | krahe: I'd suggest doing what pabelanger-lap said and give "canreinvite = yes" a try |
| 02:31.01 | krahe | I didn't get to any canreinvite part on the book, I was just following the book and they say on the chapter 4 on starting config, that on this point sip users should be able to talk, as I am having this issue I realize that I must have done something wrong already |
| 02:31.47 | rustyclarkson | krahe: it's incredibly odd with matching sip.conf contexts that this would occur, u've done a "dialplan reload" as well recently I hope :p |
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| 02:32.12 | krahe | I will do that and put on the sip.conf for each user, thanks |
| 02:32.55 | krahe | ideaman I am a begginer on that as well, sorry I can't help |
| 02:33.08 | ideaman | it's sure fun learning though |
| 02:33.10 | pabelanger-lap | krahe: unless you are explicitly setting it to no, then RTP should flow directly between your phones. |
| 02:33.11 | krahe | but do you know if you provider have a blocking as well |
| 02:33.34 | rustyclarkson | ideaman: what timeout are you talking about? |
| 02:33.35 | pabelanger-lap | krahe: check your firewalls on your systems |
| 02:34.02 | ideaman | I had ufw enabled, but since deactivated. It's acting like it's a firewall type of issue though |
| 02:34.33 | krahe | ok thanks pabelanger-lap, I will come back latter if I can't get it working, and recheck all steps on the book. Thanks for all |
| 02:35.15 | ideaman | here's a link to more of my details: http://forums.digium.com/viewtopic.php?f=1&t=74345&sid=0d6b661a7079d975a6a80b8255f694eb |
| 02:36.15 | pabelanger-lap | ideaman: are you behind a NAT? |
| 02:38.45 | *** part/#asterisk krahe (~krahe@203-109-246-60.static.bliink.ihug.co.nz) |
| 02:39.01 | ideaman | no, i have a public ip address |
| 02:39.37 | ideaman | sorry if my answers sound odd, I'm still learning all this |
| 02:40.59 | pabelanger-lap | ideaman: we'd have to see a SIP debug trace then. |
| 02:41.05 | pabelanger-lap | ~collectdebug |
| 02:41.06 | infobot | well, collectdebug is a method of collecting logs allowing others help troubleshoot an issue. Refer to doc/HOWTO_collect_debug_information.txt or http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt |
| 02:41.17 | pabelanger-lap | ideaman: however, I'm about to log for the night |
| 02:41.29 | ideaman | k. thanks though |
| 02:42.04 | pabelanger-lap | ideaman: what version of Asterisk BTW? |
| 02:42.44 | ideaman | 1.6 |
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| 02:43.35 | ideaman | 1.6.2.5 |
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| 02:48.43 | rustyclarkson | ideaman: I agree with pabelanger-lap, a SIP debug is pretty much the only thing to figure it out. If you know how SIP works, it wouldn't be a bad idea to take a look at the problematic SIP conversation. |
| 02:48.56 | rustyclarkson | and now I want to go home |
| 02:49.01 | rustyclarkson | good luck ideaman |
| 02:49.08 | rustyclarkson | thanks again for your help pabelanger-lap |
| 02:49.53 | ideaman | i guess i just have to get good at reading the sip debug |
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| 02:52.06 | nice2teach | Looking for some help getting started. Is anyone available |
| 03:07.42 | ideaman | im new myself. but i can try and help, and I'm sure someone else can chime in |
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| 03:25.52 | ChannelZ | I guess that wasn't good enough |
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| 04:14.36 | spenguin[work] | TEST |
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| 04:37.05 | ChannelZ | there's that weird sound again |
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| 04:46.41 | whistlr | hi, Im fairly new to asterisk, and have a system with 2 vitelity trunks w/ two different number setup |
| 04:46.48 | whistlr | different host names etc |
| 04:47.02 | whistlr | but the voicemail keeps being routed to the first number |
| 04:47.07 | whistlr | and I cant figure out why |
| 04:51.03 | spenguin[work] | hey ChannelZ |
| 04:51.42 | ChannelZ | Ahoy |
| 04:52.07 | ChannelZ | whistlr: vitelity's voicemail or local vm run by you on asterisk? |
| 04:57.46 | whistlr | local vm |
| 04:57.54 | *** join/#asterisk knot (yiffstar66@unaffiliated/devemo) |
| 04:58.07 | whistlr | I have 2 numbers from vitelity |
| 04:58.24 | whistlr | and two TIDs on the system, but I cant get callers to go to the rigth place |
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| 05:02.14 | ChannelZ | how do you mean |
| 05:02.57 | ChannelZ | they should come into the extension same as your DID - exten => 2225551212 etc |
| 05:04.01 | whistlr | so when someoen dials 5552125555 they should go to that "group" of extensions etc |
| 05:04.15 | whistlr | and 5552124444 that group of extensions |
| 05:04.20 | whistlr | but they all go to teh first number |
| 05:04.33 | whistlr | disregard i guess for right now |
| 05:04.40 | whistlr | i gotta figure out how the system is setup first |
| 05:04.53 | ChannelZ | well I have no idea what you're really saying without seeing some console output and dialplan |
| 05:04.59 | whistlr | i didnt set it up and am doing some troubleshooting so I have to figure out whats going on |
| 05:05.20 | whistlr | i dont know if they have 2 seperate instances of asterisk running or what |
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| 05:10.53 | *** join/#asterisk Micc (~quassel@c-98-225-57-96.hsd1.wa.comcast.net) |
| 05:11.26 | Micc | So does a polycom ip450 support siren14? I'm guessing not since I tried to use it and it only does g722. |
| 05:12.20 | Micc | I found on the web some lists of other polycom phones that support siren14, so i'm guessing the ip450 doesn't support it since its not in that list. |
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| 05:59.24 | coppice | Micc: Polycom are really bad for specifying which codecs each phone supports. but I'm pretty sure G.722,1C is not supported by the IP450 |
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| 06:41.58 | *** join/#asterisk Chris-NB (~chris@mail.ecos.at) |
| 06:42.02 | Chris-NB | hello |
| 06:43.01 | Chris-NB | is it possible to configure a queue that every call which is placed in it is routet to agent 1 (10 Secondes) and then to agent 2. |
| 06:43.16 | Chris-NB | and not place call 1 to agent 1, call 2 to agent 2, call 3 to agent 1 .... |
| 06:43.55 | Chris-NB | every call should be routet to agent 1, if no answer, route to agent 2 and then leave the queue |
| 06:45.33 | kaldemar | you can do that in your dialplan with queue timeouts. |
| 06:49.39 | Chris-NB | I've asterisk 1.4.26.2 |
| 06:49.53 | Chris-NB | the call is only placed for 20 seconds in the queue |
| 06:50.05 | Chris-NB | and every agent gets called 10 seconds |
| 06:50.28 | Chris-NB | so, the first caller is routet 10 seconds to agent 1, 10 seconds to agent 2 and then drops out |
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| 06:51.01 | Chris-NB | but, if the first call gets answered by agent 1, the second call is routet to agent 2 (even if agent 1 is free) |
| 06:51.22 | Chris-NB | I want every call be routet to agent 1 first (if he is free) |
| 06:51.48 | kaldemar | your last two requirements conflict. |
| 06:52.55 | kaldemar | anyway, there is no queue strategy that will function exactly like that. you need to do that in your dialplan. |
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| 06:55.45 | Chris-NB | my requirements do not conflict |
| 06:56.22 | kaldemar | "if the first call gets answered by agent 1, the second call is routet to agent 2 (even if agent 1 is free)" clashes with "I want every call be routet to agent 1 first (if he is free)" |
| 06:57.51 | Chris-NB | both agents are free, first call gets routet to agent 1, call gets answered, call gets disconnected. both agents are free, call gehts routet to agent 1 and not to agent 2 |
| 06:57.57 | Chris-NB | thats what I want |
| 06:58.04 | Chris-NB | don't think that clashes |
| 06:58.09 | Chris-NB | or? |
| 07:01.20 | ChannelZ | If you want to keep using queues, I think the strategy of 'linear' should be doing what you want |
| 07:01.47 | kaldemar | if "if the first call gets answered by agent 1, the second call is routet to agent 2 (even if agent 1 is free)" is NOT what he wants after all. |
| 07:02.01 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 07:02.03 | ChannelZ | what I don't remember is if 1.4 had linear or if it was called something else, etc |
| 07:03.08 | kaldemar | it does not have a linear strategy. |
| 07:03.20 | ChannelZ | suck. Upgrade I guess. :) |
| 07:05.59 | kaldemar | or DIY in the dialplan. :) own queues for the agents and timeouts for queue apps in the extension. |
| 07:08.24 | ChannelZ | yeah probably makes more sense since none of the other features of queues are seemingly desired |
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| 08:01.20 | festr_ | karma gones from issue.digium? :) |
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| 08:23.04 | tzafrir_laptop | festr_, I'm not really sure how useful it really was |
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| 08:44.46 | Lantizia | Hey is it normal to have a line that looks like... "<3248>pickupgroup=1 " ... i.e. those brackets in-front for an extension? or have I found a typo? |
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| 08:51.16 | yskas | Good Day |
| 08:51.17 | yskas | . |
| 08:51.37 | yskas | Can any one point me in the right direction to setup a PTT with asterisk |
| 08:51.39 | kaldemar | Lantizia: where did you find that? |
| 08:51.52 | Lantizia | kaldemar, in a phone system |
| 08:52.10 | Lantizia | kaldemar, "sip.conf" basically all the extensions/peers are defined in there |
| 08:52.37 | kaldemar | Lantizia: "phone system" means nothing in asterisk. where in sip.conf? |
| 08:52.53 | Lantizia | kaldemar, inside the section for 3248 |
| 08:53.52 | kaldemar | if it is an uncommented parameter under [3248], there should be no <3248> in front of "pickupgroup". |
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| 08:55.18 | Lantizia | kaldemar, thought as much - someone must have screwed up |
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| 09:14.31 | markitoxs | hello, i was wondering if you guys could help me, whenever i connect to the asterisk console i get LOTS of messages like this one: << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/2908-Genius-b4331148] , how can i turn that off? |
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| 09:19.26 | markitoxs | anyone? |
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| 11:10.15 | soliax | hi all... is this the right channel to ask a question about problems i'm having with the uni-ast package? |
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| 11:22.50 | joobie | sup boys |
| 11:24.58 | fauxalliance | \o |
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| 11:44.08 | Yudaisrael1984 | hi everyone im installing a new, fresh , installation and i get a error "you do not appeaer to have the sources for the 2.6.18-164.15.1.el5PAE kerme; omsta;;ed |
| 11:44.26 | Yudaisrael1984 | kernel installed |
| 11:44.31 | *** join/#asterisk viq (~viq@unaffiliated/viq) |
| 11:45.06 | fenrus | try installing the headers |
| 11:45.14 | fenrus | what distribution is it? |
| 11:45.31 | Yudaisrael1984 | centos 5.4 |
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| 11:46.18 | fenrus | check the packet-system for something like kernel-headers or kernel-sources |
| 11:46.25 | fenrus | for your kernel |
| 11:47.32 | Yudaisrael1984 | meaning to do a yum list installed and grep kernel? |
| 11:49.10 | fenrus | yea |
| 11:51.35 | *** part/#asterisk neurosys (~neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
| 11:52.50 | joobie | guys is there a way with moh to allow the caller to press a number and then make moh jump to the next mp3? |
| 11:53.10 | joobie | using multiple mp3's atm for my moh.. context points to the dir |
| 11:53.28 | kaldemar | Yudaisrael1984: "yum install kernel-PAE-devel" |
| 11:53.48 | Yudaisrael1984 | ok trying that |
| 11:54.10 | kaldemar | you probably need other dependencies installed too. |
| 11:54.42 | Yudaisrael1984 | doesnt show that it needs any other dependencies |
| 11:56.11 | Yudaisrael1984 | the following are installed : kernel kernel-PAE kernel-PAE-devel kernel-devel kernel-headers |
