| 00:03.18 | pabelanger | bran: #freepbx |
| 00:07.47 | Godfather_ | anyone have paired his mobile using chan_mobile with bluez 4.X and asterisk 1.6? |
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| 00:30.55 | Micc | Is there any way to share channel group counts between servers? |
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| 00:40.10 | pabelanger | Micc: Database |
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| 00:40.56 | Micc | Thats what I was affraid of. That could be a lot of writes. |
| 00:41.59 | Micc | I guess I'll just have to write the group_count every time a call comes in or is placed. |
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| 01:05.08 | wokkad | any chan_sccp users around that can answer a quick question? |
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| 01:38.33 | afink | is there any way to test or tell the signal quality of a t1 without being local and using a loopback plug? |
| 01:38.54 | afink | I am using a TE122 w/dahdi |
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| 01:43.28 | WIMPy | Interesting question. In theory you could try a BERT against the echo application. I guess, I should try that some time... |
| 01:50.31 | drmessano | prefers to run a solid E.R.N.I.E. test |
| 01:52.20 | coppice | does that involve the exchange of cookie monsters? |
| 01:54.32 | afink | lol sesame street humor |
| 01:56.11 | afink | I am having users complain of intermittent dropped calls you guys know of anything I should check? |
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| 01:57.04 | WIMPy | Check your logs for disconnect reasons? |
| 01:58.43 | afink | Did that already |
| 02:00.11 | afink | actually I do see [Jun 27 01:17:12] NOTICE[3250] chan_dahdi.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 ............corresponding to some of the trouble reports |
| 02:01.45 | *** part/#asterisk nny (~Scott@cpe-071-076-058-253.sc.res.rr.com) |
| 02:08.14 | afink | could the card being on the same IRQ as 3 other devices be the problem? |
| 02:08.34 | Corydon76-dig | Depends upon the age of the card |
| 02:08.56 | Corydon76-dig | For current cards, that's not a possible problem |
| 02:09.38 | afink | I have a few missed interrupt messages in dmesg, I have seen much worse before |
| 02:09.46 | Corydon76-dig | If you're having troubles like that, it's best to call your vendor's support line and ask for assistance |
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| 02:30.17 | *** join/#asterisk EmleyMoor (phil@topdeck.tinsleyviaduct.com) |
| 02:30.42 | EmleyMoor | What ports are normally used for the audio of a call initiated by SIP? |
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| 02:33.41 | p3nguin | emleymoor: Check your devices and asterisk's rtp.conf. |
| 02:33.49 | EmleyMoor | I understand it's not standard... |
| 02:33.57 | p3nguin | usually udp 10000-20000 |
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| 02:50.16 | tripps | hello. I'm trying to debug why one extension isn't getting calls. What does ExtensionState of -1 mean in dialparties.agi? That's what this extension is returning. Successful ones are returning 0. |
| 02:53.39 | tripps | perhaps this is a freepbx Q. I'll ask there |
| 02:59.20 | ChannelZ | yes |
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| 04:04.12 | EmleyMoor | What amount of bandwidth is typically needed per phone call, a-law? |
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| 04:12.50 | Kyosh | [00:00] <EmleyMoor> What amount of bandwidth is typically needed per phone call, a-law? |
| 04:12.55 | Kyosh | umm |
| 04:12.58 | Kyosh | 64k |
| 04:13.03 | Kyosh | there's charts all over the web |
| 04:13.26 | p3nguin | kyosh: Your clock is off. |
| 04:13.45 | Kyosh | oh damn dont tell me that :( |
| 04:14.55 | Kyosh | how much off? |
| 04:14.56 | p3nguin | I thought you should know. |
| 04:15.09 | p3nguin | ~4 minutes |
| 04:15.10 | Kyosh | thinks he went to sleep yesturday and woke up next year :( |
| 04:15.24 | Kyosh | dude its this virtualization crap |
| 04:15.31 | Kyosh | cant sync to host |
| 04:15.38 | Kyosh | dont care either really :( |
| 04:15.48 | p3nguin | I have that problem with my openbsd virtual machine. |
| 04:16.04 | Kyosh | if i sync to host, my timestamps are all over the fukin place |
| 04:16.19 | p3nguin | What OS are you using? |
| 04:16.23 | Kyosh | if i dont sync to host, im not syncd to anything |
| 04:16.39 | Kyosh | host, ESXi. guest, XP pro |
| 04:17.08 | p3nguin | XP's internet time service won't keep it right? |
| 04:17.17 | Kyosh | i aint using that :p |
| 04:17.18 | Kyosh | hah |
| 04:17.46 | Kyosh | i dont own a watch. what makes you think i care about my workstations time? :-p |
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| 07:50.43 | athom | I use AsteriskNOW 1.7 with FreePBX 2.8. Before the Caller ID number that I want to be there is showing the digit "1". From what can be obtained this? |
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| 07:53.17 | ChannelZ | see #freepbx and/or #asterisknow |
| 07:54.53 | athom | ok |
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| 08:01.14 | DelphiWorld | hi |
| 08:01.20 | DelphiWorld | how do i show my registered peers? |
| 08:01.47 | DelphiWorld | in sip |
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| 08:04.09 | ChannelZ | sip show peers |
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| 08:09.11 | tzafrir_laptop | DelphiWorld, sip show subscriptions ? |
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| 08:13.42 | DelphiWorld | thank you tzafrir_laptop i will try it out |
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| 08:48.50 | joobie | hey guys.. having problems stripping out noise in a recording that i have .. managed to do it but it then warps the voice a little - anyone have experience with this sorta stuff? |
| 08:57.37 | *** join/#asterisk festr_ (~festr@nostromo.flh.cz) |
| 08:57.55 | festr_ | hi, anyone here using/tried chan_oss witch video support? |
| 08:58.20 | festr_ | I cant figure it out which version of ffmpeg libraries use |
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| 09:20.32 | DelphiWorld | tzafrir_laptop, please where i can my sip domain? |
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| 09:22.14 | *** join/#asterisk kami (~user@unaffiliated/kami-) |
| 09:22.17 | kami | hello |
| 09:22.44 | kami | I am an asterisk newbie who is trying to set up sip on asterisk 1.4. |
| 09:24.17 | *** join/#asterisk mpe (~mpe@gate.ipvision.dk) |
| 09:27.09 | tzafrir_laptop | kami, and? |
| 09:27.22 | kami | tzafrir_laptop: sorry, I was called on the phone :) |
| 09:27.38 | kami | I can call other sip users, but for any extension which is defined in extensions.conf, I get 'User not available' in my sip client. |
| 09:27.41 | tzafrir_laptop | hopes the phone is Asterisk |
| 09:28.08 | tzafrir_laptop | festr_, not sure what minimal version is needed |
| 09:28.19 | kami | has to admit that it was his mobile phone, but he'll promise to change that in future |
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| 09:35.09 | *** part/#asterisk DelphiWorld (~Delphi@41.103.48.6) |
| 09:40.18 | kami | I know that that extensions work in principle, because I also have a Digium Wildcard which works. |
| 09:40.59 | kami | The problem seems to be that I probably have made a mistake in configuring sip |
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| 09:53.34 | tzafrir_laptop | core set verbose 3 |
| 09:58.20 | kami | tzafrir_laptop: I call 6002@my.domain and see: Looking for 6002 in sipphones, and then 404 not found |
| 10:00.31 | *** join/#asterisk mathslinux (~user@74.86.0.138) |
| 10:00.39 | kami | dialplan show gives me: [ Context 'sipphones' created by 'pbx_config'] ... '6002' => 1. Answer() .... |
| 10:01.45 | *** join/#asterisk garymc (~chatzilla@host81-148-64-72.in-addr.btopenworld.com) |
| 10:04.16 | tzafrir_laptop | kami, btw: dialplan show 6002@sipphones |
| 10:05.51 | garymc | can i use asterisk for SMS for my business? |
| 10:06.21 | garymc | sending sms to customers or potential customers? |
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| 10:10.38 | kami | tzafrir_laptop: http://paste.lisp.org/+2EEJ |
| 10:10.44 | kami | it is there |
| 10:13.00 | tzafrir_laptop | And what do you see in the CLI when a call comes in there? |
| 10:13.37 | kami | tzafrir_laptop: the mentioned 404 not found, but I'll paste that too. Wait. |
| 10:14.55 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
| 10:43.53 | garymc | so nobody know about SMS texting through Asterisk or voip? |
| 10:49.39 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 10:49.42 | SiNGLer | garymc: contact your sms/mobile providers and ask them what they can offer for sms sending |
| 10:50.09 | garymc | ok, just thought i seen Asterisk could do it somewhere? |
| 10:50.15 | SiNGLer | probably it will be some http based service, but I thing it may be more convenient than using asterisk |
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| 11:06.39 | kami | garymc: how often do you want to send how many sms to how many people (just to get an idea of what you're trying to achieve) |
| 11:06.53 | garymc | say 300 a month |
| 11:07.11 | garymc | ive had a price of 6.5p per text |
| 11:07.17 | garymc | with a third party company |
| 11:08.07 | kami | garymc: how much would sending an sms over a normal SIM card cost? |
| 11:08.22 | kami | garymc: given an optimised plan |
| 11:08.50 | garymc | free normaly |
| 11:09.11 | kami | garymc: do you have a linux server where you could install a piece of software? |
| 11:09.16 | garymc | yes |
| 11:09.21 | kami | and attach a gsm modem to it? |
| 11:09.42 | garymc | hmmm possibly if i can atach the gsm modem via usb or ethernet |
| 11:09.47 | garymc | i could |
| 11:10.19 | kami | garymc: then buy a cheap gsm modem (there are some with USB), install http://kannel.org/ on the server and you're ready to go |
| 11:10.39 | kami | you can send sms by calling a URL on the server |
| 11:10.57 | garymc | how much does it cost to send sms ? |
| 11:11.00 | garymc | this way? |
| 11:11.28 | kami | garymc: the SIM card which is in the modem determines the costs |
| 11:11.42 | garymc | ahhh ok cool |
| 11:11.47 | kami | garymc: if you have a plan where sms are free, then they're free |
| 11:11.56 | garymc | nice |
| 11:12.07 | kami | garymc: it's just like putting the SIM in a mobile phone and sending sms |
| 11:12.28 | kami | garymc: no difference from the point of view of your mobile phone operator |
| 11:12.32 | garymc | right i get you. Is Kannel easy to install? |
| 11:15.09 | kami | garymc: 'easy' is of course debatable :) It consists of some background processes with file based config. With a setting as yours, you don't have to configure very much. Just the device of your modem (e.g. /dev/ttyUSB0), the MSISDN of the SIM card and the type of gsm modem. |
| 11:15.31 | garymc | right |
| 11:15.43 | garymc | ill look into it thanks for your help and info Kami |
| 11:16.47 | *** join/#asterisk guilhermebr (~Guilherme@200.103.96.98) |
| 11:17.04 | kami | garymc: you're welcome. |
| 11:19.03 | festr_ | searching for video phone which can put video fulscreen and is controlable via scripts or another ways for embeding it to video station. any ide? |
| 11:19.06 | festr_ | idea? |
| 11:22.39 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 11:24.15 | kami | tzafrir_laptop: I tried to reproduce the problem, but there is nothing in the logs. And even tcpdump shows nothing. I'm using ekiga for calling and it looks like the call is not even initiated, although ekiga reports 'user is not available'. It looks like I have to either use another client. |
| 11:24.26 | kami | tzafrir_laptop: thank you for your help, so far. |
| 11:24.58 | tzafrir_laptop | hmm... so it does not get to the syste? |
| 11:29.01 | DND | guys need help.. im having problem with my G729 codec. i have two servers connected via IAX. server a can call server B using G729 but when server B calls Server A, it switches to gsm (only g729&GSM are allowed codecs) |
| 11:29.34 | *** join/#asterisk Pimmetje (~Pimmetje@83.119.156.35) |
| 11:31.29 | SiNGLer | kami: are you sure tcpdump listened correct interface? about akiga, try dialing by specifing number@server_ip |
| 11:31.48 | SiNGLer | DND: does it work if you allow only g729? |
| 11:31.55 | DND | no |
| 11:32.00 | DND | i cant hear anything |
| 11:32.22 | DND | though it shows in cmd line im calling |
| 11:32.31 | DND | bot it doesnt show the codec used |
| 11:32.34 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
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| 11:32.58 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 11:33.07 | SiNGLer | how does "core show translations" show g729? |
| 11:33.49 | DND | from server A or B? the one having problems receiving G729 is A |
| 11:33.57 | SiNGLer | on both |
| 11:34.16 | SiNGLer | and "core show translation", not "core show translations" :) |
| 11:34.43 | festr_ | use tabs :) |
| 11:34.51 | DND | a chart of numbers most probably the time it takes to translate |
| 11:35.04 | DND | ulaw to G729 takes 6 |
| 11:35.54 | DND | that's for Server A |
| 11:36.01 | SiNGLer | on both servers time is displayed? |
| 11:36.05 | DND | server B ulaw to g729 takes 14 |
| 11:36.13 | DND | yes |
| 11:36.36 | DND | this is a digium g729a codec |
| 11:36.46 | kami | SiNGLer: I dial 6002@server_ip and yes, tcpdump is run on the correct interface |
| 11:37.15 | kami | SiNGLer: but, maybe ekiga uses the wrong interface (I'll try another softphone) |
| 11:37.22 | SiNGLer | kami, can you run tcpdump/wireshark on your PC? check if you see outgoing packets |
| 11:37.46 | SiNGLer | check ekigas preferences, there are some settings related to ip and such |
| 11:38.41 | DND | hmm do you think the problem is the codec itself? in the instruction there's a tool that do benchmark on CPU so that i will download the most compatible g729 codec for my machine |
| 11:38.43 | SiNGLer | DND: on both servers does peer information show g729? and are they ordered correctly? |
| 11:38.57 | kami | SiNGLer: thank you. will do. |
| 11:39.05 | DND | sip show peers? |
| 11:39.26 | SiNGLer | iax show peer peer_name |
| 11:39.37 | SiNGLer | if peer is iax |
| 11:40.25 | *** part/#asterisk nextime (~nextime@unaffiliated/nextime) |
| 11:43.03 | DND | ok done! i know now the problem |
| 11:43.28 | DND | the new version of freepbx already has a module for iax settings |
| 11:43.49 | *** join/#asterisk mpe (~mpe@94.127.49.1) |
| 11:43.51 | DND | and g729 is disabled. but i have it in the conf file |
| 11:44.20 | DND | i think it disregards the _custom.conf modification |
| 11:44.33 | DND | thanks a lot guys!! |
| 11:45.28 | DND | btw, what does iaxcompat and trunkrealloc does if i set it to yes? |
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| 12:12.19 | kami | SiNGLer: wireshark shows that sip messages are sent out, but they really don't appear on the server side, although the registration process is clearly visible in the tcpdump logs on the server. But, during sip registration, requests are sent out using all source IPs of my machine, whereas the SIP INVITE which goes out for a call only uses one of the IPs. |
| 12:14.16 | kami | hmm, OTOH: this seems quite reasonable. After finding out that the server can be contacted through one of the interfaces, that interface is used for further communication. |
| 12:15.06 | *** join/#asterisk freakynl (~freaky@unaffiliated/freakynl) |
| 12:15.13 | SiNGLer | k, forget about SIP for now. does UDP packets have correct source and destination? is server gw set correctly? |
| 12:15.20 | kami | Anyway. Thanks a lot SiNGLer and tzafrir_laptop, I guess I rather have an ekiga or network problem rather than an asterisk config problem (at least for the moment). I'll solve that one first, and bug you again, later. Thanks. |
| 12:15.24 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 12:15.43 | SiNGLer | np |
| 12:16.13 | kami | SiNGLer: I'll look into that and get back. No need to bother the whole channel with networking issues. |
| 12:16.21 | SiNGLer | k |
| 12:16.30 | kami | Bye |
| 12:16.31 | *** part/#asterisk kami (~user@unaffiliated/kami-) |
| 12:17.20 | tzafrir_laptop | kami, how can you tell it did not get to your server? tcpdump on it? |
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| 12:19.34 | SiNGLer | as I understand he used tcpdump on server |
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| 12:24.54 | festr_ | anyone using app_rtsp? |
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| 12:35.29 | evilbit | hi, I'm using the included extensions.conf as a starting point for dialing out with entries like [trunklocal] and [trunktollfree]. Is there a way I can setup the CID to be a specific number depending on which SIP client is attempting to dial out? |
| 12:35.53 | [TK]D-Fender | evilbit: Yes. Set it before you dial |
| 12:36.02 | *** join/#asterisk Alton35 (~alton@69.45.116.128) |
| 12:36.03 | [TK]D-Fender | evilbit: "core show function CALLERID" |
| 12:36.12 | evilbit | thx :-) |
| 12:36.21 | [TK]D-Fender | evilbit: Assuming they allow you to do so |
| 12:36.29 | evilbit | yeah, they do |
| 12:36.43 | evilbit | I just wasn't sure how to set it depending upon the client |
| 12:36.49 | [TK]D-Fender | evilbit: I highly recommend yuo trash every bit you are not expressly using in the sample config |
| 12:37.03 | [TK]D-Fender | evilbit: And trash all the extra unneeded comments as well |
| 12:37.07 | evilbit | oh, ok |
| 12:37.40 | evilbit | I have a extensions.conf from asterisk1.2 which seems to work but alot has changed so I was trying to keep it as "up to date" as possible |
| 12:38.21 | [TK]D-Fender | evilbit: Per client what I would do is add something like "SetVar=OUTCID=2125551234" to their SIP peer entry and set the CID to that as ${OUTCID} |
| 12:38.37 | [TK]D-Fender | evilbit: Many things have changed since 1.2 |
| 12:38.40 | evilbit | oh, nice! |
| 12:38.49 | evilbit | yeah, it's very confusing, lol |
| 12:39.13 | evilbit | most of the docs seem to still reference 1.2 |
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| 12:43.04 | [TK]D-Fender | evilbit: The basics are mostly the same. Just things like "|" not being a delimiter, certain apps permanently removed, SIP peer handling, etc |
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| 12:46.53 | *** join/#asterisk ZeXr0 (~ZeXr0@ip216-239-95-218.vif.net) |
| 12:48.40 | ZeXr0 | Hi, a client report a very small delay between the number of ring and the redirection in the voicemail, when an Agent transfer a call. They say it ring only once and then it's transferred to the voicemail. Where should I look to change the delay ? They'd like the phone to ring at least 3 times before transfering to the voicemail |
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| 12:54.37 | ndemir | hello. my provider says that i should configure my ISDN card without crc check. How to do this in dahdi configuration |
| 12:55.02 | BugKhaM | the trunk r4692+ for zaptel 1.4 branch is able to compile under kernel 2.6.28+ |
| 12:55.58 | BugKhaM | but not for zaptel 1.2, anyone knows what changes I need to make to zaptel source inorder to have it compiled under new kernels |
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| 13:00.32 | [TK]D-Fender | ZeXr0: What is this "agent"? What "voicemail"? |
| 13:00.44 | [TK]D-Fender | ZeXr0: Got an actual call for us to look at? |
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| 13:09.43 | tzafrir_laptop | ndemir's question, for ther record: remove ',crc4' from the span line in system.conf (assuming this is E1) |
| 13:10.03 | tzafrir_laptop | BugKhaM, backporting those fixes should hopefully be simple |
| 13:10.19 | tzafrir_laptop | If you actually test them, I won't mind commiting them |
| 13:10.55 | BugKhaM | tzafrir_laptop, sure I will test it |
| 13:11.09 | BugKhaM | tzafrir_laptop, I actually need it badly |
| 13:12.19 | BugKhaM | tzafrir_laptop, T.30 works perfectly with zaptel 1.2 but not 1.4+ |
| 13:13.26 | tzafrir_laptop | What specific kernel? |
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| 13:15.42 | *** join/#asterisk rare1980_ (~as@109.169.28.42) |
| 13:15.50 | BugKhaM | tzafrir_laptop, it's 2.6.33.5-124.fc13.x86_64 |
| 13:16.01 | ZeXr0 | [TK]D-Fender : Sorry, I found what I was looking for. The email I received was not very clear. It's not while a call transfer, it was the delay between the Queue command and the VoiceMail command. |
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| 13:17.01 | BugKhaM | tzafrir_laptop, but if you need me to test it now I'm running 2.6.32.14-127.fc12.x86_64 |
| 13:17.36 | tzafrir_laptop | hmm... what specific hardware? |
| 13:18.00 | tzafrir_laptop | BugKhaM, frankly, we'd be more interested to know how it works with latest dahdi |
| 13:19.12 | BugKhaM | tzafrir_laptop, one TDM400P and one tormenta II |
| 13:19.34 | rare1980_ | hi all |
| 13:20.16 | BugKhaM | tzafrir_laptop, hmm it's working with zaptel branch 1.4 4692+ |
| 13:20.26 | [TK]D-Fender | ZeXr0: And you aren't any clearer about how these 2 apps are called relative to each other. |
| 13:20.41 | rare1980_ | i have configure this inbound number on my asterisk ... but i m getting issues.... |
| 13:20.45 | rare1980_ | exten => 01916700186,1,Answer() |
| 13:20.46 | rare1980_ | exten => 01916700186,n,Background(demo-instruct) |
| 13:20.46 | rare1980_ | exten => 01916700186,n,Hangup() |
| 13:20.56 | rare1980_ | chan_sip.c:14035 handle_request_invite: Call from '' to extension '01916700186' rejected because extension not found. |
| 13:21.05 | rare1980_ | can't under stand wht i m missing |
| 13:21.06 | BugKhaM | tzafrir_laptop, should that work for latest dahdi too? |
| 13:21.13 | [TK]D-Fender | rare1980_: Do NOT flood in here |
| 13:21.20 | rare1980_ | sorry |
| 13:21.24 | [TK]D-Fender | rare1980_: it isn't in the CONTEXT the call is looking in <--- |
| 13:21.38 | ZeXr0 | [TK]D-Fender : Well as I said, I found the problem, and solve it. Sorry if that wasn't clear :S (It was only a setting TimeOut for the Queue command somewhere in my script) |
| 13:22.09 | [TK]D-Fender | ZeXr0: Glad you've fixed it |
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| 13:23.06 | *** part/#asterisk jtrimmer (~jtrimmer@75-151-66-133-WestFlorida.hfc.comcastbusiness.net) |
| 13:23.24 | rare1980_ | let me check |
| 13:24.52 | BugKhaM | tzafrir_laptop, I can test the dahdi packages too if you want |
| 13:25.32 | tzafrir_laptop | BugKhaM, why shouldn't it work with latest DAHDI? |
| 13:35.18 | BugKhaM | tzafrir_laptop, yeah I just compiled latest trunk version of dahdi-kernel on 2.6.32.14-127.fc12.x86_64 without any problem |
| 13:36.14 | BugKhaM | tzafrir_laptop, so only zaptel 1.2 has problems compiling under new kernels |
| 13:37.06 | tzafrir_laptop | yes |
| 13:37.13 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.112.66.dsl.dyn.forthnet.gr) |
| 13:39.13 | BugKhaM | tzafrir_laptop, i don't remember how I can trace the differences between zaptel r.4692 and prior releases off the website |
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| 13:39.59 | tzafrir_laptop | svn diff ? |
| 13:40.09 | tzafrir_laptop | Do you have a local svn checkout? |
| 13:40.29 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 13:41.14 | tzafrir_laptop | svn diff http://svn.asterisk.org/svn/zaptel/tags/1.4.12 http://svn.asterisk.org/svn/zaptel/branches/1.4 |
| 13:41.21 | BugKhaM | tzafrir_laptop, using svn command line? |
| 13:41.38 | tzafrir_laptop | but 'svn log' would give you a better start |
| 13:42.12 | tzafrir_laptop | If you prefer http://svnview.digium.com/svn/zaptel/branches/1.4/ , use it |
| 13:42.38 | BugKhaM | tzafrir_laptop, ohh that's the one |
| 13:43.09 | tzafrir_laptop | http://svnview.digium.com/svn/zaptel/branches/1.4/?view=log |
| 13:43.29 | BugKhaM | tzafrir_laptop, by the way, I'll also try from svn command line |
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| 13:58.39 | mintee | I've been looking, and cannot find a good reason why mysql cdr logging isn't in the default * install. |
| 13:58.51 | mintee | there must be a reason... |
| 13:58.53 | ndemir | i have this: "chan_dahdi.c: Unable to determine channel for data." how to solve this? |
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| 14:00.21 | SiNGLer | mintee: afaik it's licensing |
| 14:00.29 | pabelanger | mintee: Licensing |
| 14:00.37 | mintee | ah, ok, that makes sense... |
| 14:00.49 | mintee | it was concerning me that maybe there was some other issue |