| 11:56.38 | *** join/#asterisk giany (~giany@shifu.x83.org) |
| 11:56.41 | giany | hello |
| 11:56.49 | *** join/#asterisk bodie (~bodie@fcnoos-nd-fw01.freecode.no) |
| 11:57.09 | bodie | hi all, have someone recommendations for SIP softphone for appkonference plugin? |
| 11:57.23 | giany | anyone can tell me why I have calls in : sip show channels that do not get deleted? e.g : myip ffd24fb757 00b153774767905 0x0 (nothing) No (d) Rx: ACK |
| 11:57.24 | bodie | I'm trying Ekiga, but when I call to conference I can't use webcamera |
| 11:57.38 | bodie | it doesn't show as option to enable it for this call |
| 12:03.46 | Yudaisrael1984 | any ideas? |
| 12:05.45 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 12:06.53 | joobie | re TK |
| 12:11.04 | *** join/#asterisk xEBIx (~ebi@188-194-122-0-dynip.superkabel.de) |
| 12:11.25 | xEBIx | hello |
| 12:13.09 | xEBIx | if in users.conf a context is set than it wil search for it it [] in extensions.conf in search in it for the extension, right? |
| 12:13.36 | [TK]D-Fender | xEBIx: And it matches that user, yes |
| 12:13.43 | [TK]D-Fender | !users.conf |
| 12:13.48 | [TK]D-Fender | ~users.conf |
| 12:13.59 | infobot | [~users.conf] users.conf is a flaming pile of sh1t that takes the fine control of several perfectly usable * config files and reduces them to the lowest common denominator and makes your system behave like a "toaster grade" PBX system. |
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| 12:14.20 | joobie | TK |
| 12:14.23 | joobie | u might know this |
| 12:14.33 | joobie | is there a way with moh to allow the caller to press a number and then make moh jump to the next mp3? |
| 12:14.58 | joobie | i mean, without going and creating my own thing outside of moh |
| 12:16.07 | xEBIx | [TK]D-Fender, so using users.conf is not that good idea? |
| 12:16.37 | [TK]D-Fender | joobie: No |
| 12:16.43 | joobie | fuk |
| 12:16.55 | joobie | that sucks |
| 12:17.00 | [TK]D-Fender | xEBIx: Less than the individual pieces are worth and creates useless dialplan, etc |
| 12:17.52 | xEBIx | [TK]D-Fender, so i should better set up my clients in sip.conf? |
| 12:18.00 | joobie | Yudaisrael1984, whats ur prob again? i lost my backlog |
| 12:18.18 | [TK]D-Fender | xEBIx: Yes |
| 12:19.37 | xEBIx | the most of entris in users.conf i can copy to sip.conf, don't I? |
| 12:20.14 | [TK]D-Fender | xEBIx: Were you using AsteriskGUI before? |
| 12:20.26 | Yudaisrael1984 | i get a error "you do not appeaer to have the sources for the 2.6.18-164.15.1.el5PAE kernel installed |
| 12:20.32 | xEBIx | I am not using AsteriskGUI at all |
| 12:20.35 | Yudaisrael1984 | i have the dependencies installed |
| 12:20.40 | Yudaisrael1984 | yet i still get that message |
| 12:21.27 | xEBIx | i even don't know if its installed, i guess not |
| 12:21.53 | joobie | type.. |
| 12:21.54 | joobie | hmm |
| 12:22.02 | joobie | rpm -qa | grep kernel-headers |
| 12:22.05 | kaldemar | Yudaisrael1984: does "yum info kernel-PAE-devel" list same kernel version as "uname -r"? |
| 12:22.06 | joobie | wat do u get |
| 12:22.25 | joobie | ahh |
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| 12:22.42 | Yudaisrael1984 | uname -r is 2.6.18-164.15.1.e15PAE |
| 12:22.49 | joobie | if do umm.. |
| 12:22.55 | joobie | rpm -qa | grep kernel- |
| 12:23.15 | joobie | based on kaldemar's comment it may be PAE |
| 12:23.32 | Yudaisrael1984 | kernel-headers-2.6.18-194.3.1.e15 |
| 12:23.43 | joobie | i didnt know there was a PAE kernel |
| 12:23.44 | kaldemar | Yudaisrael1984: there's a mismatch. |
| 12:23.45 | joobie | for linux |
| 12:23.48 | joobie | i thought that was a linux thing |
| 12:23.59 | joobie | i mean windows thing |
| 12:24.16 | Yudaisrael1984 | i agree how do i fix that |
| 12:24.31 | [TK]D-Fender | xEBIx: Then yes I'd highly recomend undoing your use of users.conf then |
| 12:24.46 | kaldemar | Yudaisrael1984: upgrade your kernel to the latest in package management? |
| 12:24.57 | Yudaisrael1984 | im using only cli |
| 12:24.57 | joobie | yum install kernel-headers-2.6.18-164.15.1.e15PAE |
| 12:25.06 | xEBIx | ok doing it |
| 12:25.06 | joobie | do that |
| 12:25.14 | joobie | u will install the headers for your current kernel |
| 12:25.35 | joobie | if u wana upgrade ur kernel tho do wat kaldemar said |
| 12:25.38 | Yudaisrael1984 | non are available i get from yum |
| 12:25.52 | joobie | can u upgrade ur kernel? |
| 12:25.59 | joobie | like any reason not to? |
| 12:26.13 | Yudaisrael1984 | no i can do that no problem |
| 12:26.18 | joobie | then do that |
| 12:26.26 | joobie | then install the same version of hte kernel-hedaers pkg |
| 12:26.31 | joobie | and bobs ur bitch |
| 12:26.32 | joobie | or uncle |
| 12:27.01 | Yudaisrael1984 | so i am upgrading kernel with yum? |
| 12:27.11 | Yudaisrael1984 | yum upgrade kernel* |
| 12:27.19 | Yudaisrael1984 | ? |
| 12:27.48 | joobie | yum install kernel |
| 12:27.57 | joobie | u should always install ur kernel rather than upgrade |
| 12:28.00 | joobie | so u can fallback |
| 12:28.04 | joobie | if needed |
| 12:28.13 | joobie | something i learnt in rhce |
| 12:28.15 | joobie | makes sense |
| 12:28.22 | joobie | faggot teacher.. but made sense.. |
| 12:28.42 | Yudaisrael1984 | so i did that got the response package already installed nothing to do? |
| 12:28.54 | joobie | specify the specific kernel |
| 12:28.56 | Yudaisrael1984 | is there anything to do maybe in the yum folder? |
| 12:29.02 | Yudaisrael1984 | oh |
| 12:29.11 | Yudaisrael1984 | whats the kernel that i should be upgrading to? |
| 12:29.21 | joobie | actually i think im trippen .. that install cmd might be for rpm |
| 12:29.29 | joobie | you may need to do upgrade inyum |
| 12:29.38 | joobie | try the specific ver.. if it doesnt work., upgrade with yum |
| 12:29.44 | joobie | sry, had a few scotches |
| 12:29.55 | joobie | but the advise is still good |
| 12:29.59 | joobie | i think |
| 12:30.05 | Yudaisrael1984 | upgrading yum is the same? yum upgrade? |
| 12:30.20 | joobie | rpm -i on kernels |
| 12:30.26 | joobie | will keep the current kernel |
| 12:30.28 | joobie | and install another |
| 12:30.36 | joobie | so you can roll back if your system shits its dacks |
| 12:30.45 | joobie | yum, i think you may need to do upgrade |
| 12:30.49 | joobie | but try install first |
| 12:31.01 | joobie | yum install kernel-ver |
| 12:31.13 | joobie | if fails, yum upgrade kernel* |
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| 12:35.30 | Yudaisrael1984 | doing all that now i will update soon |
| 12:39.41 | xEBIx | how fo i rebuild the fullname directive in users.conf? |
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| 12:45.32 | [TK]D-Fender | xEBIx: What did it do? |
| 12:46.35 | xEBIx | the argument shows up in the display of a called SIP phone |
| 12:46.46 | [TK]D-Fender | xEBIx: callerid <---- |
| 12:47.18 | sputnick | hi there |
| 12:47.18 | xEBIx | callerid is more a redirect, isnt it |
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| 12:48.45 | sputnick | anyone can explain me how make a call to a phone from the command line from my debian asterisk box ( only SIP ) ? ( just ringing, that's enough for now... ) |
| 12:49.10 | xEBIx | ahh i see |
| 12:49.33 | [TK]D-Fender | sputnick: help console dial |
| 12:49.35 | *** join/#asterisk jrz (~jrz@dhcp-077-250-159-146.chello.nl) |
| 12:50.14 | sputnick | [TK]D-Fender: where ? |
| 12:50.23 | [TK]D-Fender | sputnick: * CLI of course |
| 12:50.49 | sputnick | [TK]D-Fender: I don't get you |
| 12:51.05 | [TK]D-Fender | sputnick: Go to * CLI and type that in and read the instructions |
| 12:51.15 | sputnick | ho ! sorry |
| 12:52.19 | bodie | hmmm... it's really funny. What people use for appkonference and video? |
| 12:52.48 | bodie | because Ekiga doesn't offer webcam during call, linphone crashes on stack smashing detected, Xlite is just for Windows, Twinkle doesn't have video support |
| 12:53.36 | [TK]D-Fender | " Xlite is just for Windows".. No |
| 12:53.38 | [TK]D-Fender | ~xlite |
| 12:53.39 | infobot | [~xlite] X-Lite is a free SIP soft-phone for Windows, MacOSX, and Linux that can be downloaded from http://www.counterpath.com/ |
| 12:54.42 | sputnick | [TK]D-Fender: do you have a real example to call someone from "console dial" ? |
| 12:55.02 | [TK]D-Fender | sputnick: The instructions show you waht to do. Do it. |
| 12:55.27 | sputnick | done [TK]D-Fender, only : "console dial [extension[@context]] " |
| 12:56.20 | [TK]D-Fender | sputnick: So go do it |
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| 13:01.41 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 13:02.32 | eduzimrs | \/ |
| 13:05.15 | *** join/#asterisk m_tadeu (~quassel@89.180.164.45) |
| 13:06.07 | eduzimrs | u know something about DTMFMODE Problem between asterisk servers ? |
| 13:08.02 | tzafrir_laptop | sputnick, there's also originate, if you want to generate a call from a different phone |
| 13:08.25 | m_tadeu | hi everyone...I want to AddQueueMember as soon as the agent registers the softphone in the server...is that possible? |
| 13:08.38 | xEBIx | the context directive does not make sense in client definitions in sip.conf, is it? |
| 13:09.37 | [TK]D-Fender | eduzimrs: Failure to agree on mode will kill dtmf. |
| 13:10.03 | [TK]D-Fender | xEBIx: Of course it does. Every device points to a context. That's where their calls go <- |
| 13:10.41 | [TK]D-Fender | m_tadeu: You'd have to monitor with AMI or similar and lauch that yourself. All external scripting |
| 13:10.49 | xEBIx | ok and the context directive points were incoming calls go? |
| 13:11.03 | xEBIx | in general i mean |
| 13:12.04 | m_tadeu | [TK]D-Fender: so asterisk sends me an event when some phone registers? |
| 13:13.05 | [TK]D-Fender | xEBIx: calls from that device go where you tell them to. |
| 13:13.29 | [TK]D-Fender | m_tadeu: there may be. Or you can use REGEXTEN as a trigger for something more detectable |
| 13:17.11 | eduzimrs | [TK]D-Fender where i set dtmfmode in E1 channel ? |
| 13:17.28 | [TK]D-Fender | eduzimrs: You don't that is always inband. |
| 13:18.59 | xEBIx | hmm problem i don't get, Ive context=default in [general] in sip.conf, in [default] in extensions.conf is set the correct extension but incoming calls aren't routed, can't find extension |
| 13:21.13 | [TK]D-Fender | xEBIx: perhaps you should really REALLY look at that call then. |
| 13:21.34 | [TK]D-Fender | xEBIx: and you should NEVER use [default] as a context |
| 13:22.24 | xEBIx | i even can copy and search it, and it will find |
| 13:23.08 | xEBIx | [TK]D-Fender, what do you mean with dont use default as context? |
| 13:23.09 | *** join/#asterisk otavio (~otavio@debian/developer/otavio) |
| 13:24.31 | [TK]D-Fender | xEBIx: You seem to have trouble with the most direct statements. You should never create or use a context named [default] EVER. Change the name of the context your phone will be using and in [general] you should NOT point to a context that lets a call co out of your system, etc |
| 13:25.29 | russellb | [TK]D-Fender: you seem to have trouble being nice :-p |
| 13:26.47 | xEBIx | hmmm context=default is default in the config here... |
| 13:27.00 | [TK]D-Fender | russellb: Oh we're very far from the magma I'm capable of... you know it :) This is only the slightest of edge... |
| 13:27.14 | [TK]D-Fender | xEBIx: Don't do that then. make it ANYTHING else. |
| 13:27.18 | otavio | Hello; I'm having a hangup when transfering the call but only when the call is using a SIP trunk; when using dahdi it works |
| 13:27.36 | russellb | [TK]D-Fender: ANYTHING? what about empty |
| 13:27.52 | [TK]D-Fender | russellb: No, that would be NOTHING :p |
| 13:27.56 | xEBIx | [TK]D-Fender, why? |
| 13:28.02 | [TK]D-Fender | xEBIx: Security reasons |
| 13:28.03 | russellb | [TK]D-Fender: whitespace? |
| 13:28.29 | xEBIx | [TK]D-Fender, can you explain that a bit more? |
| 13:28.36 | [TK]D-Fender | xEBIx: [default] is a built-in fall-back for many things and you do not want some accident allowing a caller to do something you did not specifically choose to allow them to. |
| 13:28.42 | eduzimrs | [TK]D-Fender hum, letme explain u my problem, i have an asterisk server where DTMFMODE=rfc2833 when a dial to another asterisk that uses E1 trunk the digits typed in the IVR are not recognized |
| 13:28.45 | [TK]D-Fender | russellb: MORE "nothing" :p |
| 13:28.53 | [TK]D-Fender | russellb: *poke* |