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| 14:01.13 | mintee | i've been using it stable for 2 years, so I donno why I just got concerned out of no where. :P |
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| 14:04.35 | [TK]D-Fender | mintAnd because * can do just fine on its own without extra dependencies for those not needing to administer a DB server as well |
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| 14:05.05 | jtrimmer | does h, always get called twice when you try to send a fax? It seems to call it after P30EVN_SEND_STARTED and then again after P30EVN_COMPLETE |
| 14:06.03 | jtrimmer | sorry my mistake it calls it after auto fallthrough |
| 14:06.17 | jtrimmer | not after the P30EVN_SEND_STARTED |
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| 14:13.09 | defswork | what do BT call disconnect supervision ? |
| 14:15.28 | defswork | Disconnect Clear Time apparently |
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| 14:35.28 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 14:44.26 | structz | Hi i'm having a problem when seding a fax using a D-LINK ata i get this message on asterisk CLI and my asterisk justs stops working - channel.c:1044 __ast_queue_frame: Exceptionally long voice queue length queuing to IAX2/mediagateway-1755 |
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| 14:47.20 | structz | btw i'm using asteriks 1.6.2.9 on debian |
| 14:50.32 | WIMPy | Is someone here using 1&1? I have the problem, that incomming calls get terminated by them immediately after answer. Outgoing works fine. |
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| 14:54.04 | tzafrir_laptop | structz, the IAX is iaxmodem? |
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| 15:04.41 | structz | tzafrir_laptop, no.. de iax account is used to connected to the mediagateway(asterisk 1.6.2.6 with E1 devices) |
| 15:07.48 | raceme | i'm trying to configure a digium TDM410 on my debian squueze and i've got some problem... the wctdm24xxp module is loaded; when I launch dahdi_genconf it says "Empty configuration -- no spans" and generate empty /etc/dahdi/system.conf (except from loadzone and defaultzone) |
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| 15:25.47 | tzafrir_laptop | raceme, what's the output of: dahdi_hardware; lsdahdi |
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| 15:27.35 | raceme | tzafrir_laptop: i'm rebooting the host right now... I tell you once it is up |
| 15:27.52 | *** part/#asterisk jtrimmer (~jtrimmer@75-151-66-133-WestFlorida.hfc.comcastbusiness.net) |
| 15:32.10 | raceme | tzafrir_laptop: http://pastebin.com/XJ3JMm2w |
| 15:32.35 | raceme | tzafrir_laptop: i think it is better now after the reboot; i did not have the span details begore |
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| 15:36.47 | emora | I hava a couple of machines running Asterisk 1.6.1.12. Their consoles are constantly showing messages saying "doing dnsmgr_lookup for...". Any idea what we can do to get this to stop? |
| 15:37.34 | ChannelZ | It is because you are using hostnames for SIP peers etc. and you have DNS manager turned on |
| 15:37.54 | ChannelZ | You can turn it off with the dnsmgr.conf |
| 15:38.03 | raceme | Should I worry about that message: "wctdm24xxp 0000:04:08.0: Host failed to service card interrupt within 128 ms which is a hardunderun." ? |
| 15:38.16 | emora | Thanks ChannelZ |
| 15:38.20 | ChannelZ | if it's happening a lot, yes |
| 15:39.01 | raceme | ChannelZ: once a boot just at module loading but i don't use the card as it is not yet configured |
| 15:39.20 | emora | ChannelZ: What functions would we loose by turning off DNS manager? |
| 15:40.23 | ChannelZ | emora: well Asterisk won't notice an IP change while it's running |
| 15:41.09 | emora | I see |
| 15:41.18 | ChannelZ | emora: That is if you use "outgoing.phone.com" for your ITSP and they have 5 servers with different IPs, your system will pick up whichever one it gets initially and then never refresh it. If thta particular server goes down, it will keep trying to use the same IP until you reload, etc. |
| 15:41.30 | *** join/#asterisk Benwa (~Benwa@ip-62-235-215-176.dsl.scarlet.be) |
| 15:41.40 | emora | I understand |
| 15:41.41 | ChannelZ | DNS manager polls every so often and updates the IP for the peer if it changes |
| 15:41.58 | ChannelZ | I have mine set to 4 hours |
| 15:42.34 | raceme | perhaps it is a silly question but is it normal to have fxsks lines in dahdi/system.conf for my fxO modules ? |
| 15:42.38 | emora | I'm surprised that the dnsmgr.conf has enable=no |
| 15:42.57 | ChannelZ | raceme: FXO ports use FXS signalling and vice-versa |
| 15:43.17 | raceme | ChannelZ: ok so this is normal |
| 15:43.22 | structz | Hi i'm having a problem when seding a fax using a D-LINK ata i get this message on asterisk CLI and my asterisk justs stops working - channel.c:1044 __ast_queue_frame: Exceptionally long voice queue length queuing to IAX2/mediagateway-1755 (asterisk 1.6.2.9 on debian) |
| 15:43.29 | ChannelZ | raceme: yes |
| 15:43.59 | ChannelZ | runs off to work |
| 15:44.08 | raceme | thanks ChannelZ and tzafrir_laptop |
| 15:44.42 | *** join/#asterisk anonymouz666 (~anonymouz@189.24.7.73) |
| 15:47.15 | drmessano | russellb: "Let's hope I actually know how Asterisk works." <--- Feel free to post your questions in here. Thx. |
| 15:47.24 | russellb | :-) |
| 15:47.41 | russellb | could you explain the Asterisk threading model for me? |
| 15:48.21 | *** join/#asterisk coppice (~chatzilla@19.176.64.202.dyn.pacific.net.hk) |
| 15:48.41 | drmessano | Have you ever used a quesadilla maker? |
| 15:48.54 | drmessano | That is what Alton Brown calls a "unitasker" |
| 15:49.02 | drmessano | Asterisk = Not a unitasker |
| 15:49.07 | drmessano | Next? |
| 15:49.19 | emora | ChannelZ: CLI dnsmngr status says "DNS Manager: disabled" |
| 15:49.39 | emora | and on the next line "doing dnsmgr_lookup for..." |
| 15:49.53 | emora | well, almost the next line. ;) |
| 15:50.29 | drmessano | russellb: Sorry, I just finished up writing "Asterisk: The Future Of Telephony - Middle Management/Sales Edition" |
| 15:50.53 | russellb | awesome |
| 15:51.05 | tzafrir_laptop | structz, can you try to test each link of the chain separately? |
| 15:51.34 | structz | what you mean!? |
| 15:51.35 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 15:51.39 | drmessano | Did you know the Asterisk engine is optimized for both inbound AND outbound calls? |
| 15:51.51 | structz | the message only gets on one of the servers |
| 15:51.52 | drmessano | Chapter 4 |
| 15:52.00 | structz | the other one(media gateway) works just fine |
| 15:52.45 | Qwell | drmessano: quesadilla maker can make lots of things |
| 15:52.51 | Qwell | like really thin sandwiches |
| 15:53.34 | *** join/#asterisk BugKhaM (~BugKhaM@180.180.102.84) |
| 15:55.09 | drmessano | Qwell: Asterisk can make really thin sandwiches. It also has a small carbon footprint, is made from 100% post-consumer content, and leverages emerging technologies. |
| 15:55.31 | Qwell | also, UC. |
| 15:55.55 | *** join/#asterisk Entulho (~foo@189.72.127.218) |
| 15:56.12 | drmessano | UC, FMC, and IPv6 |
| 15:56.18 | Qwell | FMC? |
| 15:56.26 | drmessano | Fixed Mobile Convergence |
| 15:56.29 | Qwell | of course |
| 15:56.52 | Qwell | IPv6 is actually happening real soon now |
| 15:56.58 | coppice | as opposed to Broken Mobile Convergence? |
| 15:57.17 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 15:57.31 | drmessano | Asterisk also supports the entire IPv4 address space |
| 15:58.28 | Qwell | drmessano: it doesn't support the 169.254.0.0/16 block |
| 15:58.43 | drmessano | Asterisk also supports the entire IPv4 address space* |
| 15:58.58 | drmessano | better? |
| 16:00.10 | rare1980_ | i am trying to recive inbound call to any extension but its not working fine for me... all calls are going to only one extension. ........ here is pastebin link.... please any help on this???? http://pastebin.com/Y213LP5f |
| 16:00.11 | *** join/#asterisk pls_wrk_fs (~Miranda@119.94.193.194) |
| 16:00.54 | pls_wrk_fs | ola.does IVR have a number where i can route my call into? |
| 16:01.01 | *** join/#asterisk mifadir (29891ea3@gateway/web/freenode/ip.41.137.30.163) |
| 16:01.34 | [TK]D-Fender | pls_wrk_fs: Huh? |
| 16:02.12 | leifmadsen | LUNCH! |
| 16:02.17 | Qwell | 's' is for 'secret' |
| 16:02.23 | [TK]D-Fender | rare1980_: Where is the CALL for us to look at? |
| 16:03.23 | pls_wrk_fs | i have a switch.. all of the DID calls goes thru that switch. now i want that switch to forward it to that specific asterisk install. i got it working but for a specific extension only.i am just wondering if there is a number for a specific IVR that has been made earlier |
| 16:03.37 | pls_wrk_fs | i only have freepbx access sorry |
| 16:03.42 | Qwell | ~freepbx |
| 16:03.43 | 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 |
| 16:03.43 | rare1980_ | [TK]D-Fender: wht do u mean by? |
| 16:04.25 | [TK]D-Fender | rare1980_: You are showing us configs. This doesn't explain what is wrong with your CALL. Look. At. The. CALL <--------------- |
| 16:06.25 | rare1980_ | [TK]D-Fender: there is nothing wrong but this config file shows that calls is only going to only 1 softphone... i need wht ever calls comes in it will go to all softphone.. |
| 16:06.49 | *** join/#asterisk Z_God (~julius@wlan226192.mobiel.utwente.nl) |
| 16:11.05 | *** join/#asterisk bran (~chatzilla@unaffiliated/bran) |
| 16:11.33 | bran | should the skype for asterisk module show up as a channel in FreePBX? |
| 16:12.13 | drmessano | no |
| 16:13.17 | *** part/#asterisk mifadir (29891ea3@gateway/web/freenode/ip.41.137.30.163) |
| 16:13.21 | bran | how can I configure it as a normal outgoing route then :( ? |
| 16:14.38 | rare1980_ | any help on this pleasE? |
| 16:14.41 | pabelanger | bran: #freepbx |
| 16:15.14 | pabelanger | ~collectdebug |
| 16:15.15 | 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 |
| 16:15.18 | pabelanger | rare1980_: ^^ |
| 16:15.36 | *** join/#asterisk jameswf (~james@unaffiliated/jameswf-home) |
| 16:16.48 | *** part/#asterisk pls_wrk_fs (~Miranda@119.94.193.194) |
| 16:17.47 | *** join/#asterisk DelphiWorld (~Delphi@41.200.23.121) |
| 16:17.49 | DelphiWorld | hi hi |
| 16:18.04 | DelphiWorld | doe asterisk support IpV6? |
| 16:18.23 | Corydon76-dig | Not yet, but the work is nearly ready |
| 16:18.44 | anonymouz666 | IPV6, async DNS, children processes and transaction support in chan_sip. |
| 16:18.55 | Corydon76-dig | https://reviewboard.asterisk.org/r/743/ |
| 16:20.23 | DelphiWorld | Corydon76-dig: thanks |
| 16:20.34 | DelphiWorld | Corydon76-dig: dig mean? |
| 16:20.35 | DelphiWorld | :P |
| 16:20.48 | Corydon76-dig | So, yes, 1.8 will have IPv6 |
| 16:20.56 | Corydon76-dig | Short for Digium |
| 16:21.18 | anonymouz666 | Corydon76-dig: what about lap? |
| 16:21.40 | Corydon76-dig | Laptop |
| 16:21.50 | DelphiWorld | Corydon76-dig: no digite? |
| 16:21.51 | DelphiWorld | hehe |
| 16:22.25 | Corydon76-dig | Huh? |
| 16:22.51 | Corydon76-dig | Must be lost in translation |
| 16:22.55 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 16:22.59 | anonymouz666 | sends a T.38 fax to coppice |
| 16:23.28 | DelphiWorld | coppice: :P |
| 16:24.04 | *** part/#asterisk DelphiWorld (~Delphi@41.200.23.121) |
| 16:30.49 | *** join/#asterisk t_dot_zilla (~chatzilla@firewall-a.buf.ny.i-evolve.net) |
| 16:31.18 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.157) |
| 16:32.19 | [TK]D-Fender | rare1980_: You aren't dialing multiple devices in there at all |
| 16:32.26 | [TK]D-Fender | rare1980_: Only a single one. |
| 16:32.43 | [TK]D-Fender | rare1980_: and a SPECIFIC one at that |
| 16:39.32 | *** join/#asterisk RobH (~robh@wikimedia/RobH) |
| 16:39.32 | *** join/#asterisk Mango (~iMango@d154-20-89-230.bchsia.telus.net) |
| 16:40.16 | *** join/#asterisk grinder13 (~grinder@146.176.165.57) |
| 16:40.47 | *** join/#asterisk Godfather_ (~Godfather@193.153.129.150) |
| 16:40.57 | *** join/#asterisk mpe (~mpe@gate.ipvision.dk) |
| 16:42.19 | Godfather_ | o/ |
| 16:44.54 | anonymouz666 | the godfather? |
| 16:45.16 | *** join/#asterisk Faithful (~Faithful@202.6.145.116) |
| 16:45.30 | WIMPy | But no Soul. |
| 16:46.07 | Godfather_ | Yes, Michael Corleone. |
| 16:46.21 | *** join/#asterisk aidinb (~Aidin@wsip-98-190-45-225.oc.oc.cox.net) |
| 16:46.48 | *** join/#asterisk netmax (~netmax@is.linux-administrator.com) |
| 16:46.52 | WIMPy | Doing new MOH files? |
| 16:48.10 | Godfather_ | WIMPy, my moh audio is "Shakira - waka waka (This time for africa)" |
| 16:48.16 | Godfather_ | :-) |
| 16:52.00 | grinder13 | hello! having problem with my IAX channel. i cannot register my IAX extensions. checked with ngrep and the IAX clients send normally the registration messages to the Asterisk server. SIP channel works as expected. any ideas what to check? |
| 16:52.54 | structz | Hi i'm having a problem when seding a fax using a D-LINK ata i get this message on asterisk CLI and my asterisk justs stops working - channel.c:1044 __ast_queue_frame: Exceptionally long voice queue length queuing to IAX2/mediagateway-1755 (asterisk 1.6.2.9 on debian) |
| 16:55.34 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 16:56.37 | *** join/#asterisk Faithful (~Faithful@202.6.145.116) |
| 17:03.50 | *** join/#asterisk boch (~boch@200.61.191.9) |
| 17:10.14 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 17:12.05 | *** join/#asterisk rare1980_ (~as@109.169.28.42) |
| 17:13.13 | *** join/#asterisk oldhack (~jfincher@cpe-24-27-56-221.austin.res.rr.com) |
| 17:13.15 | grinder13 | anyone for this? having problem with my IAX channel. i cannot register my IAX extensions. checked with ngrep and the IAX clients send normally the registration messages to the Asterisk server. SIP channel works as expected. any ideas what to check? |
| 17:13.35 | [TK]D-Fender | IAX debug & your firewall <- |
| 17:13.42 | structz | Hi i'm having a problem when seding a fax using a D-LINK ata i get this message on asterisk CLI and my asterisk justs stops working - channel.c:1044 __ast_queue_frame: Exceptionally long voice queue length queuing to IAX2/mediagateway-1755 (asterisk 1.6.2.9 on debian) |
| 17:15.25 | Mango | I want my Asterisk server to accept calls for only two numbers. What is the correct way to reject calls to other numbers? |
| 17:16.12 | tzafrir_laptop | structz, can you test a call from the PSTN to ast 1.6.2.9 separately? e..g to an echo test? |
| 17:16.27 | evilbit | in sip.conf I have defined [user] SetVar=OUTCID=2155551212 is there a way to check that this is being set within a context when the sip user calls a outside number? |
| 17:16.44 | tzafrir_laptop | Test different parts of the call path to see where the breakage is |
| 17:18.29 | grinder13 | Fender, I don't have a firewall. i am checking the IAX debug messages now |
| 17:18.40 | [TK]D-Fender | evilbit: I showed you that yesterday |
| 17:18.54 | [TK]D-Fender | evilbit: You reference it just like normal. ${OUTCID} |
| 17:19.06 | evilbit | no, you showed me today (i believe) but it doesn't seem to be setting it :-( |
| 17:19.20 | *** join/#asterisk ruben23 (~ITadmin@125.212.40.2) |
| 17:19.21 | [TK]D-Fender | evilbit: So use that as the value to set the callier ID as using the function I ALSO pointed you to the instructions for. |
| 17:19.57 | [TK]D-Fender | evilbit: I didn't say that would set the caller ID. Isaid that would let you set the value within the PEER. You would then REFERENCE that var when SETTING it jsut before yuo DIAL <- |
| 17:20.03 | [TK]D-Fender | evilbit: Please pay attention |
| 17:20.32 | structz | tzafrir_laptop, help me out, whats the point of that test? because the message occurs only in one asterisk(the one the ata is registered to) the other one is the mediagateway is working just fine |
| 17:21.14 | evilbit | [TK]D-Fender: perhaps there's a difference btwn paying attention and understanding ;-) I think that latter is my problem |
| 17:23.21 | *** join/#asterisk Godfather_ (~Godfather@79.109.251.13.dyn.user.ono.com) |
| 17:24.31 | *** join/#asterisk nextime (~nextime@unaffiliated/nextime) |
| 17:25.17 | evilbit | for my sip.conf I have [user] ... SetVar=OUTCID=2155551212 ... and in extensions.conf I have: [internal] ... include => local and [local] include => trunklocal ... and [trunklocal] exten => s,1,Set(CALLERID(number)=${OUTCID}) |
| 17:25.26 | tzafrir_laptop | evilbit, core show application DumpChan |
| 17:26.09 | tzafrir_laptop | or simply: NoOp(OUTCID=${OUTCID}) |
| 17:26.13 | nextime | hello all. I have what it seem to be a common problem looking at what google say, but i can't find a working solution. Two did from the same provider, i get the username mismatch error if i try to call the second did, and if i set insecure=invite,port all calls are redirected to the context of the first one |
| 17:26.25 | nextime | is there any reliable solution to this issue? |
| 17:30.04 | *** join/#asterisk rare1980_ (~as@109.169.28.42) |
| 17:30.12 | [TK]D-Fender | evilbit: Show us the failed CALL |
| 17:30.22 | [TK]D-Fender | evilbit: PASTEBIN is your friend |
| 17:30.24 | [TK]D-Fender | ~pb |
| 17:30.24 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 17:30.25 | [TK]D-Fender | ^^^^^^^6 |
| 17:30.47 | evilbit | ok, sorry... let me get it together and pastebin... you want what the console shows? |
| 17:32.47 | [TK]D-Fender | evilbit: Clearly. The ENTIRE call and the relevent bits of extensions.conf and sip.conf |
| 17:33.40 | *** join/#asterisk devdvd (~myemail@173-31-175-216.client.mchsi.com) |
| 17:34.06 | evilbit | [TK]D-Fender: Ok, bear with me please... getting through all of this is a little overwhelming... it's quite a large learning curve :-) |
| 17:34.24 | *** join/#asterisk guilhermebr (~Guilherme@ns2.aser.com.br) |
| 17:35.22 | devdvd | hey all. Im looking to purchase 70 polycom 321 and the power supplies. Any recommendations on where to order them? The only vendor that is out is voipsupply |
| 17:35.58 | *** join/#asterisk RobH (~robh@wikimedia/RobH) |
| 17:36.25 | [TK]D-Fender | devdvd: I highly recommend getting PoE switches instead |
| 17:36.41 | devdvd | TK, im gonna push for that |
| 17:36.43 | *** join/#asterisk Faithful (~Faithful@202.6.145.116) |
| 17:36.48 | devdvd | just needed to get pricing on all options |
| 17:36.58 | beek | devdvd: www.e5strategies.com or 888voipstore.com |
| 17:37.17 | beek | s/5/4/ |
| 17:37.24 | *** join/#asterisk freeedrich| (~eeePC@hansaserver.de) |
| 17:38.46 | [TK]D-Fender | devdvd: PS = $17 ~ x 24 = 408$. Yuo can get a PoE switch for that easily enough and NOT haev to take up space on switches you have now. |
| 17:40.03 | [TK]D-Fender | devdvd: So 3 switches and a lot of peace of mind. |
| 17:40.14 | [TK]D-Fender | devdvd: Also allows you central power backup. |
| 17:41.25 | evilbit | ok, here we go: http://pastebin.com/Pf8NsQUz |
| 17:42.18 | evilbit | doh |
| 17:42.35 | *** join/#asterisk Pan3D (~Pan3D@63.208.160.190) |
| 17:42.55 | [TK]D-Fender | FAIL |
| 17:43.25 | evilbit | http://pastebin.com/Cecs8VT3 |
| 17:43.34 | evilbit | yes, FAIL... :-( |
| 17:43.45 | evilbit | but at least a noticed fail ;-) |
| 17:45.52 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 17:46.17 | [TK]D-Fender | evilbit: Yeah.. so where is a GOOD link for us? |
| 17:46.40 | evilbit | http://pastebin.com/Cecs8VT3 |
| 17:46.45 | evilbit | ^ |
| 17:47.27 | *** join/#asterisk aidinb (~Aidin@wsip-98-190-45-225.oc.oc.cox.net) |
| 17:50.05 | *** join/#asterisk SiNGLer (~singler@88.118.8.230) |
| 17:51.40 | [TK]D-Fender | evilbit: exten => s,1,Noop(CALLERID(number)=${OUTCID}) <---- what good does it do having this in "s"? You need this in the PATTERN that gets CALLED |
| 17:52.33 | [TK]D-Fender | evilbit: and no, "s" is not magical, global, automatic, hydromatic, or GREASE LIGHTNING |
| 17:54.14 | evilbit | ok, but the second one should match, no? |
| 17:54.57 | [TK]D-Fender | evilbit: ... -- Executing [912155551212@internal:1] Dial("SIP/evilbit-00000001", "IAX2/user@teliax/12155551212") in new stack |
| 17:55.07 | [TK]D-Fender | evilbit: Look at the LENGTH of that number being matched |
| 17:55.25 | [TK]D-Fender | evilbit: exten => _9NXXXXXX,1,Noop(CALLERID(number)=${OUTCID}) <- Look at the LENGTH of this PATTERN |
| 17:55.38 | [TK]D-Fender | evilbit: And then try any tell me they aren't apples & oranges... |
| 17:55.51 | [TK]D-Fender | :) |
| 17:55.58 | evilbit | thanks :-) |
| 17:56.00 | *** join/#asterisk neurosys (~neurosys@173-9-159-177-miami.txt.hfc.comcastbusiness.net) |
| 17:56.19 | p3nguin | And for crying out sakes, stop using 9 on the front of your phone numbers. |
| 17:56.27 | *** join/#asterisk CoffeeIV (~rgr@dsl093-217-226.aus1.dsl.speakeasy.net) |
| 17:56.48 | evilbit | I'm open to suggestions... I'm just starting off with what the examples are! |
| 17:57.10 | [TK]D-Fender | p3nguin: HE'S STILL PRE-OCCUPIED WITH ..... NINETEEN ... NINETEEN EIGHTY FIVE! |
| 17:57.10 | p3nguin | There's your first suggestion -- get rid of the friggin' 9. |
| 17:57.12 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 17:57.17 | [TK]D-Fender | rocks out |
| 17:58.02 | evilbit | lolz... you guys gotta take it a bit easier on us newbies... there's so much conflicting examples out there it's a wonder anything non-trivial works at all |
| 17:58.31 | p3nguin | Read. Comprehend. Then take all the examples with a grain of salt. |
| 17:58.58 | evilbit | trust me, I'm trying |
| 17:58.59 | p3nguin | Once you comprehend what is going on, you'll know what information is worthless and what can be helpful. |
| 17:59.07 | p3nguin | Did you read The Book yet? |
| 17:59.17 | evilbit | the book is 1.2, no? |
| 17:59.19 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 17:59.21 | p3nguin | ~book |
| 17:59.22 | 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 |
| 17:59.36 | p3nguin | Plenty of good theory in there. |
| 17:59.48 | structz | Hi i'm having a problem when seding a fax using a D-LINK ata i get this message on asterisk CLI and my asterisk justs stops working - channel.c:1044 __ast_queue_frame: Exceptionally long voice queue length queuing to IAX2/mediagateway-1755 (asterisk 1.6.2.9 on debian) |
| 18:00.07 | evilbit | right, but that's the point... when you go from theory to example is where the problems occur ;-) |
| 18:00.20 | evilbit | but I got you :-) |
| 18:00.25 | p3nguin | You skipped step 2 (comprehend). |
| 18:00.31 | evilbit | haha, ok |
| 18:00.43 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 18:01.07 | [TK]D-Fender | evilbit: And well.. this was all nice & neat in the book. I didn't even swear or anything :) |
| 18:01.57 | [TK]D-Fender | evilbit: and the Book is quite fine for learning the dialplan. Only minor little differences you haven't run into yet |
| 18:02.02 | evilbit | hehe, the book is good |
| 18:02.11 | [TK]D-Fender | evilbit: Well.. as good as we've got. |
| 18:02.31 | [TK]D-Fender | considers doing some mroe substantial writeups on his blog |
| 18:02.50 | evilbit | I wasn't trying to insult anyone... I was just trying to say that the step from newbie to guru is pretty large and once you're familiar with a app it's easy to forgot how difficult the learning curve can be |
| 18:03.30 | evilbit | is in the process of removing all _9s ;-) |
| 18:03.59 | p3nguin | You'll want to keep the _ though. |
| 18:04.06 | p3nguin | _ indicates a pattern. |
| 18:04.12 | evilbit | ah, right... gotcha |
| 18:04.15 | tzafrir_laptop | structz, do you get the same from a voice call in the same path? |
| 18:04.23 | structz | nope |
| 18:04.28 | structz | only when using fax thru ata |
| 18:04.36 | structz | tzafrir_laptop, ! |
| 18:04.39 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 18:04.40 | p3nguin | _NXXNXXXXXX for 10-digit dialing |
| 18:04.46 | p3nguin | _1NXXNXXXXXX for 11-digit dialing |
| 18:04.58 | p3nguin | _NXXXXXX for 7-digit dialing |
| 18:05.15 | evilbit | ok, so now I have: exten => _9NXXXXXXXXXX,1,Noop(CALLERID(number)=${OUTCID}) |
| 18:05.18 | tzafrir_laptop | structz, fax is sent as simple audio or as T.38? |
| 18:05.23 | p3nguin | kill the 9. |
| 18:05.30 | p3nguin | And stop dialing with 9s. |
| 18:05.34 | structz | tzafrir_laptop, ata is configured to use T.30 only |
| 18:05.36 | evilbit | I'll remove that 9 in a minute... but it still doesn't match :-( for 912155551212 |
| 18:05.49 | p3nguin | of course not. |
| 18:06.01 | p3nguin | If you'll read the three lines I gave you and use them, it will work. |
| 18:06.11 | evilbit | ok, I will |
| 18:06.17 | coppice | an ATA using T.30 is an interesting concept :-\ |