| 13:28.57 | russellb | [TK]D-Fender: a binary blob that is invalid text? |
| 13:29.09 | otavio | Do someone have any idea how I can discover it? |
| 13:29.52 | anonymouz666 | russellb: sorry to ask you directly, but about this mail you wrote long ago: http://www.mail-archive.com/asterisk-dev@lists.digium.com/msg30902.html - is this already available in 1.6.2.x? |
| 13:29.58 | russellb | looks |
| 13:30.01 | xEBIx | [TK]D-Fender, right now i've got nothing than demo in default so what could happen? |
| 13:30.19 | [TK]D-Fender | eduzimrs: See you mention using E1, and then seem to imply SIP. The bits are slow to add up. So you are going server to server via SIP and out #2's E1? |
| 13:30.41 | *** join/#asterisk retentiveboy (~pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net) |
| 13:31.19 | [TK]D-Fender | xEBIx: What could happen is that you'll continue to build up your system forgetting these common good practices and then some day accidentally have a dialplan failure allow callers to use your outbound resources at YOUR cost. |
| 13:31.26 | russellb | anonymouz666: nope, it still behaves the same way. |
| 13:31.59 | xEBIx | hmm i see your point |
| 13:32.01 | anonymouz666 | alright thanks |
| 13:32.12 | anonymouz666 | no reason yet to move to 1.6.2.X |
| 13:32.14 | *** part/#asterisk bodie (~bodie@fcnoos-nd-fw01.freecode.no) |
| 13:32.23 | russellb | anonymouz666: 1.8 will be in beta soon anyway :-) |
| 13:32.30 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 13:32.35 | m_tadeu | [TK]D-Fender: do you know where can I find a list of events send to the AMI? |
| 13:32.38 | [TK]D-Fender | anonymouz666: Here's one : Every lower branch is about to be in security-fix-only status |
| 13:33.03 | [TK]D-Fender | m_tadeu: * CLI has all sorts of help and lists... take a look. |
| 13:33.23 | anonymouz666 | [TK]D-Fender: yeah sure. I read the asterisk versions table... |
| 13:33.55 | eduzimrs | [TK]D-Fender the call outgoing in a sip channel and is received by a dahdi-channel in E1 trunk |
| 13:34.23 | [TK]D-Fender | anonymouz666: And depending how far back your configs are standard for you may have a larger than necessary upgrade bump by putting it off too long. |
| 13:34.47 | [TK]D-Fender | ediCare to tell me how you go out SIP and in E1? |
| 13:34.52 | [TK]D-Fender | eduzimrs: Care to tell me how you go out SIP and in E1? |
| 13:35.20 | [TK]D-Fender | eduzimrs: You are leaving off important bits again |
| 13:42.42 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 13:43.51 | *** join/#asterisk Pazzo (~ugelt@reserved-225136.rol.raiffeisen.net) |
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| 13:51.50 | mallchin | Hi guys -- I am getting no audio when dialling via SIP -- could it be to do with trunking? |
| 13:51.55 | mallchin | What is trunking, and do I need it? |
| 13:53.46 | mallchin | as dahdi replaces zaptel, is there a replacement to ztdummy? |
| 13:55.22 | kaldemar | there is no trunking in sip. if you have a NAT involved in your setup, see this: |
| 13:55.26 | kaldemar | ~sipnat |
| 13:55.27 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 13:55.41 | kaldemar | if there's not NAT, check your codec settings. |
| 13:56.07 | bn-7bc | mallchin: dahdidummy |
| 13:56.15 | *** join/#asterisk MiserySoft (~LND@92.40.119.58.sub.mbb.three.co.uk) |
| 13:56.16 | kaldemar | the dummy module functionality is in the core dahdi module in newer versions, there is no separate dahdi_dummy anymore as far as i know. |
| 13:56.24 | anonymouz666 | sipnat is trademark of [TK]D-Fender |
| 13:56.37 | *** join/#asterisk Lord_Rahl (~quassel@173-162-32-1-michigan.hfc.comcastbusiness.net) |
| 13:57.12 | eduzimrs | [TK]D-Fender ok, im dialing from a IAX peers configured in a asterisk server that allow calls through a internet connection and im trying to call another asterisk that is configured to make/receive calls by an E1 trunk so it doesnt accept my digits typed in IVR |
| 13:57.37 | russellb | i don't understand why people in this channel try to say that "SIP trunk" doesn't exist. It's extremely common terminology in the VoIP industry ... |
| 13:57.45 | [TK]D-Fender | eduzimrs: IAX doesn't even HAVE a "dtmfmode". It is always Out Of Band. |
| 13:58.25 | xEBIx | [TK]D-Fender, I don't get it working. please have a look at my config sip.conf http://nopaste.info/e5f33840c2.html extensions.conf |
| 13:59.26 | *** part/#asterisk giany (~giany@shifu.x83.org) |
| 13:59.32 | xEBIx | [TK]D-Fender, sip.conf is already outdatet, I commented context in [general] out and put the sipgate accounts to the correct clients |
| 13:59.52 | kaldemar | russellb: if you refer to what i said, i didn't say that. i meant that there is no trunking (as in iax2 trunking) in sip. |
| 14:00.16 | russellb | ah, fair enough |
| 14:00.18 | Lord_Rahl | Need help setting the moitor_filename for queue calls recording. here is how I like my format to go MONITOR_FILENAME=${TIMESTAMP}-${CALLERIDNAME}-${AGENTEXTEN} I am not sure where to place it? |
| 14:00.53 | russellb | kaldemar: i've seen some others in here go on rants about the term "SIP trunk" when used in the context of connectivity to a termination provider. |
| 14:00.59 | russellb | and I think it's silly is all :-) |
| 14:01.53 | *** join/#asterisk jhirley (~jhirley@c-98-211-237-248.hsd1.fl.comcast.net) |
| 14:02.04 | [TK]D-Fender | Lord_Rahl: Before calling Monitor |
| 14:02.27 | Lord_Rahl | [TK]D-Fender: in the queue.conf |
| 14:03.02 | [TK]D-Fender | xEBIx: Go look at the CALL. Enable SIP DEBUG at * CLI and look at what the request actually looks like |
| 14:04.01 | Lord_Rahl | [TK]D-Fender: like this ? http://pastebin.com/8Aebdr1t |
| 14:06.03 | *** join/#asterisk jpeeler (~jpeeler@asterisk/digium-software-dev/jpeeler) |
| 14:06.50 | [TK]D-Fender | Lord_Rahl: What does the sample config show? |
| 14:07.06 | [TK]D-Fender | Lord_Rahl: I'm rather certain the () are inappropriate |
| 14:07.44 | Lord_Rahl | [TK]D-Fender: the queue context |
| 14:08.01 | [TK]D-Fender | Lord_Rahl: Sorry, you answer doesn't make any sense. |
| 14:09.35 | Lord_Rahl | [TK]D-Fender: sorry it was the queue i set up. Here is the whole queue.conf http://pastebin.com/rXn6uAvx |
| 14:09.47 | eduzimrs | [TK]D-Fender my aoutgoing call is from a SIP Channel and is received by a E1 trunk (dahdi) |
| 14:10.35 | *** join/#asterisk TheDavidFactor (~chatzilla@nc-71-52-236-209.dhcp.embarqhsd.net) |
| 14:11.36 | bent_screwdriver | Anyone upgraded to dahdi 2.3.0.1 that has the VPM (hw echo can) module for the digium TE122? I had to downgrade from 2.3.0 to 2.2.1.1 becuase of a bug where the VPM module was constantly resetting and they say it is fixed in 2.3.0.1. |
| 14:14.05 | *** join/#asterisk nighty^ (~nighty@x122091.ppp.asahi-net.or.jp) |
| 14:14.58 | xEBIx | hmm this is strange sip debug says Looking for 6344662e0 in sipin (domain 188.194.122.0) at a point, ist reload resetting thin sipin is nowere set anmore |
| 14:15.48 | m_tadeu | [TK]D-Fender: cool...I'm taking the proper eventsin AMI...I think I can manage with this. thanx |
| 14:16.14 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:16.15 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:16.19 | mallchin | kaldemar: excellent, thank you, I'll have a read |
| 14:17.40 | *** join/#asterisk hugorebelo (~hugo@200-171-132-124.completo.com.br) |
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| 14:23.43 | xEBIx | hmm i ssee reload in cli is not a good idea in every case |
| 14:23.49 | xEBIx | it works |
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| 14:28.51 | *** join/#asterisk QbY (~kelvin@adsl-065-012-166-106.sip.asm.bellsouth.net) |
| 14:29.25 | QbY | Whatwould make a 1.6.2.6 box not respond to anything. I look at network traffic and see it being flooded with registration requests, while sip debugs show nothing |
| 14:29.32 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
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| 14:30.05 | QbY | no call processing whatsoever |
| 14:32.12 | russellb | a deadlock |
| 14:32.17 | QbY | after a reboot? |
| 14:32.19 | russellb | try upgrading asterisk. |
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| 14:50.01 | *** join/#asterisk wcselby (~wcselby@208.180.112.123) |
| 14:51.08 | wcselby | o/ |
| 14:51.56 | wcselby | http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.33-rc1 seems to be broke |
| 14:52.22 | wcselby | ahh, nevermind |
| 14:52.26 | wcselby | i think i see why |
| 15:03.32 | *** join/#asterisk btsteve (~tstevens@24-196-234-39.dhcp.gwnt.ga.charter.com) |
| 15:04.01 | *** join/#asterisk rocksfrow (~kyle@74-92-146-97-WashingtonDC.hfc.comcastbusiness.net) |
| 15:05.14 | *** join/#asterisk siya (~djerk@87-194-171-55.bethere.co.uk) |
| 15:06.08 | siya | simple question: why is one trunk listed as monitored and the other one (different ITSP but same config) unmonitored |
| 15:06.42 | rocksfrow | can you dial into another phones voicemail if you know the password? |
| 15:06.49 | rocksfrow | i'm trying to share a voicemail box with a few users |
| 15:06.56 | rocksfrow | what is the best way to do this? |
| 15:07.02 | rocksfrow | a few phones** |
| 15:07.29 | wcselby | rocksfrow - setup an exten that goes to VoicemailMain(@context), and then follow the prompts |
| 15:07.45 | wcselby | siya - you've got a qualify=yes statement on one, and not on the other |
| 15:07.56 | rocksfrow | wcselby, just curious, how could i dial into the main voicemail from a phone? |
| 15:08.09 | wcselby | rocksfrow - setup an exten that goes to VoicemailMain(@context), and then follow the prompts |
| 15:08.23 | wcselby | i.e exten => 3500,1,VoicemailMain(@default) |
| 15:08.30 | siya | wcselby, changed my google search and found that so tx for the confirmation. (I tend to trust irc more than google) |
| 15:09.00 | wcselby | then, on a phone that has access to that contex tthat you put that exten, dial 3500 |
| 15:09.00 | rocksfrow | wcselby, *98 is what i was looking for. |
| 15:09.00 | wcselby | and follow the prompt |
| 15:09.05 | siya | wcselby, and with regards to extensions? created three equal extensions and one has status unmonitored |
| 15:09.10 | wcselby | rocksfrow - well, you didn't say you were using freepbx |
| 15:09.35 | rocksfrow | wcselby, =P |
| 15:09.55 | siya | is there a way to stop the following messages in console: "Manager 'admin' logged on/off" |
| 15:09.56 | p3nguin | That was a strategic calculation on his part. |
| 15:10.00 | wcselby | siya - the same applies, if an extension is unmonitored it's either missing a qualify statement or it's "qualify=no" |
| 15:10.11 | siya | wcselby, tx |
| 15:10.30 | wcselby | siya - don't have the freepbx status page up when you access your asterisk console |
| 15:10.41 | wcselby | ~freepbx |
| 15:10.41 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 15:11.02 | wcselby | rocksfrow - *98 in freepbx just does what I said up a few lines |
| 15:11.04 | siya | wcselby, apart from the obvious... ;) |
| 15:11.26 | wcselby | :) |
| 15:11.30 | siya | facepalms |
| 15:12.06 | siya | I changed the wrong setting (qualify as opposed to nat, I really should read better... 8sigh* |
| 15:12.09 | wcselby | ugh, i open a ticket on issues.asterisk.org and it's closed before I can try the suggested fix |
| 15:12.14 | siya | s/8/*/ |
| 15:12.35 | wcselby | siya - :) |
| 15:16.18 | *** join/#asterisk hugorebelo (~hugo@200-171-132-124.completo.com.br) |
| 15:16.21 | wcselby | i'm confused, why was 1.4.32 released on 06-01-10, but changes made to the code on 05-06-10 are in the 1.4.33-rc1 release.....? |
| 15:17.09 | pabelanger | wcselby: Thats the way releases work. |
| 15:17.27 | siya | wcselby, "displayconnects = no" >> manager.conf |
| 15:18.05 | *** join/#asterisk c0rnoTa (~c0rnoTa@109.188.15.127) |
| 15:18.05 | *** part/#asterisk c0rnoTa (~c0rnoTa@109.188.15.127) |
| 15:18.35 | pabelanger | wcselby: once we create a 1.4.32-rc1, we only fixes bugs related to RC-1. So when we release 1.4.33-rc2, we take a new snapshot from 1.4 branch, not the previous RC |
| 15:18.44 | *** join/#asterisk sulex (~sulex@dynamic-adsl-78-14-171-203.clienti.tiscali.it) |
| 15:19.46 | wcselby | pabelanger - i guess I'm just too tired to understand it this morning. man i'm dragging |
| 15:20.10 | wcselby | days like today I really wish caffiene didn't give me migraines |