| 18:09.24 | evilbit | [Jun 29 14:08:33] WARNING[26230]: chan_iax2.c:10276 socket_process: Call rejected by 63.211.239.29: No such context/extension |
| 18:09.24 | [TK]D-Fender | evilbit: And when you can count digits you'll be a whole lot more successful. |
| 18:09.26 | [TK]D-Fender | ^^^^^^^^^ |
| 18:09.59 | p3nguin | counting, as well as understanding the patterns. |
| 18:10.10 | [TK]D-Fender | evilbit: also paying attention to the pattern's requirements |
| 18:10.12 | structz | tzafrir_laptop, and iax codecs used is alaw |
| 18:10.32 | *** part/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 18:11.00 | evilbit | ok, I see it... |
| 18:11.03 | [TK]D-Fender | evilbit: take Note, there is somethiNg iN there that is stoppiNg your exteN from matchiNg |
| 18:11.24 | [TK]D-Fender | take a hiNt <--------- |
| 18:11.28 | p3nguin | hahaha |
| 18:11.32 | rare1980_ | this syntax is not correct i know.. exten => 01915800186,1,Dial(SIP/{EXTEN}) ... but wht would be the correct syntax to recevie incoming calls on any extension? |
| 18:11.34 | evilbit | I've got it... thanks |
| 18:11.49 | [TK]D-Fender | rare1980_: that isn't "any", that is a SPECIFIC number |
| 18:12.12 | p3nguin | rare1980_: If you REALLY mean ANY extension, the pattern would be '_.' |
| 18:12.18 | [TK]D-Fender | rare1980_: and that exten will ONLY call SIP/01915800186 which I guess had better be a valid peer |
| 18:12.20 | p3nguin | But that's a terrible idea. |
| 18:12.32 | [TK]D-Fender | p3nguin: ALMOST any :) |
| 18:12.39 | [TK]D-Fender | p3nguin: but yeah, it's horrid |
| 18:12.43 | rare1980_ | like this all fones will ring? |
| 18:12.48 | p3nguin | no |
| 18:12.48 | boch | why i have no sound when playback on channels that hasnt been answered?? |
| 18:12.52 | rare1980_ | then |
| 18:12.55 | [TK]D-Fender | rare1980_: No. You aren't dialing multiple devices <----------- |
| 18:12.59 | p3nguin | You didn't ask to make phones ring. |
| 18:13.03 | [TK]D-Fender | rare1980_: Loko at what you are Dial()-ing <- |
| 18:13.08 | rare1980_ | i need all my sip fone to ring when any call comes in |
| 18:13.08 | p3nguin | You asked how to make it match any extension. |
| 18:13.16 | rare1980_ | sorry my bad.. |
| 18:13.20 | [TK]D-Fender | rare1980_: One device. Which because of yor lack of comprehension probably isn't even a device |
| 18:13.27 | rare1980_ | i mean that |
| 18:13.47 | [TK]D-Fender | rare1980_: ,Dial(SIP/{EXTEN}) <--- this isn't MULTIPLE devices. it is ONE. |
| 18:13.56 | rare1980_ | humm |
| 18:14.04 | p3nguin | If you're trying to take calls from outside, use your DID number for the extension (as long as that is the number the telco is sending). |
| 18:14.09 | [TK]D-Fender | rare1980_: and as I saaid.. not even really "variabel" because the exten that calls it is fixed |
| 18:14.11 | rare1980_ | wht syntax would be for multiple devices |
| 18:14.12 | rare1980_ | |? |
| 18:14.32 | [TK]D-Fender | rare1980_: Dial(SIP/john&SIP/fred&SIP/mary) |
| 18:14.42 | rare1980_ | humm |
| 18:14.43 | [TK]D-Fender | rare1980_: Go read the INSTRUCTIONS |
| 18:14.48 | [TK]D-Fender | rare1980_: "core show application dial" |
| 18:14.48 | rare1980_ | thanks |
| 18:14.57 | rare1980_ | let me read it |
| 18:14.59 | rare1980_ | thanks |
| 18:15.02 | p3nguin | Don't forget SIP/peter and SIP/paul! |
| 18:15.18 | rare1980_ | right |
| 18:16.58 | evilbit | exten => _NXXNXXXXXX,1,Noop(CALLERID(number)=${OUTCID}) should match 2155551212, right? |
| 18:17.22 | *** join/#asterisk geneticx (~geneticx@host-208-88-126-198.biznesshosting.net) |
| 18:19.01 | geneticx | Hello, I've been getting a lot of log messages with "Call from '1021' to extension '00...' has been rejected" this happens with different extensions.. should I be concerned? |
| 18:19.34 | [TK]D-Fender | evilbit: Looks good |
| 18:19.57 | mocker | geneticx: Interesting, do the numbers look like international call? |
| 18:20.05 | evilbit | ok, great... and it seems to match according to console output but the CID doesn't get reset to OUTCID |
| 18:20.06 | [TK]D-Fender | geneticx: probably being hacked at |
| 18:20.17 | [TK]D-Fender | evilbit: PASTEBIn <-------- |
| 18:20.24 | geneticx | mocker: yes they do |
| 18:20.32 | geneticx | 9011.. |
| 18:21.02 | pabelanger | geneticx: WHo is user 1021? |
| 18:21.33 | geneticx | this an extension I have set up for someone here |
| 18:21.54 | geneticx | but it happens on different extensions as well |
| 18:22.20 | pabelanger | geneticx: I assume Asterisk is on the Internet with no firewalls / ACLs? |
| 18:22.44 | evilbit | http://pastebin.com/XEgSfE4e |
| 18:22.53 | mocker | geneticx: And you probably have short digit based passwords for all your users? |
| 18:23.18 | pabelanger | geneticx: *CLI> sip show peer 1021 |
| 18:23.57 | freezey | i have 2 extensions that are remote employees.. when trying to transfer them over dundi/iax2 the call fails all other extensions work.. any reason why? |
| 18:24.05 | evilbit | does OUTCID need to be a global? |
| 18:25.05 | pabelanger | freezey: No, but if you provide a debug log we can help |
| 18:26.07 | p3nguin | evilbit: What is OUTCID? |
| 18:27.00 | evilbit | OUTCID is defined in sip.conf: http://pastebin.com/Cecs8VT3 |
| 18:27.08 | freezey | i have 2 offices.. all extensions transfer over no issues.. i have 2 extensions that have a follow-me numbers added to them and dont have an actual physical phone.. those 2 numbers are the only ones that get a 504 error when calling from the other PBX where the extensions are not housed. |
| 18:27.08 | [TK]D-Fender | evilbit: PASTEBIN <----------------- |
| 18:27.11 | freezey | let me grab the log |
| 18:27.16 | evilbit | I did :-) |
| 18:27.25 | structz | tzafrir_laptop, ?!?! |
| 18:27.41 | [TK]D-Fender | evilbit: - Executing [12155551212@internal:1] NoOp("SIP/evilbit-00000004", "CALLERID(number)=2155551212") in new stack |
| 18:27.53 | [TK]D-Fender | evilbit: Because you NOOP'd it instead of SET-ing it <-- |
| 18:28.11 | evilbit | ok, someone here said to Noop |
| 18:28.17 | evilbit | lol... let me change it to set |
| 18:28.18 | freezey | pabelanger: http://pastebin.ca/1891933 |
| 18:28.20 | [TK]D-Fender | evilbit: SILLY |
| 18:28.32 | [TK]D-Fender | evilbit: They just wanted to prove what it held. Whoopi-shit :p |
| 18:28.38 | *** join/#asterisk waKKu (~waKKu@unaffiliated/wakku) |
| 18:28.40 | waKKu | hi fellow |
| 18:28.41 | [TK]D-Fender | evilbit: Doing = proving :) |
| 18:28.47 | evilbit | hahah, ok |
| 18:29.02 | p3nguin | I really dislike NoOp() at times. |
| 18:29.18 | pabelanger | freezey: Asterisk is telling you the problem. 'Sent into invalid extension '7042' in context 'lookupdundi' on SIP/5320-b7a0a670' |
| 18:29.28 | [TK]D-Fender | p3nguin: Nothing wrong with it... when what you need isn't equally proven in a more practical place |
| 18:29.42 | pabelanger | freezey: So, check your lookupdundi macro |
| 18:29.52 | p3nguin | It's not the app itself that I don't like... |
| 18:29.58 | p3nguin | It's that so many people abuse it. |
| 18:29.59 | [TK]D-Fender | pabelanger: context 'lookupdundi' <--------- not a macro |
| 18:30.01 | evilbit | nice! thanks very much!!!!!!! |
| 18:30.11 | *** part/#asterisk mintee (442a1371@gateway/web/freenode/ip.68.42.19.113) |
| 18:30.12 | [TK]D-Fender | evilbit: 27th times the charm... |
| 18:30.26 | waKKu | hey folks.. where can i find docs about how to use "userfield" option for cdr billing? |
| 18:30.30 | evilbit | hahaha, now I can go drink myself into obilivion ;-) |
| 18:30.40 | grinder13 | i 'm still having problems with the IAX channel. IAX extensions fail to register. could anybody take a look at this, please? http://pastebin.com/q5YyYz6k Maybe it's something so obvious that I can't see |
| 18:30.51 | tzafrir_laptop | structz, are you sure that the fax call is not detected as such somewhere? (and maybe mishandled)? |
| 18:31.00 | waKKu | i set it in cdr.conf (loguserfield=yes), but no changes on either Master.csv or postgresql db |
| 18:31.02 | pabelanger | [TK]D-Fender: Ya, my mistake |
| 18:31.26 | evilbit | thanks again, I really do appreciate the help |
| 18:31.44 | SiNGLer | waKKu: do you set CDR(userfield) in dialplan? |
| 18:31.51 | freezey | pabelanger: yeah but the lookupdundi works perfectly for physical phones.. but taht error you see only shows up when its trying to do a lookupdundi to the other PBX for a virtual extension |
| 18:32.06 | [TK]D-Fender | evilbit: Progress... you should have picked up some very important essentials here. |
| 18:32.08 | waKKu | SiNGLer nop... |
| 18:32.08 | freezey | when it transfers to an actual hardware phone it works fine |
| 18:32.11 | pabelanger | grinder13: enable tcpdump and see if you can trace the connection. |
| 18:32.18 | waKKu | SiNGLer what is the docs about this option? userfield |
| 18:32.20 | structz | tzafrir_laptop, the mediagateway detects fax on port X .. but the problem is not that.. is on the other asterisk server the sends the call thru IAX2(not trunk) |
| 18:32.51 | SiNGLer | waKKu: on userfield you can set your data, setting mentioned variable |
| 18:32.53 | pabelanger | freezey: *CLI> dialplan show 7042@lookupdundi |
| 18:33.34 | waKKu | hm.. SiNGLer ok.. i got it.. but what is the right docs to read about this ? I'm going to send it to a friend |
| 18:33.49 | waKKu | don't want to say: type it, 'coz i said so ;) |
| 18:34.34 | SiNGLer | not sure where you can find docs, you can try voip-info.org :) |
| 18:34.40 | grinder13 | pabelanger, can it be a problem with the IAX code? I am running Asterisk from SVN... |
| 18:34.56 | *** join/#asterisk githogori (~githogori@SJC-Office-DHCP-135.mail-abuse.org) |
| 18:35.49 | geneticx | pabelanger: it's not behind a firewall, the only port open though is 2000 tcp |
| 18:36.31 | pabelanger | grinder13: possible, but most likely a network issue. Can you ping 192.168.1.2? Telnet into 192.168.1.2:4569? |
| 18:36.35 | waKKu | SiNGLer hm.. another doubt about it.. what value will be inserted in cdr? My idea is group extensions using a field in sip.conf, for example... |
| 18:37.07 | p3nguin | Why would you be able to telnet to a daemon listening on udp? |
| 18:37.10 | pabelanger | geneticx: huh? |
| 18:37.21 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 18:37.38 | SiNGLer | waKKu: if you do Set(CDR(userfield)=foo) then foo will be inserted into that field |
| 18:37.50 | Mango | Is it possible to send a fake auth rejection using the dial plan, if the caller dials a number I don't like? |
| 18:38.08 | waKKu | yeah.. not funny.. SiNGLer so i'll need to create one Dial for each exten ;/ |
| 18:38.19 | p3nguin | mango: You've mixed up two things: dialing and authentication. |
| 18:38.20 | grinder13 | pabelanger, yes I can ping. not sure about telnet have to check, but the thing is that using ngrep I can see that the PC running Asterisk receives messages on port 4569 |
| 18:38.24 | geneticx | pabelanger: sorry, just getting back to your prior statement about my asterisk box being on the internet |
| 18:38.25 | p3nguin | mango: So what do you actually mean? |
| 18:38.25 | waKKu | i was looking for something i could group extensions ... |
| 18:38.52 | pabelanger | Mango: Umm, why? |
| 18:38.54 | SiNGLer | waKKu: you can group using variables or something :) |
| 18:39.00 | Mango | p3nguin: What is the best way to reject invalid calls in the dial plan? I want to accept anonymous calls for only two numbers. |
| 18:39.02 | p3nguin | wakku: You can dial multiple PHONES using a single extension. |
| 18:39.14 | waKKu | p3nguin yeah.. isn't what i want |
| 18:39.26 | p3nguin | mango: Create extensions to match those numbers and nothing else. |
| 18:39.30 | waKKu | i need to record a value for a group of extensions into cdr |
| 18:39.37 | pabelanger | geneticx: You must have other ports open if you are accepting SIP connections. Either way, move your box off the web and setup a firewall. |
| 18:39.40 | [TK]D-Fender | Mango: Only put those 2 extens in the context unauthed calls LAND IN <- |
| 18:39.51 | p3nguin | mango: You should have a context where unauth ... what he said. |
| 18:39.52 | waKKu | i thought i could use userfield=Value in sip.conf for each exten.. ;/ |
| 18:40.01 | SiNGLer | waKKu: you can set some variable in sip.conf, then use that on userfield |
| 18:40.07 | waKKu | hmmm |
| 18:40.11 | waKKu | that would be useful |