| 15:20.42 | russellb | it's not that bad .. 1.4.X-rc1 is a copy of the 1.4 branch. additional changes to 1.4.X-rcX are just to fix identified regressions introduced in that release, not general fixes from the 1.4 branch made while RCs are being tested |
| 15:21.14 | *** join/#asterisk clintc (~clintc@n128-227-87-199.xlate.ufl.edu) |
| 15:21.29 | russellb | and then once it's out, we start over, copy the 1.4 branch to 1.4.X-rc1, test for regressions, release |
| 15:21.44 | russellb | a monthly cycle |
| 15:21.57 | wcselby | russellb - i think i get it :) |
| 15:22.02 | russellb | be careful talking to leifmadsen when it's that time of the month |
| 15:22.08 | wcselby | russellb - it's slowly penetrating the fog in my head |
| 15:22.15 | wcselby | russellb - haha |
| 15:22.32 | rocksfrow | from asterisk CLI how can i debug why my voicemail emails arent sending out?" |
| 15:22.42 | leifmadsen | O.O |
| 15:22.43 | rocksfrow | -rvvvvvvvvvvv doesnt output anything about even trying to send an email |
| 15:22.55 | rocksfrow | is there some debugging i need to enable? |
| 15:22.58 | wcselby | rocksfrow - not sure you can....try reading through your /var/log/maillog file |
| 15:22.59 | rocksfrow | for voicemail details? |
| 15:23.01 | russellb | leifmadsen: i was explaining the monthly (release) cycle |
| 15:23.04 | rocksfrow | okay, thanks |
| 15:23.38 | leifmadsen | wcselby: all RC1's come directly from the 1.4 branch. If additional RCs are required, then a copy of the last RCx (RC1 for example) is copied to RCx+1 (RC2) and then the changes that are causing the RC to be released are merged to that tag. Of course the changes are also merged to the 1.4 branch. |
| 15:23.46 | leifmadsen | russellb: how dare you :) |
| 15:24.14 | leifmadsen | rocksfrow: nothing in the CLI for that -- check mail logs per wcselby |
| 15:24.38 | *** join/#asterisk thegoat (~thegoat@c-71-224-155-44.hsd1.pa.comcast.net) |
| 15:24.44 | russellb | leifmadsen: it's almost like we should write a blog post about our monthly development iterations |
| 15:24.51 | thegoat | anyone here using the chan-sccp-b drivers? |
| 15:24.52 | leifmadsen | oh snap yo |
| 15:25.00 | Kobaz | russellb: that sounds lovely |
| 15:25.05 | leifmadsen | russellb: hell, let me just go start on that like I said I would a month ago |
| 15:25.15 | leifmadsen | pabelanger: non-root install will have to wait for this afternoon I think |
| 15:25.18 | russellb | or i could like i said i would months ago ... but i won't |
| 15:25.24 | leifmadsen | :) |
| 15:25.33 | leifmadsen | well I do the releases, so I might as well write it up |
| 15:26.19 | rocksfrow | should have checked that his mail server wasn't blocking the emails first :-p |
| 15:26.30 | rocksfrow | my pbx box is just fine, lol |
| 15:26.43 | russellb | ohhhh technology, you so silly |
| 15:26.43 | wcselby | speaking of releases - any ideas when 1.8 will become an official release? |
| 15:26.48 | thegoat | i've been using the chan-sccp-b driver from sourceforge, and it likes to crash for some reason when i pick up a call. It causes asterisk to seg fault. but it's still more stable thatn the canned sccp driver....was just wondering if anyone else had the problem and if they found a solution |
| 15:26.52 | leifmadsen | wcselby: in the future |
| 15:26.54 | russellb | wcselby: hoping for beta within a month |
| 15:26.59 | russellb | then hoping to release in a few months |
| 15:27.01 | wcselby | leifmadsen - haha |
| 15:27.09 | wcselby | russellb - ahh, cool. |
| 15:27.15 | russellb | by October |
| 15:27.15 | leifmadsen | it'd be nice to have a 1.8.0 by Christmas |
| 15:27.19 | leifmadsen | :) |
| 15:27.23 | russellb | leifmadsen: AstriCon, dude! |
| 15:27.23 | wcselby | russellb - was just wondering if you were going to try and release by astricon |
| 15:27.26 | wcselby | haha |
| 15:27.32 | leifmadsen | russellb: ya I don't believe in artificial deadlines :) |
| 15:27.46 | russellb | all release deadlines are artificial |
| 15:27.46 | Kobaz | christmas... heh |
| 15:27.52 | russellb | but without them, it'll drag out forever |
| 15:28.05 | leifmadsen | amen |
| 15:28.26 | wcselby | i realize you guys are coders and not marketing people, but have you heard anything about a convention discount at the Gaylord Hotel for astricon? |
| 15:28.44 | russellb | if it's not on astricon.net, then i have no idea |
| 15:28.46 | *** join/#asterisk asamoah (~caio@190.244.49.108) |
| 15:29.15 | Kobaz | mmm, earlybird discount |
| 15:29.28 | wcselby | ooooh, call for papers is still open |
| 15:29.39 | wcselby | maybe I could do a talk on the asterisk release cycles.......... ;) |
| 15:31.33 | russellb | i'll probably include that in my talk :-p |
| 15:31.54 | wcselby | russellb - heh |
| 15:31.56 | beek | russellb: Same talk as last year's then? Or has the release cycle changed again? ;-) |
| 15:32.02 | pabelanger | leifmadsen: no problems |
| 15:32.14 | russellb | beek: we change it monthly |
| 15:32.23 | mallchin | hi guys, I'm getting lots of errors loading modules |
| 15:32.25 | mallchin | [Jun 17 16:29:58] WARNING[26314] loader.c: Error loading module 'app_getcpeid.so': /usr/lib/asterisk/modules/app_getcpeid.so: undefined symbol: ast_adsi_unload_session |
| 15:32.26 | russellb | not really ... it's the same, so I don't actually have anything to say about it |
| 15:32.33 | mallchin | :-/ |
| 15:32.39 | beek | russellb: Just like your underwear. |
| 15:32.48 | russellb | mallchin: you need to load res_adsi.so before that module |
| 15:33.07 | mallchin | russellb: thank you |
| 15:33.12 | russellb | which should happen automatically unless you use a custom modules.conf |
| 15:33.16 | russellb | in which case, add it :-) |
| 15:33.21 | mallchin | russellb: I have autoload set to yes, is loading all modules a good idea? |
| 15:33.46 | leifmadsen | I could probably talk about the release cycle stuff :D |
| 15:34.01 | russellb | well ... that's sort of a loaded question. it may or may not be a good idea, heh. |
| 15:34.06 | leifmadsen | that reminds me, I need to email jtodd to find out what talks, if any, I'm doing so I can determine how much I need to prepare |
| 15:34.21 | russellb | leifmadsen: did you put in a proposal? |
| 15:34.27 | leifmadsen | russellb: yes, I think I put in 2-3 |
| 15:34.31 | *** join/#asterisk mrchrisadams (~Adium@87-194-125-43.bethere.co.uk) |
| 15:36.04 | russellb | nice moves |
| 15:36.26 | leifmadsen | he'll probably come back with all 3 talks |
| 15:36.38 | leifmadsen | I put 3 in so they could pick 1 of the 3, heh |
| 15:36.43 | mallchin | russellb: hehe, well, I can't help it would be better to load support specifically, but I don't know what support I need |
| 15:37.04 | mallchin | russellb: are there particular modules one would usually load? I have no pri but am using SIP and IAX2 |
| 15:37.17 | russellb | i just use autoload, i'm lazy |
| 15:37.29 | leifmadsen | I use autoload along with menuselect to make sure I only enable what I need |
| 15:37.39 | mallchin | russellb: autoload works for me too, not if it errors like this |
| 15:37.40 | pabelanger | leifmadsen: About 'How to submit a bug report to issue tracker'? :) |
| 15:37.46 | leifmadsen | it's too hard to use a long list of load => directives and then have a problem when I try to upgrade |
| 15:37.55 | pabelanger | s/About/How about |
| 15:38.01 | leifmadsen | pabelanger: I spoke to some people about that thing last year informally actually |
| 15:38.30 | russellb | i think it would be fun to do a talk called "Silly Asterisk Demos" ... with a bunch of silly asterisk examples that involve asterisk participation |
| 15:38.37 | russellb | like the ones The_Boy_Wonder and I did this past weekend |
| 15:38.51 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.157) |
| 15:38.54 | russellb | a MeetMe() that people can call into where you get a random PITCH_SHIFT() before you enter |
| 15:38.55 | mallchin | russellb: http://pastebin.com/QNXQr9VM |
| 15:39.05 | Lantizia | Anyone got any experience with the Aastra XML input screens and perhaps why they don't go-the-hell-away after you press Done? |
| 15:39.10 | mallchin | russellb: does that look right? normal to get all those errors? |
| 15:39.13 | pabelanger | leifmadsen: or better yet! How NOT to submit a report. m17446 |
| 15:39.25 | russellb | no, that does not look normal. |
| 15:39.29 | russellb | looks like you're missing res_features? |
| 15:40.02 | russellb | and res_smdi |
| 15:40.16 | russellb | you broke it. |
| 15:40.32 | mallchin | :( |
| 15:40.45 | mallchin | hits it with a stick |
| 15:41.17 | wcselby | pabelanger - haha, I thought you were linking my issue from last night...... |
| 15:41.27 | wcselby | dodged a bullet |
| 15:41.39 | russellb | no, we made fun of your issue in a private chat room |
| 15:41.44 | pabelanger | lol |
| 15:41.47 | wcselby | russellb - i figured as much |
| 15:41.52 | russellb | (not really) |
| 15:41.58 | wcselby | russellb - you could have made fun of it in here, i wouldn't have minded |
| 15:42.10 | russellb | i don't even know what issue it is |
| 15:42.11 | pabelanger | wcselby: I don't even remember the issue |
| 15:42.34 | mallchin | russellb: does this look good? :) http://pastebin.com/1AgpjRfm |
| 15:42.38 | wcselby | lol, I wouldn't expect you guys to. |
| 15:42.54 | russellb | mallchin: yup. |
| 15:43.02 | russellb | mallchin: ls /usr/lib/asterisk/modules |
| 15:43.16 | russellb | also, what asterisk version? |
| 15:43.24 | wcselby | man I'm tired |
| 15:43.34 | russellb | ~thwack wcselby |
| 15:43.35 | infobot | ACTION bludgeons wcselby on the arm with a AS/400 |
| 15:43.41 | russellb | did that wake you up? |
| 15:43.51 | mallchin | russellb: http://pastebin.com/JfPeAtW1 |
| 15:43.55 | wcselby | ow, as/400's are big |
| 15:44.05 | wcselby | :P |
| 15:44.11 | mallchin | russellb: asterisk-1.4.22.1 |
| 15:44.16 | russellb | dies |
| 15:44.18 | russellb | that's ollllld |
| 15:44.24 | mallchin | I know :( |
| 15:44.37 | mallchin | but try getting my developers to move to 1.6.x |
| 15:44.43 | russellb | we've made 1130 changes to asterisk 1.4 since then :-p |
| 15:45.01 | russellb | how about 1.4.3X where X is the latest, heh |
| 15:45.04 | mallchin | they'd need to make 1130 to our code to use them :) |
| 15:45.07 | brycebaril | Heh you think that's old... I'm updating our software from a custom ast 1.0 -> 1.6 |
| 15:45.22 | russellb | 1.0 ftw |
| 15:45.28 | mallchin | I'm using Gentoo -- 1.4.22.1 is the latest in portage |
| 15:45.36 | mallchin | I'd have to compile 1.4.3.x from source |
| 15:45.42 | russellb | make it happen! |
| 15:46.01 | wcselby | 1.4.32, or 1.4.33-rc2 (which resolves my issue from 1.4.32) |
| 15:46.08 | mallchin | still, these should work on 1.4.22.1, right? |
| 15:46.08 | Corydon76-dig | mallchin: have you considered contributing some of those changes back to the project? |
| 15:46.28 | russellb | wget http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.4-latest.tar.gz ; tar xvzf asterisk-1.4-latest.tar.gz ; cd asterisk-1.4.3X ; ./configure ; make ; sudo make install |
| 15:46.29 | russellb | done! |
| 15:46.49 | russellb | mallchin: i can't remember all 1130 fixes since 1.4.22 :-) |
| 15:47.09 | mallchin | Corydon76-dig: they're not to the source code, but the dialplans and such have changed from 1.4 to 1.6 and would require re-writing them |
| 15:47.12 | russellb | we (dev team) just have a general policy of only spending time debugging problems on the latest release |
| 15:47.17 | wcselby | russellb - I think at this year's astricon you need to announce that after 1.6.2, the new version numbers will be named after odd african animals, progressing one letter up the alphabet for each release. |
| 15:47.40 | mallchin | russellb: that's fair enough, I should jolly them to move to 1.6, but that's another day |
| 15:47.43 | mallchin | (or year) |
| 15:47.47 | russellb | not 1.6, latest 1.4 |
| 15:47.52 | russellb | we still support 1.4 |
| 15:48.11 | mallchin | aah okay, well, I could try and get latest 1.4 installed |
| 15:48.17 | russellb | xlnt. |
| 15:48.22 | mallchin | but I'll need to compile it from source |
| 15:48.31 | mallchin | hugs portage |
| 15:48.31 | russellb | yes. |
| 15:48.33 | wcselby | asterisk aardvark. asterisk basilisk. asterisk chimera (okay so not from africa, but still, fun names) |
| 15:48.38 | russellb | see commands i already provided :-p |
| 15:48.48 | russellb | except fix the "X" in 3X |
| 15:49.00 | russellb | because i'm lazy and don't remember the number ... even though it's in the channel topic...... |
| 15:49.02 | leifmadsen | wcselby: you can be in charge of creating that list :) |
| 15:49.02 | russellb | 32. |
| 15:49.14 | leifmadsen | 1.4.32 :) |
| 15:49.19 | leifmadsen | oh you just said that |
| 15:49.20 | leifmadsen | lol |
| 15:49.24 | leifmadsen | goes back to writing |
| 15:49.29 | russellb | goes back to reading |
| 15:49.31 | mallchin | russellb: yep, I'll see if I can create an ebuild for it :) |
| 15:50.36 | russellb | closes IRC for a while ... |
| 15:51.06 | mallchin | okay, thanks guys! o/ |
| 15:51.58 | *** join/#asterisk Sipster (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 16:00.16 | *** part/#asterisk BlueJay (~foobar@b.in-server.net) |
| 16:10.37 | Qwell | pabelanger: I've given the "how not to report a bug" talk |
| 16:12.17 | Kobaz | Qwell: you mean like.... "asterisk broken, pls fix.. tks" |
| 16:12.39 | Qwell | one of the examples I used was "Asterisk are crash!" |
| 16:12.58 | Qwell | (yes, that's a real example) |
| 16:13.21 | Kobaz | heh |
| 16:13.41 | Kobaz | reminds me if an email i got when i was running a hosting business |
| 16:13.49 | Kobaz | "I am incrit in yo business" |
| 16:15.06 | thegoat | or i like ' can you reboot the internet' |
| 16:15.29 | wcselby | Kobaz - what the hell was that supposed to translate to? |
| 16:15.45 | thegoat | i am interested in your business? |
| 16:15.47 | thegoat | maybe |
| 16:15.52 | Kobaz | wcselby: hah |
| 16:20.53 | Naikrovek | incrit |
| 16:21.10 | Naikrovek | hm |
| 16:24.58 | *** join/#asterisk vadi (davi@unaffiliated/vadi) |
| 16:26.31 | vadi | Can configure asterisk to pick up the phone which is connected to the PC via a classic phone modem via serial port? |
| 16:29.46 | KavanS | wtf? |
| 16:30.23 | vadi | Is it a must use a PCI card |
| 16:30.36 | vadi | or can I use Asterisk to manage |
| 16:30.47 | vadi | pick up a phone line |
| 16:31.05 | wcselby | KavanS - i think he's trying to saying, "Can asterisk be configured to pick up the phone which is connected to the box asterisk is running on, if the phone is connected via an old-school serial modem" |
| 16:31.13 | wcselby | to which the answer would be, no, I don't hink so. |
| 16:31.15 | vadi | of classic phone modem connected via serial port? |
| 16:31.19 | wcselby | but I could have my translation wrong |
| 16:31.37 | KavanS | heh had a bit of trouble understanding that :P |
| 16:31.38 | tzafrir_laptop | vadi, do you use chan_dahdi / chan_zap ? |
| 16:32.10 | wcselby | the only way to connect an analog phone to the asterisk box is with an FXS port on a telephoney card, or an ATA sitting on the network |
| 16:32.21 | BarthezZ | !extension mobility |
| 16:32.23 | tzafrir_laptop | Also: is this an analog phone, or a line to your provider? |
| 16:32.36 | vadi | I would like experiment with Asterisk to pick the phone and so avoid the phone-spam (you know) |
| 16:32.37 | BarthezZ | hmm, what were the infobot commands? :p |
| 16:32.54 | vadi | <PROTECTED> |
| 16:33.01 | tzafrir_laptop | wcselby, you mention two methods. Which one of them is "the only one"? |
| 16:33.11 | tzafrir_laptop | ~infobot |
| 16:33.12 | infobot | tzafrir_laptop, i love abuse, feed me!, or whack, yo |
| 16:33.25 | BarthezZ | ~extension mobility |
| 16:33.35 | vadi | tzafrir_laptop, It is an analog phone which I use in my home |
| 16:33.36 | wcselby | tzafrir_laptop - haha, the only ways* |
| 16:33.37 | BarthezZ | hmm, he doesn't listen to me :p |
| 16:33.48 | tzafrir_laptop | ~fxsfxo |
| 16:33.49 | infobot | [~fxsfxo] An FXO port expects to receive dialtone and receive ring voltage. You can connect it to a PSTN line from the telco. An FXS port expects to provide dialtone and provide ring voltage. You can connect a phone or a fax to it. |
| 16:33.49 | kn0x | how many times should asterisk Retransmit SIP message if it is expecting ACK (UDP) |
| 16:34.11 | kn0x | i counted asterisk retransmitting 183 8 times |
| 16:34.24 | tzafrir_laptop | vadi, a modem is a FXO device. It's like a phone. If you want to connect a phone you need an FXS adapter |
| 16:34.46 | vadi | I see |
| 16:34.54 | tzafrir_laptop | vadi, that said, Asterisk does not support using serial modems as FXO devices |
| 16:34.54 | vadi | How much does it cost? The cheaper one? |
| 16:35.41 | vadi | <PROTECTED> |
| 16:36.01 | wcselby | it sounds like vadi is wanting asterisk to pick up incoming calls to combat phone spam |
| 16:36.17 | vadi | yes |
| 16:36.41 | kn0x | you will need an fxo device then vadi |
| 16:37.00 | vadi | any guess about FXO device price? |
| 16:37.03 | wcselby | so he would need an FXO card. if he also wants to connec this phone to the asterisk box as well (so he can eventually answer the call), he'll need an FXS port. So a two port card, one FXO, one FXS. |
| 16:37.12 | kn0x | vadi: sangoma b600 |
| 16:37.37 | vadi | searching for it |
| 16:37.38 | vadi | thanks |
| 16:37.38 | wcselby | meh, i'm not thinking things through properly |
| 16:37.47 | *** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no) |
| 16:37.56 | BarthezZ | oh btw tzafrir_laptop, I again want to thank you, your solution worked perfectly :) |
| 16:37.59 | wcselby | if he's got an analog phone, he'll need the FXO/FXS card. If he's got an IP phone, all he'll need is an FXO card. |
| 16:38.01 | BarthezZ | been running stable for about a week now :p |
| 16:40.40 | vadi | wcselby, My home have two phone ports, so I could use an FXO card with a Software Phone in my PC? |
| 16:41.36 | vadi | <PROTECTED> |
| 16:41.49 | vadi | <PROTECTED> |
| 16:42.17 | wcselby | vadi - you should speak with tzafrir_laptop, he's more up to speed on all the analog telephony stuff than I am. :) |
| 16:42.57 | *** join/#asterisk aidinb (~Aidin@wsip-98-190-45-225.oc.oc.cox.net) |
| 16:43.40 | vadi | thanks wcselby |
| 16:43.50 | tzafrir_laptop | vadi, that's one very specific modem. Basically it's a soft-modem for which the host-processing is already written |
| 16:44.36 | vadi | I would need to know the hardware which would be supported by my Debian GNU/Linux squeeze PC, and |
| 16:44.52 | vadi | the software phone advised to use |
| 16:44.58 | vadi | on that Debian box |
| 16:45.24 | *** join/#asterisk QaDeS (~mklaus@p54A18410.dip0.t-ipconnect.de) |
| 16:45.31 | tzafrir_laptop | install the package dahdi, and run 'dahdi_hardware' . If you don't see anything, you don't have the hardware :-( |
| 16:45.44 | vadi | thanks |
| 16:46.15 | *** join/#asterisk tzanger (~tzanger@gromit.mixdown.ca) |
| 16:46.20 | tzanger | good afternoon |
| 16:46.27 | tzanger | it's been a while since I've been in here :-) |
| 16:46.34 | tzafrir_laptop | hi! |
| 16:46.40 | tzanger | tzafrir_laptop: hello |
| 16:47.08 | tzanger | what was teh name of that third-party utility that nicely abstracted the AMI |
| 16:47.09 | *** join/#asterisk nighty^ (~nighty@x122091.ppp.asahi-net.or.jp) |
| 16:47.34 | tzanger | it basically held one connection to asterisk and did all the multiplexing itself, provided different machine-parseable connections, etc. |
| 16:47.43 | Qwell | astmanproxy? |
| 16:47.53 | tzanger | that's it! |
| 16:48.26 | tzanger | thanks Qwell |
| 16:48.35 | Qwell | np |
| 16:48.37 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
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| 16:51.33 | kn0x | anywhere to configure sip retransmit attempts? |
| 16:52.28 | kn0x | or at least a timeout for retransmits? |
| 16:54.10 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.219.69.dsl.dyn.forthnet.gr) |
| 16:54.58 | *** join/#asterisk raden_work (~jon@69-179-99-17.stat.centurytel.net) |
| 16:55.04 | kn0x | also any reason asterisk would send 183 Session Progress *AFTER* 200 OK ?? |
| 16:56.21 | *** join/#asterisk Jumpie (n3rdz@ip68-98-28-19.ph.ph.cox.net) |
| 16:59.06 | Naikrovek | launches Portal |
| 16:59.09 | Naikrovek | weeeeeeeee |
| 16:59.39 | *** join/#asterisk xayto (~xayto@202-89-161-53.static.dsl.amnet.net.au) |
| 17:01.53 | Jumpie | hehe |
| 17:02.15 | *** join/#asterisk eduzimrs (~eduzimrs@201.22.86.124.static.gvt.net.br) |
| 17:02.38 | wcselby | Naikrovek - did you see the portal 2 teaser? |
| 17:02.51 | wcselby | You monster.... |
| 17:04.09 | Jumpie | im waitin for it :) |
| 17:04.14 | Jumpie | portal was fun but it was too short imho |
| 17:06.14 | Naikrovek | yes i saw the teaser |
| 17:06.22 | Naikrovek | there are some decent gameplay vids on youtube as well |
| 17:06.30 | Naikrovek | portal 1 was an experiment |
| 17:06.37 | Naikrovek | they didn't expect anything to come of it |
| 17:06.42 | Naikrovek | kaboom - hugest game they've done |
| 17:07.34 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 17:09.06 | Naikrovek | i love those turrets too: "i don't blame you" |
| 17:10.50 | Qwell | "Are you still there?" |
| 17:10.58 | Qwell | "There you are!" |
| 17:11.11 | Qwell | the audio is what really made that game ;p |
| 17:12.49 | wcselby | Qwell - yeah, that was awesome |
| 17:12.51 | wcselby | that, and the cake |
| 17:13.16 | wcselby | i was talking to a buddy, and he was like "yeah, I'm almost done with the game, I'm on the last puzzle I think..." I started laughing at him |
| 17:13.37 | *** join/#asterisk ariel_ (~chatzilla@c-24-127-196-248.hsd1.fl.comcast.net) |
| 17:15.10 | *** join/#asterisk aidinb (~Aidin@wsip-98-190-45-225.oc.oc.cox.net) |
| 17:18.15 | Naikrovek | lol |
| 17:18.41 | Naikrovek | guy i work with here is on the glados fight |
| 17:18.46 | Naikrovek | i'm like come on |
| 17:19.05 | Naikrovek | missile turret, portals, enemy... how is that hard |
| 17:19.31 | Naikrovek | then he goes "what missile" |
| 17:19.36 | Sedorox | When calling Queue() and passing an AGI to it, is there a way to pass parameters to the called AGI? |
| 17:21.47 | Naikrovek | i play that whole game over and over just to listen to glados harass me |
| 17:22.33 | Naikrovek | Sedorox: yes |
| 17:22.43 | Naikrovek | i think |
| 17:23.03 | Sedorox | agi(aginame.agi,arguments), instead of just the aginame.agi? |
| 17:23.09 | Sedorox | ( I just came across that) |
| 17:23.11 | *** join/#asterisk neurosys (~neurosys@adsl-233-64-31.mia.bellsouth.net) |
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| 17:25.05 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 17:27.14 | ryanlin | anyone familiar with the dialplan in cme? |
| 17:27.18 | ryanlin | callmanager express |
| 17:31.15 | *** join/#asterisk t_dot_zilla (~chatzilla@firewall-a.buf.ny.i-evolve.net) |
| 17:32.11 | t_dot_zilla | can you adjust the audio volume of calls in asterisk? |
| 17:33.05 | angryuser | zap has a gain |
| 17:33.13 | angryuser | i suppose |
| 17:33.27 | angryuser | PAP2T devices has a gain |
| 17:33.29 | *** join/#asterisk Benwa (~Benwa@ip-62-235-220-239.dsl.scarlet.be) |
| 17:33.33 | t_dot_zilla | we're getting complaints our MOH is too loud, so we lowered the volume on the actual wav files, but in asterisk they sound exactly the same |
| 17:33.57 | angryuser | t_dot_zilla, from different clients ? |
| 17:34.08 | t_dot_zilla | yeah |
| 17:34.30 | angryuser | hm, how have you lowered the volume ? |
| 17:34.47 | t_dot_zilla | in audacity, i lowered the gain significantly |
| 17:35.06 | angryuser | that should do the trick normally |
| 17:36.00 | angryuser | you can not lower the volume of sip channels, thats for sure |
| 17:36.14 | t_dot_zilla | would asterisk automatically adjust the volume of the MOH ? |
| 17:36.24 | angryuser | t_dot_zilla, no |
| 17:40.06 | ChannelZ | what format are the audio files in? |
| 17:42.19 | *** join/#asterisk buttons840 (~buttons84@c-76-27-4-93.hsd1.ut.comcast.net) |
| 17:49.01 | Kobaz | heh |
| 17:49.18 | *** join/#asterisk btsteve (~tstevens@24-196-234-39.dhcp.gwnt.ga.charter.com) |
| 17:49.56 | *** join/#asterisk btsteve (~tstevens@24-196-234-39.dhcp.gwnt.ga.charter.com) |
| 17:51.36 | t_dot_zilla | PCM 8kHz Mono 16bit 128kbps wav files |
| 17:51.59 | t_dot_zilla | ChannelZ: what is the recommonded format ? |
| 17:52.14 | [TK]D-Fender | t_dot_zilla: THE FORMAT YOUR CHANNELS WILL USE |
| 17:53.39 | tzafrir_laptop | t_dot_zilla, is disk space an issue? |
| 17:53.47 | *** join/#asterisk italorossi (~italoross@201.76.154.130.intranet.digi.com.br) |
| 17:54.10 | t_dot_zilla | tzafrir_laptop: no |
| 17:54.23 | tzafrir_laptop | t_dot_zilla, also note that you can have the samee file in multiple formats |
| 17:54.50 | t_dot_zilla | tzafrir_laptop: what purpose would that serve? |