| 18:40.24 | waKKu | SiNGLer what should i read/learn to do that? |
| 18:40.31 | Mango | Thanks gentlemen. |
| 18:40.46 | SiNGLer | waKKu: http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf search setvar |
| 18:40.55 | waKKu | ok |
| 18:41.05 | pabelanger | grinder13: does it transmit a response after it receives a packet? |
| 18:41.12 | grinder13 | pabelanger if you see here: http://pastebin.com/q5YyYz6k, you 'll notice that the local user 1002, cannot register. as for 4000 from client, i dont see anything in the CLI, even though messages are sent to the machine running Asterisk as I said (checked with ngrep) |
| 18:41.36 | waKKu | sounds promising .. lemme test it |
| 18:42.24 | waKKu | btw.. lemme guess, it doesn't work for iax, does it? |
| 18:42.27 | pabelanger | grinder13: Yes, but does the Asterisk box at the far end transmit a reply? Are theses boxes on the name network? Can you ping each box from the other? Telnet from each? |
| 18:42.43 | geneticx | pabelanger: Ok I will |
| 18:43.02 | *** join/#asterisk doctorray (~ray@static-71-177-137-76.lsanca.fios.verizon.net) |
| 18:43.08 | grinder13 | pubelanger: yes, i can. they are on the same subnet. |
| 18:43.34 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.112.66.dsl.dyn.forthnet.gr) |
| 18:44.12 | pabelanger | grinder13: pb the tcpdump from your Asterisk box you are registering too |
| 18:44.14 | freezey | pabelanger: yeah thats not going to make a difference |
| 18:44.20 | geneticx | pabelanger: just out of curiosity, how do you think they are able to try all these international calls though? |
| 18:44.27 | freezey | pabelanger: its how the extesions are interacting with dundi which is causing the problem |
| 18:44.38 | [TK]D-Fender | geneticx: Your PW got hacked |
| 18:44.40 | freezey | pabelanger: what it seems like is dundi doesnt like extensions that are not assigned to a phsycai phone |
| 18:44.45 | [TK]D-Fender | geneticx: Now go lock them down |
| 18:45.07 | structz | tzafrir_laptop, ? saw my last message? |
| 18:45.12 | freezey | pabelanger: and that cant return a mac address to it |
| 18:45.52 | pabelanger | geneticx: Next time it happens, *CLI> sip show peer 1024 : And capture the results. |
| 18:46.10 | tzafrir_laptop | structz, for whatever reason it tries to queue (for sending) too many frames to through the IAX connection |
| 18:46.22 | tzafrir_laptop | Any other messages as to what's blocking it? |
| 18:47.36 | geneticx | [TK]D-Fender: Ok, but locking the box would be enough? I would hate to have to reformat the machine.. =( |
| 18:47.40 | pabelanger | freezey: Asterisk tells you the problem: Sent into invalid extension '7042' in context 'lookupdundi'. Like I said, *CLI> dialplan show 7042@lookupdundi, pb the results |
| 18:47.46 | *** join/#asterisk Ryushin (proxy@windwalker.openinnovations.com) |
| 18:47.48 | geneticx | pabelanger: ok I will, thank you. |
| 18:48.04 | p3nguin | geneticx: If the file system has not become damaged, there's no sense in formating anything. |
| 18:48.19 | *** join/#asterisk RobH (~robh@wikimedia/RobH) |
| 18:48.26 | freezey | pabelanger: yeah i feel like your just not getting it which is the case |
| 18:48.49 | pabelanger | freezey: Fair enough |
| 18:48.50 | freezey | pabelanger: its passing that error because its trying to transfer over DUNDI which needs a mac address to pass from PBX to PBX |
| 18:48.52 | SiNGLer | waKKu: never tryed on iax, but I guess it should work |
| 18:49.09 | geneticx | p3nguin: ok |
| 18:49.30 | freezey | pabelanger: all other extensions transfers fine from PBX A to PBX B but once you try to transfer to an extension without a mac address it fails from PBX A to PBX B |
| 18:49.34 | [TK]D-Fender | geneticx: Hoefully only a SIP hack |
| 18:49.46 | [TK]D-Fender | geneticx: that's what I'd bet on. |
| 18:50.29 | geneticx | I hope so too. Anyway, thanks all |
| 18:51.31 | pabelanger | freezey: o.0 mac address and extension have nothing to do with each other. |
| 18:51.35 | evilbit | is anyone using SkypeForAsterisk? |
| 18:52.00 | pabelanger | freezey: One last time: *CLI> dialplan show 7042@lookupdundi |
| 18:52.02 | [TK]D-Fender | evilbit: No, the other guy just died from an aneurism last week... |
| 18:52.15 | freezey | pabelanger: for dundi does it now? |
| 18:52.16 | evilbit | hahaha |
| 18:52.18 | freezey | does it not |
| 18:52.43 | evilbit | [TK]D-Fender: so, it's not worth messing around with then? |
| 18:52.59 | freezey | pabelanger: when doing your typical dundi lookups it returns extension@macaddress |
| 18:53.12 | [TK]D-Fender | evilbit: I was clearly kidding. First it isn't something I normally consider for "playing around" with as it is a commercial product |
| 18:53.24 | waKKu | SiNGLer hmm.. I tried Set(CDR(userfield)) after add an userfield entry in sip.conf and got a error message: [Jun 29 15:52:17] WARNING[16162]: pbx.c:5959 pbx_builtin_setvar: Ignoring entry 'CDR(userfield)' with no = (and not last 'options' entry) |
| 18:53.31 | freezey | pabelanger: because i tried that command you gave me and it works for none of my extensions but yet the calls still tranfer |
| 18:53.34 | waKKu | did something wrong ? |
| 18:54.03 | SiNGLer | show you dialplan and sip.conf (pastebin.com or smth) |
| 18:54.14 | [TK]D-Fender | waKKu: Yes.. you didnt' SET IT TO ANYTHING |
| 18:54.22 | freezey | its just odd this is happening |
| 18:54.26 | evilbit | [TK]D-Fender: well, the price makes it affordable for a single user installation |
| 18:54.28 | [TK]D-Fender | [14:53]<waKKu>SiNGLer hmm.. I tried Set(CDR(userfield)) <--- no "=", no VALUE |
| 18:54.30 | freezey | pabelanger: let me show you this |
| 18:54.33 | waKKu | [TK]D-Fender but i want to get this value from sip.conf .. |
| 18:54.48 | waKKu | so i dont need to create one dial for each extension |
| 18:54.55 | freezey | pabelanger: http://pastebin.ca/1891967 |
| 18:54.56 | [TK]D-Fender | waKKu: then YOU should put the variable reference there. |
| 18:55.00 | SiNGLer | Set(CDR(userfield)=${var_in_sip_conf}) |
| 18:55.00 | freezey | pabelanger: but 7042 does exist |
| 18:55.02 | waKKu | my idea is to group extensions by a common value for some of them |
| 18:55.11 | waKKu | ohhh |
| 18:55.12 | [TK]D-Fender | waKKu: [14:54]<SiNGLer>Set(CDR(userfield)=${var_in_sip_conf}) <-------- |
| 18:55.13 | waKKu | so easy ;) |
| 18:55.20 | waKKu | thanks folks.. |
| 18:55.27 | SiNGLer | np |
| 18:55.37 | [TK]D-Fender | evilbit: that is per-channel of course. |
| 18:55.40 | freezey | pabelanger: because on the side which i am querying to i can dial that extenion and it works perfectly |
| 18:55.47 | [TK]D-Fender | evilbit: So what do you intend to use it for? |
| 18:55.53 | pabelanger | freezey: entityid :) not mac address. |
| 18:56.20 | evilbit | well, I have a internet tablet that I use to connect to my asterisk box... it would be nice to use the skype module to talk to skype friends from asterisk |
| 18:56.26 | freezey | yeah my bad |
| 18:56.29 | freezey | i just checked haha |
| 18:56.32 | freezey | but check those out tho |
| 18:56.38 | evilbit | but the tablet doesn't have skype that is recent |
| 18:56.38 | freezey | pabelanger: its just odd it is doing that |
| 18:57.21 | pabelanger | freezey: Are 7066@priv and 7042@priv on the same asterisk box? |
| 18:57.27 | freezey | pabelanger: yes sir |
| 18:57.37 | [TK]D-Fender | evilbit: Get the SIP softphones instead |
| 18:57.39 | [TK]D-Fender | them* |
| 18:57.55 | freezey | pabelanger: so if i dial from phone 7066 it calls 7042 perfectly... but if i dial from another PBX where you see that query coming from it fails going to 7042 but not 7066 |
| 18:58.10 | evilbit | they are heavy into skype so it's not really a option |
| 18:58.26 | [TK]D-Fender | evilbit: If its worth it to you... sure, why not |
| 18:58.38 | evilbit | meh, for $66 that's pretty cheap |
| 18:58.42 | pabelanger | freezey: and 7042 exists in the same context as 7066? |
| 18:58.51 | freezey | pabelanger: yup |
| 18:59.05 | evilbit | and my tablet doesn't fall over from trying to act as a supernode ;-) |
| 18:59.12 | freezey | pabelanger: the only difference is 7066 is assigned to an actual deskphone and 7042 is not its just has a follow-me numbers assigned to it |
| 19:00.14 | *** join/#asterisk carrar (tim@osburn.com) |
| 19:01.01 | *** join/#asterisk [Outcast] (~james@64.202.62.5) |
| 19:02.05 | [Outcast] | I just patch asterisk to put the rdnis number in the msgxxxx.txt file for voicemail. Is this useful to anyone else? Should I submit a patch? |
| 19:02.21 | grinder13 | pabelanger, i 've tcpdumped. as i said the IAX request reaches the Asterisk server, but the Asterisk server does not respond back to the client. Furthermore, the extension I configured to be registered by the Asterisk server itself, does not do so, but keeps retrying |
| 19:02.50 | carrar | Outcast, what do you use it for |
| 19:03.14 | [TK]D-Fender | [Outcast]: Yes. I'm sure someone else will care. |
| 19:03.36 | carrar | We all CARE! |
| 19:03.43 | pabelanger | grinder13: which host was running TCPdump? IE: (A) --- Registering ---> (B) |
| 19:04.09 | grinder13 | i 've checked from both the machines just to be sure |
| 19:04.13 | [Outcast] | So if someone redirects their phone to their asterisk exten/voicemail you can tell what number redirected the call |
| 19:04.21 | pabelanger | grinder13: We |
| 19:04.45 | grinder13 | ?? |
| 19:04.52 | pabelanger | grinder13: So, the problem is with (B). Not sending the reply to the request. |
| 19:04.59 | carrar | nice |
| 19:05.09 | pabelanger | grinder13: PB the tcpdump from (B) when (A) registers. |
| 19:05.47 | [Outcast] | We are working on a host VM service of sorts |
| 19:06.03 | [Outcast] | *hosted |
| 19:06.14 | evilbit | another question... are people still using cepstral instead of festival? |
| 19:06.26 | [Outcast] | niether here |
| 19:07.02 | [TK]D-Fender | evilbit: Yes, it is considerably better. And of course isn't free |
| 19:07.11 | [TK]D-Fender | evilbit: though very affordable |
| 19:07.35 | evilbit | yeah, I have a license from several years ago... trying to see if it's valid with the latest asterisk version |
| 19:08.16 | evilbit | hmm, digium doesn't seem to have that module in their rpm repo |
| 19:08.38 | p3nguin | Which module? |
| 19:08.53 | evilbit | app_cestral |
| 19:09.00 | grinder13 | the problem is not only from (A)<--->(B). As I said, Asterisk server does not even register the extension which is configured locally with "register => 1000:1000@192.168.1.2" (192.168.1.2 is the IP of my Asterisk machine). it keeps sending endlessly "Tx-Frame Retry". |
| 19:09.01 | evilbit | er, app_cepstral |
| 19:09.23 | doctorray | Trying to use ExtenSpy for testing here; if the specified extension (SIP) is not in use, it seems to find the first available channel to spy on. Any way to avoid that? |
| 19:10.09 | p3nguin | Sounds like a problem with not understanding the difference between ChanSpy and ExtenSpy. |
| 19:10.11 | [TK]D-Fender | doctorray: SIP is not an "extension", it is a channel type. |
| 19:10.35 | waKKu | [TK]D-Fender should it work in 1.4.26? Set(CDR(userfield)=${userfield}) .. it isn't returning the value from sip.conf (yes, i ran sip reload)... ;/ |
| 19:10.41 | p3nguin | ExtenSpy is for EXTENSIONS. |
| 19:10.45 | doctorray | [TK]D-Fender: I understand, just stating that they are sip extensions that I am looking at |
| 19:10.51 | p3nguin | ChanSpy is for CHANNELS. |
| 19:10.52 | [TK]D-Fender | waKKu: I am not seeing anything worth commenting on here... |
| 19:11.11 | p3nguin | There is no such thing as a SIP extension, so no you don't understand. |
| 19:11.11 | [TK]D-Fender | doctorray: do not use the term "sip extensions" around here. |
| 19:11.16 | pabelanger | grinder13: Yes, but asterisk should ACK the first REQREQ packet. For whatever reason, box (B) is not transmitting the reply. So, (A) will resend the packet again, and again... |
| 19:11.26 | doctorray | I misspoke, I suppose. If I try to extenspy on sip/110 and that phone is onhook, it starts spying on sip/120 |
| 19:11.28 | [TK]D-Fender | doctorray: extension = "exten" line in extensions.conf . |
| 19:11.37 | p3nguin | Again, that's not an extension. |
| 19:11.45 | pabelanger | grinder13: This is why I want to see the TCPDump from box B |
| 19:11.45 | doctorray | *ducks* |
| 19:11.47 | p3nguin | SIP/110 would be a channel to spy on. |