| 17:55.09 | tzafrir_laptop | if remote users connect through gsm, having gsm files there will save on transcoding |
| 17:58.08 | *** join/#asterisk neurosys (~neurosys@adsl-233-64-31.mia.bellsouth.net) |
| 17:58.22 | *** join/#asterisk grapsus (~grapsus@che21-2-82-245-89-120.fbx.proxad.net) |
| 17:58.32 | grapsus | Hi! |
| 17:59.13 | *** join/#asterisk fifer (~fifer@67.208.108.228) |
| 17:59.26 | grapsus | Is it possible to force asterisk to reply to SIP messages to the IP adress in the UDP header and not the one in SIP ? |
| 17:59.43 | fifer | Anyone know of a way to control the mic volume on/for an individual Aastra phone? |
| 17:59.59 | *** join/#asterisk x-demon (xdemon@2001:ba8:1f1:f0b8:216:5eff:fe00:135) |
| 18:00.08 | *** join/#asterisk Slugs_ (~Slugs@unaffiliated/slugs) |
| 18:00.13 | Slugs_ | hi |
| 18:00.24 | t_dot_zilla | tzafrir_laptop: most calls are using g711(ulaw), if we put the MOH in that format, would it still be able to transcode for calls that use other codecs ? |
| 18:00.25 | fifer | I'm specifically dealing with 6731i and 6757i phones |
| 18:01.26 | tzafrir_laptop | t_dot_zilla, transcoding between that and slinear or wav is minimal |
| 18:01.59 | tzafrir_laptop | What I like about wav files is that they have proper headers and thus easier to play in other tools |
| 18:02.01 | Slugs_ | can somebody identify the error, here is a error log of a non working ext and a working one. face_ears/136 |
| 18:02.22 | Slugs_ | can somebody identify the error, here is a error log of a non working ext and a working one. http://pastebin.com/EsqGPYc7 |
| 18:02.37 | t_dot_zilla | tzafrir_laptop: but when you have 1000 calls on MOH, it could start to lose quality. so it will be able to transcode ulaw to calls using other codecs ? |
| 18:02.40 | *** join/#asterisk mrchrisadams (~Adium@78-105-1-158.zone3.bethere.co.uk) |
| 18:03.42 | tzafrir_laptop | t_dot_zilla, if so, slap in ulaw files as well |
| 18:04.45 | t_dot_zilla | tzafrir_laptop: will asterisk know to use the ulaw instead of wav on g711 calls? |
| 18:10.40 | *** join/#asterisk neurosys (~neurosys@adsl-77-28-231.mia.bellsouth.net) |
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| 18:13.27 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 18:17.15 | [TK]D-Fender | Slugs_: clearly the dialplan doesn't have a match |
| 18:19.39 | *** join/#asterisk Arsenick (~y@modemcable230.231-70-69.static.videotron.ca) |
| 18:21.36 | *** join/#asterisk neurosys (~neurosys@adsl-233-64-134.mia.bellsouth.net) |
| 18:24.03 | t_dot_zilla | for MOH will asterisk know to use the ulaw instead of wav on g711 calls? |
| 18:24.21 | Qwell | why would it know that? |
| 18:24.25 | fifer | Found it: "headset tx gain" config file parameter |
| 18:24.39 | wcselby | anyone here use queuemetrics much? |
| 18:24.48 | t_dot_zilla | because the call uses ulaw |
| 18:25.15 | Qwell | sure |
| 18:25.26 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 18:26.09 | t_dot_zilla | my question is, if i have two files, lets say moh.wav and moh.ulaw, a caller is using the ulaw codec and is put on hold, will the caller hear the wav or the ulaw ? |
| 18:28.24 | *** join/#asterisk SaiSoma (~chatzilla@client105.jdcc.edu) |
| 18:31.37 | wcselby | anyone know how to setup queuemetrics so that only certain people can view stats for certain queues? |
| 18:36.17 | p3nguin | t_dot_zilla: Put the call on hold and use lsof to find the open file. |
| 18:37.35 | wcselby | doh, nevermind, fount it in their FAQ |
| 18:39.47 | *** join/#asterisk ccesario (~ccesario@189-19-6-236.dsl.telesp.net.br) |
| 18:40.37 | *** join/#asterisk neurosys (~neurosys@adsl-8-187-124.mia.bellsouth.net) |
| 18:40.51 | [TK]D-Fender | t_dot_zilla: ulaw clearly |
| 18:43.38 | p3nguin | t_dot_zilla: Something like the following could be useful: lsof -u asterisk |grep -i "sln\|mp3\|ulaw\|wav" |
| 18:43.51 | p3nguin | This helps me find which sound files are being used. |
| 18:44.43 | neurosys | Have remote phones that are disconnecting exactly 20 secs after retriving parked calls. I'm stumped :( |
| 18:44.44 | neurosys | http://pastebin.ca/1885256 |
| 18:54.41 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 18:55.10 | *** join/#asterisk Traderz (~traderz@173-161-87-147-Illinois.hfc.comcastbusiness.net) |
| 18:56.13 | Traderz | can anyone share there list of good voip/sip providers that can be used for unlimited business service and would be helpful to be able to pass my own caller id.. i know about broadvoice but looking for other suggestions. |
| 18:56.31 | *** join/#asterisk DarkRift (~dark@modemcable015.68-200-24.mc.videotron.ca) |
| 18:59.01 | wcselby | ~itsplist |
| 18:59.16 | wcselby | ~itsp-list |
| 18:59.16 | infobot | itsp-list is, like, Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 18:59.30 | wcselby | Traderz ^^^ |
| 18:59.35 | Traderz | thanks |
| 19:01.52 | [TK]D-Fender | "unlimited" isn't, and isn't often worth it |
| 19:02.23 | neurosys | [TK]D-Fender: Any ideas on mine ? |
| 19:02.42 | [TK]D-Fender | neurosys: Nope |
| 19:04.04 | *** join/#asterisk TheDavidFactor (~chatzilla@nc-71-52-236-209.dhcp.embarqhsd.net) |
| 19:09.44 | *** join/#asterisk citywok (~chatzilla@67-134-194-33.dia.static.qwest.net) |
| 19:10.08 | *** join/#asterisk x303 (~x303@187.159.121.70.cfl.res.rr.com) |
| 19:11.28 | chuckf | wonders why vitelity is at the end of that list |
| 19:13.26 | p3nguin | ~itsplist-us |
| 19:13.26 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 19:13.28 | *** join/#asterisk rocksfrow (~kyle@74-92-146-97-WashingtonDC.hfc.comcastbusiness.net) |
| 19:13.57 | rocksfrow | could anybody assist me in figuring out why lspci says 'unknown device' for my digium card |
| 19:14.00 | rocksfrow | everything is working great |
| 19:14.12 | p3nguin | Did you update the data? |
| 19:14.13 | Qwell | update-pciids |
| 19:14.20 | p3nguin | points at qwell |
| 19:14.21 | rocksfrow | but, i just got a second _identical_ server... and put the same card in it, and it shows the correct name of the card int hat one |
| 19:14.29 | rocksfrow | update-pciids? |
| 19:14.31 | rocksfrow | googling |
| 19:14.32 | Qwell | clearly it's not identical |
| 19:14.35 | Jumpie | lol |
| 19:14.37 | rocksfrow | ...it is identical |
| 19:14.44 | rocksfrow | i just figure w/e setup i did before |
| 19:14.45 | Qwell | ...clearly it isn't |
| 19:14.46 | rocksfrow | isnt' identical |
| 19:14.48 | p3nguin | It's not identical, since there is a difference. |
| 19:14.50 | rocksfrow | the hardware is the same |
| 19:14.52 | rocksfrow | .... |
| 19:15.00 | rocksfrow | the only difference is my pciids aren't udpated ont he one server? |
| 19:15.05 | p3nguin | maybe |
| 19:15.08 | rocksfrow | heh |
| 19:15.10 | Qwell | probably not |
| 19:15.11 | rocksfrow | maybe? |
| 19:15.11 | p3nguin | Don't make us keep guessing. |
| 19:15.12 | rocksfrow | te220b |
| 19:15.19 | tzafrir_laptop | lspci -q |
| 19:15.20 | p3nguin | Update the data and then let us know. |
| 19:15.43 | rocksfrow | told ya ;) |
| 19:15.50 | rocksfrow | you guys are awesome, thanks for hte help |
| 19:15.56 | rocksfrow | update-pciids did the trick |
| 19:15.58 | rocksfrow | now they're IDENTICAL |
| 19:16.00 | rocksfrow | :-p |
| 19:16.02 | Qwell | no they aren't |
| 19:16.03 | p3nguin | Now they might be identical. |
| 19:16.06 | rocksfrow | ...wtf? |
| 19:16.07 | rocksfrow | lol |
| 19:16.15 | rocksfrow | how aren't they |
| 19:16.31 | p3nguin | Run the update on both at the exact some microsecond. |
| 19:16.35 | tzafrir_laptop | rocksfrow, what does dahdi_hardware say about that card? |
| 19:17.40 | rocksfrow | ..there is no more issue |
| 19:17.44 | rocksfrow | they are identical |
| 19:17.50 | rocksfrow | just was asking why qwell is saying they are not |
| 19:17.56 | rocksfrow | by identical, imeant identical as far as lspci output |
| 19:18.13 | *** join/#asterisk lost_soul (shackett@devio.us) |
| 19:18.26 | Jumpie | anybody know how asterisk will perform in 2.6.18-164.11.1.el5xen? |
| 19:18.36 | Jumpie | any issues with xen kernel? |
| 19:19.27 | *** join/#asterisk neurosys (~neurosys@adsl-233-241-148.mia.bellsouth.net) |
| 19:20.52 | *** part/#asterisk neurosys (~neurosys@adsl-233-241-148.mia.bellsouth.net) |
| 19:20.58 | *** join/#asterisk neurosys (~neurosys@adsl-233-241-148.mia.bellsouth.net) |
| 19:22.45 | tzafrir_laptop | Jumpie, not if you use latest DAHDI |
| 19:24.09 | Jumpie | well i dont really need dahdi |
| 19:24.09 | *** part/#asterisk neurosys (~neurosys@adsl-233-241-148.mia.bellsouth.net) |
| 19:24.15 | Jumpie | although i may need it for timing? |
| 19:24.20 | Jumpie | this will be straight ip |
| 19:24.29 | Jumpie | do you still need dahdi for meetme? |
| 19:26.51 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 19:28.18 | *** join/#asterisk buttons840 (~buttons84@c-76-27-4-93.hsd1.ut.comcast.net) |
| 19:29.38 | wcselby | Jumpie - yes dahdi is still needed for meetme |
| 19:30.16 | Jumpie | so i just need to get a slightly older version |
| 19:30.37 | [TK]D-Fender | Or stop using MeetMe |
| 19:30.38 | WIMPy | But you can use Confbridge instead of Meetme. |
| 19:30.44 | Jumpie | yea |
| 19:30.47 | Jumpie | i really dont care about meetme either |
| 19:30.59 | Jumpie | i was just concerned about compilation/stability issues with xen |
| 19:31.00 | WIMPy | Unless you need enter/leave sounds, wich don't seem to be working there. |
| 19:31.15 | t_dot_zilla | fyi, asterisk does not automatically choose ulaw instead of wav MOH |
| 19:31.17 | Jumpie | i wanna tweak this image and replicate the snapshots later |
| 19:31.58 | t_dot_zilla | seems to me that asterisk chooses the MOH randomly, does not prioritize according to codec |
| 19:33.06 | *** join/#asterisk blaines (~blaines@ip70-190-67-126.ph.ph.cox.net) |
| 19:34.39 | *** join/#asterisk blaines (~blaines@ip70-190-67-126.ph.ph.cox.net) |
| 19:35.22 | [TK]D-Fender | t_dot_zilla: show us |
| 19:36.06 | t_dot_zilla | how ? i just called from my cellphone (g711) and it played a wav instead of the ulaw that is there |
| 19:36.39 | *** join/#asterisk blaines (~blaines@ip70-190-67-126.ph.ph.cox.net) |
| 19:37.47 | [TK]D-Fender | t_dot_zilla: show us <------------ |
| 19:38.06 | t_dot_zilla | [TK]D-Fender: show you what? |
| 19:38.31 | *** join/#asterisk ZeXr0 (~ZeXr0@modemcable005.121-82-70.mc.videotron.ca) |
| 19:38.51 | [TK]D-Fender | t_dot_zilla: Everything related to this clearly... |
| 19:39.01 | *** join/#asterisk blaines (~blaines@ip70-190-67-126.ph.ph.cox.net) |
| 19:39.03 | *** join/#asterisk AlHafoudh (~AlHafoudh@adsl-dyn190.78-98-189.t-com.sk) |
| 19:39.19 | t_dot_zilla | asterisk 9617 asterisk 34r REG 253,0 4464220 1599994 /var/lib/asterisk/moh/macroform-the_simplicity.wav |
| 19:39.32 | t_dot_zilla | not using ulaw |
| 19:39.45 | *** join/#asterisk nightwalk (~nightwalk@a-1-68.med-web.com) |
| 19:39.54 | ZeXr0 | How fun is that... I've setup Asterisk and everything at the office. Everythings seems to work fine and all. Doing the installation of the server at the client's datacenter. It doesn't work anymore. Unable to access the redfone. And no way to debug remotly because the computer doesn't have access to the internet ... |
| 19:40.44 | *** join/#asterisk Tarantulafudge (~Tarantula@Mail.securenets.us) |
| 19:40.53 | citywok | make sure you didn't set the IP of your office in the asterisk config file, and now it's trying to bind to a non-existent address? |
| 19:41.11 | p3nguin | t_dot_zilla: You have macroform-the_simplicity.wav and macroform-the_simplicity.ulaw in the same directory? |
| 19:41.13 | [TK]D-Fender | t_dot_zilla: Everything related to this clearly... <------------------ |
| 19:41.18 | *** join/#asterisk blaines (~blaines@ip70-190-67-126.ph.ph.cox.net) |
| 19:41.26 | Tarantulafudge | I need some help troubleshooting occasional calls not being made by the AMI originate |
| 19:41.28 | t_dot_zilla | p3nguin: yes |
| 19:41.53 | citywok | Tarantulafudge: post the AMI output on pastebin |
| 19:41.58 | [TK]D-Fender | ZeXr0: And you brought the redfone there set it all up, changing, MAC's, etc? |
| 19:42.09 | Tarantulafudge | citywok, there is no ami output |
| 19:42.13 | t_dot_zilla | asterisk 9617 asterisk 34r REG 253,0 2573886 424341 /var/lib/asterisk/moh/reno_project-system.ulaw |
| 19:42.16 | t_dot_zilla | it's random |
| 19:42.26 | citywok | how is there no output? if there's no output then it must not have received the command |
| 19:42.49 | t_dot_zilla | i just called 6 times and each time used wav, except just now it used the ulaw |
| 19:42.51 | Tarantulafudge | citywok, its the asyncronous originate AMI command via StarPy |
| 19:43.08 | Tarantulafudge | I'll post it |
| 19:43.22 | citywok | telnet in to it normally then, and log the output |
| 19:43.34 | [TK]D-Fender | t_dot_zilla: Everything related to this clearly... <------------------ |
| 19:43.35 | citywok | i'd suggest using putty so you can log it |
| 19:43.55 | t_dot_zilla | [TK]D-Fender: what else do youwant to know ? |
| 19:44.05 | [TK]D-Fender | t_is that call you can come up with? |
| 19:44.10 | [TK]D-Fender | t_dot_zilla: is that call you can come up with? |
| 19:44.13 | *** join/#asterisk blaines (~blaines@ip70-190-67-126.ph.ph.cox.net) |
| 19:44.47 | t_dot_zilla | [TK]D-Fender: huh |
| 19:44.48 | t_dot_zilla | ? |
| 19:45.02 | [TK]D-Fender | t_dot_zilla: You show us 1 little line... |
| 19:45.04 | *** join/#asterisk neurosys (~neurosys@adsl-233-241-148.mia.bellsouth.net) |
| 19:45.13 | [TK]D-Fender | t_dot_zilla: I said show us everything related to this call |
| 19:45.24 | t_dot_zilla | that line is indicating what asterisk is using for MOH |
| 19:45.47 | Tarantulafudge | citywok, http://pastebin.org/337733 |
| 19:46.35 | *** join/#asterisk stonezone (~stonezone@rrcs-66-91-131-142.west.biz.rr.com) |
| 19:46.45 | citywok | voip must be a trunk to another server, so you dial sip/voip/${EXTEN} ? |
| 19:46.57 | t_dot_zilla | i'm calling the same number from my cellphone and each time asterisk is choosing wav or ulaw as MOH, it is not choosing ulaw everytime |
| 19:47.00 | [TK]D-Fender | moves on to more productive things... |
| 19:47.36 | Tarantulafudge | citywok, yeah |
| 19:47.42 | citywok | t_dot_zilla: it sounds like it's randomly selecting what file to use, probably the way it's configured |
| 19:48.38 | citywok | but like [TK]D-Fender said, because you didn't provide the full call log, without context we have no idea what it's doing. have a nice day :) |
| 19:48.57 | citywok | well, like i suggested, connect to the AMI manually and log the output |
| 19:49.02 | citywok | then you can see what happens |
| 19:49.21 | t_dot_zilla | well, i'm not concerned with it anymore, alls i know is asterisk does not prioritize MOH files according to call codecs |
| 19:49.40 | citywok | it's supposed to |
| 19:50.00 | citywok | so all you know is on your system it isn't prioriting in a consistent manner. |
| 19:50.39 | [TK]D-Fender | Knowledge without really looking != knowledge |
| 19:51.08 | *** join/#asterisk grumpyoldman (~meanderis@buster.coredial.com) |
| 19:51.20 | citywok | now please stop filling the channel with stupid comments. if you actually want help from [TK]D-Fender i'd suggest you provide him the informatino he asks for. |
| 19:52.17 | [TK]D-Fender | Not "stupid", just "unvalidated" |
| 19:52.39 | grumpyoldman | anyone know if SRV records are being properly sorted in 1.4.29 ? config file text says no, it looks like there is code for sorting but It keeps alternating destination in practice. |
| 19:54.58 | neurosys | call drops after placed in prking after exactly 20 secs. any ideas? http://pastebin.ca/1885256 |
| 19:56.31 | Tarantulafudge | citywok, how do I do originates from telnet? I'm logged in as I type this |
| 19:56.48 | Tarantulafudge | Action: Originate ? |
| 19:58.01 | *** join/#asterisk aidinb (~Aidin@wsip-98-190-45-225.oc.oc.cox.net) |
| 19:58.04 | Tarantulafudge | The unmade calls always seem to occur when I'm making more than one call at a time |
| 19:58.20 | Tarantulafudge | but only sometimes |
| 19:58.21 | [TK]D-Fender | Tarantulafudge: LOGIN FIRST <-------- |
| 19:59.35 | Tarantulafudge | [TK]D-Fender, I'm logged in now |
| 20:00.06 | carrar | I'm logged in too |
| 20:00.13 | [TK]D-Fender | h4x)r |
| 20:00.34 | carrar | I've got a dozen laser cats pointed at you |
| 20:02.01 | [TK]D-Fender | counters usign sharks with frikken lasers on their heads |
| 20:02.20 | carrar | land sharks no less |
| 20:02.25 | kn0x | dialog_unlink_all: Unable to cancel schedule ID |
| 20:02.31 | kn0x | any idea what thats all about? |
| 20:02.48 | Tarantulafudge | Oh I see how this works |
| 20:03.01 | Tarantulafudge | this should be helpfull, thanks |
| 20:03.28 | p3nguin | t_dot_zilla: Can you provide a sip debug of a call which has chosen ulaw for moh as well as a sip debug of a call which has chosen wav for moh? |
| 20:05.19 | p3nguin | t_dot_zilla: If you don't care WHY it happens, delete the files which are of the format that you do not want to use for moh. |
| 20:06.16 | t_dot_zilla | p3nguin: i'd like to have the wavs in there incase some calls from a polycom that uses better codecs |
| 20:07.10 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 20:07.34 | p3nguin | t_dot_zilla: What codec might the phones use that would be better than ulaw? |
| 20:07.59 | t_dot_zilla | g722 |
| 20:08.20 | p3nguin | t_dot_zilla: Are you supporting g722 on your Asterisk system? |
| 20:08.27 | t_dot_zilla | yes |
| 20:08.35 | *** join/#asterisk cusco (~trilili@213.63.137.210) |
| 20:08.37 | cusco | hi |
| 20:08.55 | cusco | if PRI tells me hangupcause 27 |
| 20:09.13 | cusco | can I validade that hangup cause in h extension to playback(some audio) ? |
| 20:09.52 | leifmadsen | t_dot_zilla: I have a feeling the translation cost between ulaw and adpcm to slin is probably the same |
| 20:10.10 | leifmadsen | cusco: you can't play audio from the 'h' extension -- the call is hung up |
| 20:10.30 | cusco | can I have a goto in h extension? |
| 20:10.32 | cusco | ah |
| 20:10.34 | cusco | oops nevermind |
| 20:10.51 | cusco | so for every extension I must add that validation? |
| 20:10.51 | leifmadsen | t_dot_zilla: on my system it is the same, so it may just be random which gets picked because neither is preferred over the other |
| 20:11.04 | p3nguin | h can run commands, but it wouldn't be able to play a sound since the channel doesn't exist anymore. |
| 20:11.13 | leifmadsen | right |
| 20:11.13 | cusco | is there an extension that the call might go triough I get hungup cause 27 in one end |
| 20:11.15 | *** join/#asterisk luckyaba (~lucky@ip72-194-218-169.sb.sd.cox.net) |
| 20:11.19 | leifmadsen | what that guy said :) |
| 20:11.33 | leifmadsen | 'h' still works, just no audio |
| 20:11.37 | cusco | yes |
| 20:11.55 | cusco | so if I would like to make user listen so some audio if sip says BAD GATEWAY |
| 20:12.15 | p3nguin | That should already happen. |
| 20:12.26 | p3nguin | It should play the congestion tones. |
| 20:12.44 | cusco | ouch, I don't have congestion tones |
| 20:12.55 | p3nguin | I'd be surprised if you don't. |
| 20:13.08 | cusco | well, this configuration was mutilated before I knew about asterisk |
| 20:13.12 | leifmadsen | then use something like the ${DIALSTATUS} variable after you try to Dial() |
| 20:13.28 | cusco | I will look how to play congestion tones by default instead |
| 20:13.42 | cusco | else I have to validate ${DIALSTATUS} in 30 different extensions |
| 20:13.54 | p3nguin | That's what macros are for. |
| 20:14.01 | cusco | hmmm |
| 20:14.06 | cusco | right.. |
| 20:14.32 | cusco | but our 30 extensions are not calling any macro right now.. so I still have to call it 30 times |
| 20:15.01 | cusco | right? |
| 20:15.07 | p3nguin | maybe |
| 20:15.21 | t_dot_zilla | http://pastebin.com/d700L3cE |
| 20:15.41 | t_dot_zilla | p3nguin: MOH with wav file http://pastebin.com/d700L3cE |
| 20:16.20 | cusco | so... congestion dialtones should be played automatically, right? |
| 20:17.17 | p3nguin | Not dial tones, but congestion tones. |
| 20:17.37 | p3nguin | When there is a circuit error, congestion tones are almost always played. |
| 20:21.49 | t_dot_zilla | MOH with ulaw file: http://pastebin.com/SySgZF2v |
| 20:36.12 | leifmadsen | t_dot_zilla: like I said earlier, there is no weight difference between adpcm and ulaw |
| 20:36.14 | *** join/#asterisk bodie (~bodie@cm-84.215.50.129.getinternet.no) |
| 20:36.34 | bodie | Hi, is there someone who uses appkonference for video calls? |
| 20:43.40 | bodie | so no one uses appkonference? |
| 20:50.22 | *** part/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 20:50.59 | *** join/#asterisk Ast001 (~Ast001@cable-89-216-190-211.dynamic.sbb.rs) |
| 20:51.27 | Ast001 | hello can someone tell me what is the last line of asterisk originate command's response ? |
| 20:51.40 | *** join/#asterisk obnauticus (~obnauticu@about/windows/regular/obnauticus) |
| 20:51.46 | Ast001 | I mean AMI originate command |
| 20:52.05 | *** join/#asterisk Micc (~quassel@c-98-225-57-96.hsd1.wa.comcast.net) |
| 20:52.32 | *** join/#asterisk Benwa (~Benwa@ip-62-235-220-239.dsl.scarlet.be) |
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| 20:53.49 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 20:54.28 | Micc | I've got a problem with DTMF tones out our PRI. I think it worked before upgrading dahdi to the latest version, but I'm not sure. What kinds of things should I check? It seems like the tones are all the same no matter which number is pressed. |
| 20:55.46 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 20:56.45 | *** join/#asterisk stonezone (~stonezone@rrcs-66-91-131-142.west.biz.rr.com) |
| 20:58.09 | Ast001 | Can I catch response from AMI originate if I use originate with async ? |
| 20:59.18 | *** join/#asterisk rocksfrow (~kyle@74-92-146-97-WashingtonDC.hfc.comcastbusiness.net) |
| 20:59.33 | rocksfrow | hey guys, am i missing something? yum install libpri says pri is installed, but asterisk CLI says no such command 'pri' |
| 21:00.52 | WIMPy | rocksfrow: You're missing a driver for your hardware, I guess. Like dahdi. |
| 21:01.54 | rocksfrow | dahdi_hardware outputs the correct card |
| 21:01.54 | *** join/#asterisk CraigW76_ (~techcaw@addr33.mimc.com) |
| 21:01.54 | *** part/#asterisk bodie (~bodie@cm-84.215.50.129.getinternet.no) |
| 21:02.28 | WIMPy | Is it configured? is chan_dahdi loaded? If not, what happens if you turn up debug and try to load it manually? |
| 21:03.24 | [TK]D-Fender | rocksfrow: and your configs? What about "dahdi show channels", "dahdi show status"? |
| 21:03.54 | rocksfrow | hrm, no such command for dahdi either |
| 21:04.00 | rocksfrow | i guess i just haven't configured dahdi yet |
| 21:04.05 | rocksfrow | it's an asterisknow box |
| 21:04.07 | [TK]D-Fender | rocksfrow: You "guess"? |
| 21:04.20 | rocksfrow | [TK]D-Fender, sort of figured it would out of box |
| 21:04.30 | [TK]D-Fender | rocksfrow: what do you see when you check dahdi from OS CLI? |
| 21:04.40 | rocksfrow | no such command dahdi |
| 21:04.45 | rocksfrow | oh wait |
| 21:04.47 | rocksfrow | from OS cli |
| 21:04.51 | [TK]D-Fender | [17:04]<rocksfrow>i guess i just haven't configured dahdi yet <- YOU configure DAHIDI... |
| 21:04.55 | rocksfrow | elaborate on 'check dahdi' |
| 21:04.59 | [TK]D-Fender | rocksfrow:dahdi_cfg -vvvv |
| 21:05.23 | rocksfrow | eek |
| 21:05.33 | rocksfrow | 0 channels to configure |
| 21:05.39 | rocksfrow | configuration is empty |
| 21:05.42 | rocksfrow | dahdi_genconf? |
| 21:06.32 | rocksfrow | aha |
| 21:06.33 | rocksfrow | that did the trick |
| 21:07.15 | rocksfrow | well, atleast dahdi_cfg is showing me the spans now |
| 21:07.23 | rocksfrow | i still dont' have the management commands within asterisk CLI.. |
| 21:08.46 | rocksfrow | [TK]D-Fender, any more help? :) |
| 21:08.50 | [TK]D-Fender | rocksfrow: And I'm sure your configs are still far from complete |
| 21:09.17 | rocksfrow | [TK]D-Fender, well, i've done a backup/restore using freepbx |
| 21:09.44 | rocksfrow | which is supposed to copy most of the configs |
| 21:09.44 | rocksfrow | i'm setting up a backup esrver |
| 21:09.44 | rocksfrow | for an already working setup |
| 21:09.44 | rocksfrow | so ill just have to go through all the configs manually i guess |
| 21:09.44 | rocksfrow | i just figured the freepbx restore got most of em |
| 21:09.45 | rocksfrow | i guess not the actual PRI config |
| 21:10.17 | rocksfrow | [TK]D-Fender, any tutorials? |
| 21:10.28 | rocksfrow | nvm, got it |
| 21:17.39 | rocksfrow | can somebody please help me figure out why i cant' use the pri or dahdi commands from asterisk CLI? |
| 21:17.46 | rocksfrow | i have my dahdi configuration working |
| 21:18.37 | WIMPy | Did you restart Asterisk after completing your dahdi configuration? |
| 21:20.33 | rocksfrow | no..lol |
| 21:20.57 | rocksfrow | hey! |