| 19:11.48 | [TK]D-Fender | doctorray: ou are calling things wrong |
| 19:11.51 | waKKu | [TK]D-Fender -- Executing [9902XXXX@internacional:1] Set("SIP/402-00c0b8d0", "CDR(userfield)=") in new stack |
| 19:12.00 | doctorray | I understand now. sorry |
| 19:12.02 | p3nguin | Use ChanSpy. |
| 19:12.11 | [TK]D-Fender | waKKu: that one line proves nothing to me. |
| 19:12.18 | [TK]D-Fender | waKKu: PASTEBIN <- |
| 19:12.20 | p3nguin | If you want to spy on an extension, which you are not trying to do, use ExtenSpy. |
| 19:12.22 | waKKu | ok |
| 19:12.50 | doctorray | p3nguin, [TK]D-Fender, I always appreciate your help. Thank you for clearing that up for me. |
| 19:13.30 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 19:14.06 | grinder13 | you don't get me pabelanger. please see this: http://pastebin.com/q5YyYz6k if you take a look at my iax.conf I have an extension 1002 which is supposed to be registered by the Asterisk server in the local machine, so it is like (B)<--->(B). what I am saying is that it's not only a problem with (A)<--->(B), but with (B)<--->(B) (the server itself). |
| 19:14.14 | waKKu | [TK]D-Fender http://pastebin.com/g7j6YysG |
| 19:14.15 | jdoe | p3nguin: is there actually any advantage to using one over the other? They both use the same code internally. |
| 19:14.37 | p3nguin | Depends on if you want to spy on a channel or on an extension. |
| 19:14.57 | doctorray | I think I was more confused because the ExtenSpy worked when I tried it. |
| 19:15.01 | [TK]D-Fender | waKKu: "userfield=00001" this is not the SETVAR syntax you were shown earlier |
| 19:15.16 | pabelanger | <PROTECTED> |
| 19:15.45 | waKKu | [TK]D-Fender Asterisk sip setvar = variable=value : Channel variable to be set for all calls from this peer/user. (voip-info) |
| 19:15.54 | jdoe | p3nguin: ... yeah, but... oh, wait, nevermind. Heh. |
| 19:15.57 | [TK]D-Fender | ......... |
| 19:15.59 | [TK]D-Fender | ......... |
| 19:15.59 | waKKu | i should type Asterisk sip setvar = .... ? |
| 19:16.01 | waKKu | oh boy |
| 19:16.07 | [TK]D-Fender | waKKu: Setvar=userfield=00001 <------------------ |
| 19:16.13 | waKKu | hmmm |
| 19:16.20 | waKKu | [TK]D-Fender ta ;) |
| 19:16.54 | waKKu | works like a charm ;D |
| 19:17.39 | *** join/#asterisk eduzimrs (~eduzimrs@201.22.86.124.static.gvt.net.br) |
| 19:19.12 | *** join/#asterisk SunnyDP (~scan@bas1-montreal27-2925305223.dsl.bell.ca) |
| 19:20.13 | *** join/#asterisk Mhaddog (~Mhaddog@z65-50-118-232.ips.direcpath.com) |
| 19:21.48 | freezey | pabelanger: any ideas? |
| 19:26.44 | *** join/#asterisk lost_soul (shackett@devio.us) |
| 19:30.56 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 19:31.30 | *** join/#asterisk citywok (~chatzilla@67-134-194-33.dia.static.qwest.net) |
| 19:32.51 | *** join/#asterisk DaneoShiga (~DaneoShig@187.23.81.108) |
| 19:34.27 | DaneoShiga | hi, i'm a php developer that knows nearly nothing of asterisk, there's a command that i'm trying to execute from php, but it fails, return just a "Password:" but when i execute it on asterisk console, it works |
| 19:35.18 | DaneoShiga | someone here could help me or point me in some direction? |
| 19:35.23 | Qwell | and that command...is...what? |
| 19:35.24 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 19:35.44 | *** join/#asterisk fru311 (~mark@static-76-160-182-114.dsl.cavtel.net) |
| 19:35.56 | DaneoShiga | database put fwd $origin $destiny |
| 19:35.59 | WIMPy | DaneoShiga: Yes, elsewehere. AFAIK tere is no place, where Asterisk would prompt for a password. |
| 19:36.15 | Qwell | and how are you trying to run that command? |
| 19:36.32 | DaneoShiga | the complete line is? |
| 19:36.33 | DaneoShiga | System("sudo asterisk -rx " . escapeshellcmd("database put fwd $origem $destino") . " &> /tmp/error"); |
| 19:36.46 | Qwell | well, OBVIOUSLY, sudo is asking for the password |
| 19:36.53 | Qwell | since that's kinda what it does |
| 19:37.05 | DaneoShiga | -_-' |
| 19:37.05 | *** join/#asterisk markfeatherston_ (~markfeath@71.4.41.98.ptr.us.xo.net) |
| 19:37.10 | citywok | I'm working with microsoft Office communications server 2007 R2 (tcp sip), and with flowroute i can transfer calls in to it just fine, but with qwest i can not. flowroute uses gsm, qwest is using ulaw. here's a sip debug of the call, it fails. http://pastebin.com/zmGdT6p5 |
| 19:37.32 | markfeatherston_ | Using the manager interface, how would I get a list of channels from an extension? |
| 19:37.47 | Qwell | from an extension? |
| 19:38.53 | grinder13 | pabelanger, the tcpdump you requested: http://pastebin.com/LgbXbuKk |
| 19:40.14 | evilbit | is it possible to set a variable like OUTCID in voicemail.conf? I'm looking over the html voicemail.conf page but only see variables for timezone |
| 19:41.08 | DaneoShiga | Qwell, tested here, sudo isn't asking for the password, executing exactly the same cmd on terminal returns a update success |
| 19:41.23 | *** join/#asterisk timeshell (~timeshell@gw.lusi.on.ca) |
| 19:41.24 | Qwell | Because you are not apache. |
| 19:41.36 | Qwell | man sudo |
| 19:41.39 | ChannelZ | racist! |
| 19:41.42 | DaneoShiga | lol |
| 19:42.09 | timeshell | Greetings. Can anyone direct me to documentation for connecting a trunk from 1.6.2? |
| 19:42.25 | DaneoShiga | the php script is under asterisk:asterisk user/group |
| 19:43.00 | DaneoShiga | or not, -_-', will try again |
| 19:43.17 | ChannelZ | infobot tell timeshell about book |
| 19:43.41 | Qwell | DaneoShiga: it doesn't matter who owns the script. |
| 19:43.41 | Qwell | it's the user *running* it |
| 19:43.50 | DaneoShiga | ok |
| 19:44.03 | ChannelZ | well they matter in contrast to each other |
| 19:45.12 | *** join/#asterisk diegomad (~mad@201.236.222.130) |
| 19:49.16 | [Outcast] | patch submitted https://issues.asterisk.org/view.php?id=17566 |
| 19:52.02 | *** join/#asterisk DelphiWorld (~Delphi@41.200.23.121) |
| 19:52.04 | DelphiWorld | hi |
| 19:52.11 | DelphiWorld | ilbc is added by default in asterisk? |
| 19:52.16 | DelphiWorld | doe is compiled by default? |
| 19:52.35 | DelphiWorld | hello leifmadsen |
| 19:54.00 | *** join/#asterisk timeshell (~timeshell@gw.lusi.on.ca) |
| 19:54.39 | *** join/#asterisk Tim_Toady (~moi@178.128.16.115.dsl.dyn.forthnet.gr) |
| 19:54.43 | DelphiWorld | i say allow=ilbc right? |
| 19:54.49 | timeshell | Have parameters for registering or using a trunk changed since 1.6.0 to 1.6.2? A trunk I'm trying to register from a 1.6.2 server to a 1.6.0 server is giving a 105 status and refuses to route the call. |
| 19:56.02 | SiNGLer | DelphiWorld: check with "core show translation" if codec present and there is a translation to another |
| 19:57.02 | DelphiWorld | ok |
| 19:57.36 | [TK]D-Fender | timeshell: Nope. Pastebin.... |
| 19:58.12 | timeshell | What is status 105? |
| 20:00.10 | [TK]D-Fender | timeshell: what does "sip show ergistry" say? |
| 20:00.13 | [TK]D-Fender | registry* |
| 20:00.24 | [TK]D-Fender | timeshell: normally that is "request swent" IIRC |
| 20:00.27 | [TK]D-Fender | sent* |
| 20:01.09 | timeshell | shows registered |
| 20:01.10 | timeshell | Hmmm |
| 20:01.39 | [TK]D-Fender | timeshell: go look at the call attempt |
| 20:01.48 | timeshell | Yah, I'm already looking |
| 20:01.59 | *** join/#asterisk terracon (~greisky@CPE001cf097a750-CM0012254076d6.cpe.net.cable.rogers.com) |
| 20:02.03 | timeshell | Says it's circuit-busy |
| 20:02.06 | timeshell | Not sure why |
| 20:03.05 | *** join/#asterisk gadams999 (gadams999@173-165-184-25-atlanta.hfc.comcastbusiness.net) |
| 20:03.07 | [TK]D-Fender | timeshell: Meaningless message. Look in more detail. |
| 20:03.17 | [TK]D-Fender | timeshell: Peer status, SIP debug. |
| 20:03.29 | timeshell | lol |
| 20:03.30 | timeshell | NM |
| 20:03.41 | timeshell | When I have a further question, I'll let you know |
| 20:03.43 | timeshell | thanks |
| 20:04.33 | citywok | [TK]D-Fender: does this sip debug look like anything? Inbound call from Qwest (udp sip ulaw) being sent to microsoft OCS (tcp sip ulaw) http://pastebin.com/zmGdT6p5 |
| 20:04.50 | citywok | same call works if it comes in on either flowroute or bandwidth (udp sip gsm) |
| 20:05.01 | gadams999 | can anyone assist with an IAX call not getting accepted between two asterisknow boxes? weird lack of info on both sides) |
| 20:05.20 | timeshell | From the peer: Call from '5284' to extension '142.46.193.203' rejected because extension not found |
| 20:05.41 | timeshell | Why is it trying to use the server ip as the extension? |
| 20:06.24 | *** join/#asterisk Professional (~Pro@unaffiliated/shani) |
| 20:06.28 | *** join/#asterisk ACK-NAK (~Miranda@home.chicagoventure.com) |
| 20:06.47 | [TK]D-Fender | timeshell: Maybe if you showed us we could tell you |
| 20:06.51 | *** join/#asterisk Researcher (~user@unaffiliated/unafilliate) |
| 20:08.09 | [TK]D-Fender | citywok: I see debug for only HALF of the call |
| 20:08.53 | ACK-NAK | I want to dial a SIP URI, but for certain calls I want to restrict the codec to G.711. My sip.conf [general] seciton correctly prefers g.722. |
| 20:09.23 | [TK]D-Fender | ACK-NAK: CHANNELVARIABLES.TX <--------- read it |
| 20:09.31 | ACK-NAK | Thanks |
| 20:09.44 | citywok | shit, you are right. i had sip debug peer enabled for both peers. i wonder why its not showing up. that may be an issue. lol |
| 20:10.11 | [TK]D-Fender | citywok: You can't enable for multiple peers. |
| 20:10.18 | citywok | oh |
| 20:10.20 | [TK]D-Fender | citywok: It will override the last specific one |
| 20:10.31 | [TK]D-Fender | citywok: All or one. |
| 20:10.40 | citywok | ok |
| 20:10.54 | citywok | can i just run it with the other peer last so we can see 2 seperate halves? they are identical lol |
| 20:10.55 | gadams999 | is there a way to get more details IAX debugging that just 'iax2 set debug' under asterisk 1.6? |
| 20:11.01 | citywok | every call attempt does the exact same thing over and over |
| 20:11.12 | citywok | i can see that when i do sip debugging in OCS |
| 20:12.47 | leifmadsen | Asterisk 1.4.34-rc1 and 1.6.2.10-rc1 are now available! |
| 20:13.19 | DelphiWorld | leifmadsen: topic! |
| 20:13.25 | citywok | [TK]D-Fender: here's the other half of the call. http://pastebin.com/KjQz4HAr |
| 20:13.31 | leifmadsen | DelphiWorld: it's only a release candidate, so no topic update |
| 20:13.38 | DelphiWorld | lol |
| 20:14.06 | *** join/#asterisk kerx (~kerx@38.118.129.34) |
| 20:14.13 | kerx | hi all, anyone know if ChanSpy can stick to an agent? |
| 20:14.27 | kerx | Right now the default behaviour is when an agent get's off the phone, it skips to a new agernt |
| 20:14.28 | citywok | when i do sip debugging on the OCS server, it receives the cancel from *. I'm waiting for Qwest to get back to me with what they see on there and and if anything looks wrong there. |
| 20:15.06 | devdvd | kerx: look at extenspy |
| 20:15.43 | kerx | devdvd, by the sound of it, how would i listen into a call of an agent. do i do a ExtenSpy on the agent's extension in the macro for the queue? |
| 20:15.52 | kerx | hope my question makes sense :) |
| 20:16.05 | ACK-NAK | will CHANNELVARIABLES.TX also tell me how to solve the problem of a one-time switcing off of nat=yes for certain calls to avoid 'glare' |
| 20:16.25 | devdvd | http://www.voip-info.org/wiki/view/Asterisk+cmd+ExtenSpy |
| 20:16.33 | devdvd | ~Extenspy |
| 20:16.44 | [TK]D-Fender | ACK-NAK: Nope, on that front, you're FUBAR'd |
| 20:16.45 | devdvd | ok so that dont work :) but there is the link |
| 20:17.55 | ACK-NAK | [TK]D-Fender: I'm trying to think this through. Could I make a special peer definition in which nat=no but still be able to dial to any URI? In other words not just one single predefined host? |
| 20:18.45 | devdvd | kerx: yea you could probably do it that way |
| 20:19.08 | kerx | devdvd, thanks... do you know which function will allow me to find the channel name of a user whose inside an extension sitting idle? |
| 20:19.57 | *** join/#asterisk boodu (~antoine@host-115-126-167-240.adsl.nautile.nc) |
| 20:19.58 | kerx | i'm basically trying to have a user dial an extension. then use a web interface to redirect that user into a ExtenSpy |
| 20:20.08 | kerx | [TK]D-Fender, hey tk |
| 20:21.41 | kerx | is it possible to place a user inside a idle channel, waiting for AMI events to take over? |