| 21:21.00 | rocksfrow | haha..i'm slow |
| 21:21.02 | rocksfrow | thanks bro. |
| 21:31.24 | *** join/#asterisk stonezone (~stonezone@rrcs-66-91-131-142.west.biz.rr.com) |
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| 21:48.19 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
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| 21:52.15 | Sedorox | so any hints for arguments being passed to AGI on queue()? AGI(agi,arg) doesn't wanna work, and I can't just comma after it, as thats other options for queue() |
| 21:52.42 | Sedorox | or is from queue() only agi, no arguments that we can specify being passed to it |
| 21:56.14 | [TK]D-Fender | Sedorox: What does Queue() have to do with AGI()? |
| 21:56.14 | ChannelZ | I think the args probably come in as AGI variables as opposed to regular argv type |
| 21:58.07 | ChannelZ | AGI Tx >> agi_arg_1: poop |
| 21:58.09 | ChannelZ | AGI Tx >> agi_arg_2: pee |
| 21:58.26 | ChannelZ | but I dunno what queue has to do with it either |
| 21:58.43 | *** join/#asterisk jcims (~chatzilla@oh-69-34-176-18.sta.embarqhsd.net) |
| 21:59.28 | jcims | any idea why, during sip registration, a cisco 7960 would respond to the proxy's 401 unauthorized with a ICMP port unreachable? response packet is going to same port as source packet from phone |
| 21:59.29 | ChannelZ | Oh. There's a Queue argument to run an AGI |
| 21:59.35 | *** join/#asterisk cesar_CR (~cesar@201.192.86.30) |
| 21:59.39 | ChannelZ | My guess is it probably doesn't support args. |
| 21:59.53 | jcims | phone never attempts to authenticate, as though it just doesn't see the 401 coming back, so it just keeps sending register packets |
| 22:00.33 | ChannelZ | firewally? |
| 22:02.17 | jcims | firewall in the middle doing nat (pat), but frames are making it back to the phone (sniffing via pc attached to phone) |
| 22:03.02 | WIMPy | Err, a PC attached to the phone should not be able to see them. |
| 22:03.44 | jcims | yeah, a long-standing gripe i have with the 7960's |
| 22:03.44 | jcims | hack the pc and you can sniff comms to the phone |
| 22:03.52 | jcims | there's a way to turn it off, it's just not default |
| 22:05.19 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 22:07.26 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 22:07.45 | jcims | gah, need a beer. this thing drives me nuts. |
| 22:07.46 | *** join/#asterisk mindCrime (~chatzilla@64.241.37.140) |
| 22:08.40 | jcims | $400 phone with no backlight on it. l8r g8rs |
| 22:08.53 | citywok | :heart: aastra 6757i |
| 22:09.00 | citywok | 10 line config versus 400000000 line |
| 22:09.16 | [TK]D-Fender | citywok: I don't care about the config, I care about the phone |
| 22:09.24 | [TK]D-Fender | Polycom > All |
| 22:09.52 | citywok | i could never a 7960 to work properly, it hated nat (just going from one 192.168 subnet to another) |
| 22:10.24 | citywok | i only played with a few cheap polycoms and i was unimpressed, their speakerphone was horrible. the cheap aastra had such an amazing speakerphone everybody wanted one. so we bought aastra's. |
| 22:12.53 | *** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk) |
| 22:14.55 | p3nguin | I guess I should feel lucky that my phones are on the same network as Asterisk. |
| 22:16.01 | citywok | i've got 5 locations to deal with |
| 22:16.20 | citywok | so i get to play the MPLS cloud game to. and one site is international (fun!) |
| 22:17.01 | fenrus | does your ISP handle all the routing? |
| 22:17.10 | fenrus | and the mpls part of it |
| 22:18.22 | fenrus | mpls-vpn is the bread and butter part of what isps sell today to companys all over.. |
| 22:18.44 | citywok | yea, it's glorious how simple it is |
| 22:19.08 | citywok | i configure it the same way as i would a normal edge router, but then i send all of my internal traffic at the same gateway, and it deals with it. |
| 22:19.39 | citywok | it figures out external vs internal traffic and routes to all my locations. no special MPLS enabled devices or anything, a 10 year old cisco will do the trick if you wish. |
| 22:20.09 | fenrus | yea, since the label switching is done with the isps equipment :) |
| 22:21.12 | citywok | and it's way cheaper than dealing with International point to points. 3mbit Intl E1/T1 was 10grand. we get 4mbit in the MPLS cloud for like 4grand. and the ping is 200ms instead of 260, and the routing is redundant (i cant tell you how many hong kong earthquakes have taken us down for 2-3 days) |
| 22:21.33 | fenrus | =) |
| 22:22.11 | fenrus | the only plus with sdh/pdh is the capacity being reserved |
| 22:24.22 | Micc | what settings affect dtmf on a pri? |
| 22:26.00 | *** join/#asterisk stonezone (~stonezone@rrcs-66-91-131-142.west.biz.rr.com) |
| 22:40.52 | Deeewayne | chanspying on 2 telemarketers trying to figure out what's wrong with their dialer application because they are getting pushed into my blackhole extension |
| 22:43.11 | KavanS | Deeewayne, what does your blackhole extension do? |
| 22:43.19 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 22:43.58 | Deeewayne | KavanS, loop forever while playing a lot of different silly asterisk prompts |
| 22:45.05 | KavanS | hehe ok |
| 22:47.36 | fenrus | Theese stuff should be published so that we all can use the without effort :D |
| 22:47.58 | citywok | KavanS: that's so mean. our outbund LD gets expensive! |
| 22:48.10 | citywok | you're wasting 1 penny a minute of their money, imagine the impact on their bottom line! lol |
| 22:49.23 | KavanS | citywok, we have a "blackhole" extension we transfer people to - it serves the function of adding someone to our blacklist playing martin from the simpsons saying "ha ha" then a movie-esque feature of playing a dial tone - then hanging up |
| 22:49.31 | KavanS | when they return to call us - of course they hear ss-noservice |
| 22:49.38 | KavanS | primarily for pesky callers :) |
| 22:49.51 | ChannelZ | I have a nice one that plays really bad elevator music, tells them the hold time is 7 hours 40 mins, etc. |
| 22:50.03 | citywok | lol, that's kind of funny. you should change it to "please remove me from all of your calling lists, further attempts will be recorded and pursued" |
| 22:50.14 | ChannelZ | they dont care about such things |
| 22:50.19 | KavanS | citywok, hehe yeah we watch the logs from time to time after we dump someone |
| 22:50.23 | citywok | then log the attempts :) |
| 22:50.25 | ChannelZ | I've had a collection service with the wrong number calling for weeks now |
| 22:50.26 | fenrus | In sweden we have an opt-out organization |
| 22:50.30 | KavanS | there are times we dump em - then de-blacklist while they are attempting |
| 22:50.34 | KavanS | then blame it on our "crazy PBX" |
| 22:50.35 | fenrus | telemarketing should be opt-in. |
| 22:51.00 | hardwire | t.38 is such a pita |
| 22:51.10 | ChannelZ | death to fax |
| 22:51.13 | KavanS | ChannelZ, the ss-noservice tone works for some people - but as you probably already know some people are not discouraged easily |
| 22:51.16 | fenrus | if i want to buy something, i'm not going to do it over the phone. |
| 22:51.16 | citywok | yea, i avoid fax like he plague |
| 22:51.18 | hardwire | ChannelZ: agreed |
| 22:51.24 | citywok | we just install a POTS line at each office and install a fax machine. life is good :) |
| 22:51.26 | KavanS | lol yep fax sucks |
| 22:51.30 | hardwire | why people in china awnt to use us as a fax destination I will never know |
| 22:51.34 | KavanS | only use fax because I have to |
| 22:52.02 | hardwire | KavanS: lies |
| 22:52.14 | hardwire | we can all simply refuse to allow fax |
| 22:52.20 | hardwire | it's an "all in" death to fax |
| 22:52.30 | ChannelZ | KavanS: At home I do Zapateller and that |
| 22:52.41 | hardwire | it will either breath new life into copper pstn services or force everybody to scan things |
| 22:52.44 | hardwire | :P |
| 22:52.47 | ChannelZ | Although now it's programmed to not even answer() the line |
| 22:52.59 | KavanS | ChannelZ, yeah it is nice :) - I do not have a home line myself so I just use google voice # for such things |
| 22:53.08 | KavanS | "oh my home number? - sure!" |
| 22:53.35 | *** join/#asterisk war9407 (war@liquidswords.org) |
| 22:54.20 | *** join/#asterisk rocksfrow (~kyle@74-92-146-97-WashingtonDC.hfc.comcastbusiness.net) |
| 22:54.28 | rocksfrow | how can i debug why a module isn't loading? |
| 22:54.38 | rocksfrow | i don't get any output what so ever when manually loading it via asterisk CLI |
| 22:54.47 | rocksfrow | any suggestions? |
| 22:55.33 | *** join/#asterisk Ziaeon (~Alchemist@adsl-074-166-171-132.sip.mia.bellsouth.net) |
| 22:56.39 | ChannelZ | you sure you have notices/errors turned on and the verbosity set up a little? |
| 22:57.19 | rocksfrow | asterisk -rvvvvvvvvvvvvvvvvvvvvv |
| 22:57.29 | rocksfrow | not sure if i have notices/errors turned on |
| 22:57.31 | rocksfrow | how can i check? |
| 22:58.32 | ChannelZ | logger show channels |
| 22:58.50 | rocksfrow | /var/log/asterisk/full File Enabled - Debug Verbose Warning Notice Error |
| 22:59.10 | ChannelZ | well that one should have everything then. 'Console' might not |
| 22:59.24 | ChannelZ | but if it aint on the disk log either... hmm |
| 22:59.37 | ChannelZ | Maybe the module is loading perfectly and just has nothing to say? :) |
| 22:59.58 | rocksfrow | ChannelZ, stop there |
| 22:59.59 | rocksfrow | found it |
| 23:00.04 | rocksfrow | thanks...i didnt look at the filesystem log |
| 23:00.07 | rocksfrow | i was talking about console |
| 23:00.08 | rocksfrow | sry |
| 23:00.26 | ChannelZ | yah 'notices' got turned off console logging at one point or another. You might want to turn it back on in logger.conf |
| 23:03.06 | *** join/#asterisk jks (jks@193.189.93.254) |
| 23:15.40 | *** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk) |
| 23:21.10 | *** join/#asterisk hfb (~hfb@pool-98-112-244-147.lsanca.dsl-w.verizon.net) |
| 23:26.48 | *** join/#asterisk knarfly (~vlad@rrcs-97-76-99-34.se.biz.rr.com) |
| 23:27.20 | *** join/#asterisk italorossi (~italoross@187.111.235.214) |
| 23:28.00 | knarfly | if a VOIP phone says it has six line appearance, does that mean I actually can have six separate accounts with my asterisk server and/or other VOIP providers? |
| 23:32.13 | Belgarath | yes |
| 23:34.46 | *** join/#asterisk neurosys (~neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
| 23:35.07 | stonezone | I have little to no exprience with asterix, but wanted to know if it's possible to create something like a party line with background music.... mayve using a a shoutcast stream or local mp3 playlist where multiple people dial in and can talk and listen to music at the same time? |
| 23:35.26 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 23:35.43 | *** join/#asterisk mcrownover (~markcrown@remote.gawest.com) |
| 23:35.59 | WIMPy | yes |
| 23:36.43 | stonezone | to both, shoutcast or local music files? |
| 23:36.57 | p3nguin | Asterisk can handle both. |
| 23:37.12 | stonezone | great....thanks! |
| 23:37.34 | p3nguin | You'll probably end up using MusicOnHold() and MeetMe() to achieve it. |
| 23:38.48 | stonezone | gotcha |
| 23:39.56 | TJNII | Remember to allow good codecs. |
| 23:40.05 | TJNII | 'Cus music + gsm = crap |
| 23:40.07 | p3nguin | and disallow all |
| 23:40.48 | p3nguin | Also remember that Asterisk has to transcode when calls are in a MeetMe conference. |
| 23:41.42 | ChannelZ | *cough* 900 bootycall hotline *cough* |
| 23:41.47 | stonezone | lol |
| 23:42.07 | TJNII | Barry White in the background... Awww yeah. |
| 23:45.01 | stonezone | mp3player() would do similar? |
| 23:48.01 | p3nguin | Yeah, but be careful not to let DTMF into your conference, or they callers could make it exit. |
| 23:48.37 | tzafrir_laptop | BTW: I coun't find the following on the 'net. I believe they would be required for some Asterisk setups: |
| 23:48.51 | tzafrir_laptop | http://tzafrir.org.il/~tzafrir/silly/ |
| 23:49.06 | tzafrir_laptop | (msg / mail me if it's not OK to post them) |
| 23:49.14 | stonezone | i've got a long way to go before anything is setup. thanks for all the input! |
| 23:50.46 | tzafrir_laptop | I guess some of those could be handy for usage in an IVR |
| 23:58.50 | *** join/#asterisk gavimobile (~user@bzq-84-108-29-62.cablep.bezeqint.net) |
| 23:58.58 | TJNII | tzafrir_laptop: Are you Karl? |
| 23:59.04 | tzafrir_laptop | No |
| 23:59.22 | TJNII | Mmmmm-hmmm. |