| 20:21.46 | *** join/#asterisk diegomad (~mad@190.146.200.120) |
| 20:22.05 | gadams999 | Is there more debug info available than this: http://pastebin.com/Zj278XqV |
| 20:22.17 | ACK-NAK | [TK]D-Fender: any ideas? |
| 20:24.36 | *** part/#asterisk DelphiWorld (~Delphi@41.200.23.121) |
| 20:24.49 | [TK]D-Fender | [16:17]<ACK-NAK>[TK]D-Fender: I'm trying to think this through. Could I make a special peer definition in which nat=no but still be able to dial to any URI? In other words not just one single predefined host? <- no, but it sounds like a good idea. |
| 20:25.16 | [TK]D-Fender | kerx: Yes. make an exten that has them sit around doing nothing of any great value. |
| 20:26.11 | leifmadsen | libpri 1.4.11.3 is now available: http://www.asterisk.org/node/51385 |
| 20:26.18 | *** topic/#asterisk by leifmadsen -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.9 (2010/06/18), 1.6.0.28, 1.6.1.20 (2010/05/20), 1.4.33.1 (2010/06/22), *-Addons 1.6.1.4, 1.6.0.6 (2010/06/08), 1.6.2.1, 1.4.11 (2010/04/15), dahdi-linux 2.3.0 + dahdi-tools 2.3.0 (2010/04/13), libpri 1.4.11.3 (2010/06/29) -=- Related channels: #asterisknow,#switchvox,#freepbx,#asterisk-dev,#asterisk-b |
| 20:26.34 | Qwell | leifmadsen: what happened there? |
| 20:26.45 | Qwell | also I think it's time to remove 1.6.0/1.6.1 |
| 20:26.50 | *** join/#asterisk Godfather_ (~Godfather@193.153.129.150) |
| 20:26.52 | leifmadsen | Qwell: I think it's time to remove your face |
| 20:27.08 | *** topic/#asterisk by leifmadsen -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.9 (2010/06/18), 1.4.33.1 (2010/06/22), *-Addons 1.6.1.4, 1.6.0.6 (2010/06/08), 1.6.2.1, 1.4.11 (2010/04/15), dahdi-linux 2.3.0 + dahdi-tools 2.3.0 (2010/04/13), libpri 1.4.11.3 (2010/06/29) -=- Related channels: #asterisknow,#switchvox,#freepbx,#asterisk-dev,#asterisk-b |
| 20:27.15 | leifmadsen | apparently xchat on linux has a line limit |
| 20:27.26 | hardwire | ewps |
| 20:27.28 | russellb | or freenode has a limit :-) |
| 20:27.33 | hardwire | just type /topic then finish it up |
| 20:27.36 | hardwire | see what happens |
| 20:27.59 | *** topic/#asterisk by leifmadsen -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.9 (2010/06/18), 1.4.33.1 (2010/06/22), *-Addons 1.6.2.1, 1.4.11 (2010/04/15), dahdi-linux 2.3.0 + dahdi-tools 2.3.0 (2010/04/13), libpri 1.4.11.3 (2010/06/29) -=- Related channels: #asterisknow,#switchvox,#freepbx,#asterisk-dev,#asterisk-b |
| 20:28.02 | *** topic/#asterisk by russellb -> lol topic hax |
| 20:28.08 | leifmadsen | oh snap :) |
| 20:28.33 | *** topic/#asterisk by russellb -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.9 (2010/06/18), 1.4.33.1 (2010/06/22), *-Addons 1.6.2.1, 1.4.11 (2010/04/15), dahdi-linux 2.3.0 + dahdi-tools 2.3.0 (2010/04/13), libpri 1.4.11.3 (2010/06/29) -=- Related channels: #asterisknow,#switchvox,#freepbx,#asterisk-dev,#asterisk-bugs |
| 20:29.14 | *** topic/#asterisk by russellb -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.9 (2010/06/18), 1.4.33.1 (2010/06/22), *-Addons 1.6.2.1, 1.4.11 (2010/04/15), dahdi-linux 2.3.1 (2010/05/25), dahdi-tools 2.3.0 (2010/04/13), libpri 1.4.11.3 (2010/06/29) -=- Related channels: #asterisknow,#switchvox,#freepbx,#asterisk-dev,#asterisk-bugs |
| 20:29.58 | ACK-NAK | Setvar = Set? |
| 20:30.14 | [TK]D-Fender | ACK-NAK: in 1.0 dialplan, yes |
| 20:30.33 | leifmadsen | Setvar()? wow... |
| 20:30.50 | ACK-NAK | deprecated? |
| 20:31.13 | ACK-NAK | The wiki is old |
| 20:31.59 | leifmadsen | not only deprecated, but removed about 3 versions ago |
| 20:32.04 | ACK-NAK | LOL |
| 20:32.11 | *** join/#asterisk fullstop (~fullstop@static-173-210-91-3.ngn.onecommunications.net) |
| 20:32.14 | leifmadsen | like literally... 1.0 was the last version I think |
| 20:32.24 | ACK-NAK | leifmadsen: too funny |
| 20:32.33 | leifmadsen | since, we've got 1.2, 1.4, 1.6.0, 1.6.1, 1.6.2, and soon 1.8 |
| 20:32.39 | russellb | someone should write a good book so that people don't have to use the wiki |
| 20:32.42 | leifmadsen | good thing I'm working on writing a new book :) |
| 20:32.52 | leifmadsen | high fives russellb |
| 20:32.55 | russellb | *gasp* |
| 20:32.58 | ACK-NAK | What's a book? Is that like a paper version of an E-Book? |
| 20:33.10 | [Outcast] | what about the o'rielly book? |
| 20:33.13 | leifmadsen | it's like a scroll without the scroll |
| 20:33.15 | Qwell | ACK-NAK: russellb is just making stuff up again |
| 20:33.15 | russellb | yeah, old school |
| 20:33.22 | leifmadsen | [Outcast]: that's what we're working on |
| 20:33.25 | russellb | [Outcast]: i'm just joking around. leifmadsen is one of the authors of that book. |
| 20:33.30 | paulc | More importantly, can I get the book in iBook format? Oh, that's right - it's all there in the USA but not in Canada.. *sigh* |
| 20:33.37 | Qwell | leifmadsen practically wrote the book on Asterisk |
| 20:33.41 | paulc | is irked by Apple in that area |
| 20:33.54 | leifmadsen | paulc: actually 2/3s of the authors of the Asterisk book are Canadian if that helps :) |
| 20:33.54 | [Outcast] | leifmadsen: I use to work for Paul Mahler |
| 20:34.00 | paulc | wonders about Amazon's kindle app on iPhone/iPad and their "520,000 titles" |
| 20:34.10 | *** join/#asterisk g_r_eek (~g_r_eek@ppp-94-67-69-190.home.otenet.gr) |
| 20:34.15 | paulc | leifmadsen: aye, true that - you're oot east eh? (Vancouver BC here) |
| 20:34.20 | ACK-NAK | Ah... that explains all of the hockey references |
| 20:34.20 | leifmadsen | Toronto |
| 20:34.23 | ACK-NAK | in the book |
| 20:34.26 | leifmadsen | :) |
| 20:34.33 | [Outcast] | leifmadsen: what did you think of his book? |
| 20:34.34 | leifmadsen | there were a lot of hockey references? |
| 20:34.45 | [TK]D-Fender | checkout time, BBIAB |
| 20:34.46 | leifmadsen | [Outcast]: which one? |
| 20:34.46 | Qwell | leifmadsen isn't a real canadian. he doesn't like hockey. |
| 20:34.52 | paulc | I got told off for saying "out east" the other day. Apparently it's not very Canadian. It's "out west" or "back east". |
| 20:34.54 | leifmadsen | I love hockey! :) |
| 20:34.56 | ACK-NAK | leifmadsen: no. Just high-fivin' Canada |
| 20:35.12 | paulc | I was born in Canada, grew up in the UK, talk like a Brit.. hockey's "alright".. not sure if I'm a "real" Canadian or not ;-) |
| 20:35.18 | ACK-NAK | You know, Canada. The country that built the robotic arm that the Space Shuttle was built around? |
| 20:35.21 | *** join/#asterisk srbaker (~srbaker@S0106c03f0e8ab2c2.cc.shawcable.net) |
| 20:35.41 | russellb | there are too many Canadians in Canada |
| 20:35.50 | russellb | or at least that's what I have been told |
| 20:36.36 | [Outcast] | leifmadsen: "voip telephony with asterisk" |
| 20:36.40 | paulc | 34 million at the last count, apparently |
| 20:36.47 | leifmadsen | [Outcast]: I don't read other Asterisk books |
| 20:36.57 | ACK-NAK | russellb: Don't believe the hype. That's just Saddam's propaganda to bring down Terrence and Phillip |
| 20:37.40 | paulc | WaitExten(x) - it just continues on and doesn't go to the "t" extension. Does auto fallthrough change that (to enable t?) |
| 20:39.17 | leifmadsen | ACK-NAK: http://www.youtube.com/watch?v=CCHNL3-FZOA&feature=related :) |
| 20:39.31 | *** join/#asterisk Godfather_ (~Godfather@193.153.129.150) |
| 20:40.09 | ACK-NAK | leifmadsen: Nice |
| 20:40.37 | fullstop | I'm having a problem with my sip channel not receiving DTMF, but I see the DTMF (RFC2833) packets in the RTP log. |
| 20:41.33 | *** part/#asterisk ACK-NAK (~Miranda@home.chicagoventure.com) |
| 20:42.39 | fullstop | It works for a while, and then stops. When it stops functioning does not appear to be a function of time or location in the dial plan. |
| 20:43.26 | paulc | fullstop: I'm having similar issues.. playing around with some Background(..) and WaitExten(..) stuff.. from a Polycom.. DTMFs work fine, I navigate around, then at some point, the DTMFs no longer work - similar symptoms to yours? |
| 20:43.53 | fullstop | paulc: yes |
| 20:44.40 | fullstop | paulc: I don't have any SIP phones involved, though. We are using an ITSP -- we register the SIP and receive sip calls with g729 encoding. |
| 20:44.42 | fullstop | http://pastebin.com/BvisA5TB |
| 20:45.13 | fullstop | Also using Background/WaitExten |
| 20:45.53 | fullstop | I tried 1.4.32 and 1.6.2.9, and it appears to happen on both. |
| 20:46.56 | ChannelZ | I think the last time I saw things 'ignored' was because the signalling method was wrong, but it's odd that it works 'for awhile' and then stops |
| 20:47.11 | paulc | fullstop: hmm.. dunno what to say really.. I'm doing the exact same test as I did earlier, and it's working fine - 4 loops on.. |
| 20:47.11 | ChannelZ | Does it stop in the middle of the same call? |
| 20:47.33 | fullstop | ChannelZ: It stops in the middle of the call (DTMF, but the call proceeds) |
| 20:47.48 | fullstop | ChannelZ: no dahdi / zap here, so I don't think it is a signalling issue. |
| 20:47.59 | *** join/#asterisk devmod (~devmod@c-76-100-208-204.hsd1.md.comcast.net) |
| 20:48.01 | paulc | scrolling around the menu over and over, works a treat.. before it definitely dropped out sooner than this.. |
| 20:48.06 | devmod | hello |
| 20:48.19 | ChannelZ | dtmf signalling I mean |
| 20:48.33 | fullstop | Also, I do not believe packet loss is a problem because I see the RTP in the log. |
| 20:48.33 | ChannelZ | As in sending rfc-whateveritis vs inband vs sip info |
| 20:48.38 | devmod | how can I send callers to different contexts when receiving SIP calls? (I cant use domain= bc my sip proxy is routing the calls to an asterisk instance running on private network) |
| 20:49.13 | ChannelZ | devmod: they go to whatever context is set for that sip peer in sip.conf |
| 20:49.16 | fullstop | ChannelZ: I am using RFC2883. My provider does not send sip info, and g729 won't work with inband. |
| 20:49.41 | *** join/#asterisk iscario (~quassel@laureades.davout02.net1.nerim.net) |
| 20:49.50 | devmod | ChannelZ: the endpoints are registered to another sip proxy, they are just calling into a set of queues. |
| 20:49.56 | fullstop | paulc: I've bounced around for 30 minutes or more before I triggered it. |
| 20:51.05 | fullstop | It's like rtp.c is not passing the data to channel.c. |
| 20:51.30 | fullstop | At a high level, how does threading in asterisk look? 1 thread per call? |
| 20:52.06 | ChannelZ | fullstop: do you know what system the calls are coming from from your ITSP (Asterisk?) |
| 20:52.37 | *** join/#asterisk grinder13 (~grinder@cpc2-sgyl2-0-0-cust1286.sgyl.cable.virginmedia.com) |
| 20:52.40 | paulc | getting a bit over my head there.. but possibly related (your RTP comment rang a bell).. I was using local channels to bridge a Playback with a Meetme, to play a recording into a conference. After a while, the audio stops. Doing core show channel shows frames in or out increasing, but only one, not both. |
| 20:52.42 | fullstop | ChannelZ: It's a black box to us. Comes across MPLS, and we pick up from there. |
| 20:52.59 | lirakis | nods off |
| 20:53.08 | paulc | something timing related? (the PC doesn't have a hardware telephony card in it?) or barking up the wrong tree with DAHDI? |
| 20:53.11 | ChannelZ | devmod: the calls have to be coming into your box somehow.... |
| 20:53.45 | devmod | ChannelZ: right through a single sip proxy |
| 20:54.04 | fullstop | paulc: I don't think it is interrupt timing related. |
| 20:55.02 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 20:55.05 | ChannelZ | devdvd: ok so they are coming into your box as SIP then. So *something* has to be defined on your side to send those calls somewhere, so make them go to whatever context you want. |
| 20:55.15 | ChannelZ | err devmod tht was for |
| 20:55.17 | *** join/#asterisk kfife (~Miranda@home.chicagoventure.com) |
| 20:55.39 | kfife | Gentlemen. (Tips his hat) |
| 20:56.07 | fullstop | kfife is a spy! |
| 20:56.12 | ChannelZ | devmod: If you need to send specific callERS coming to you on the same peer, then you'll have to write whatever logic into your dialplan to figure out who that is and then Goto() them someplace else |
| 20:56.37 | kfife | fullstop: Not true. My KGB boss is only making me out to look that way |
| 20:56.45 | devmod | ChannelZ: all right i see |
| 20:57.11 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 20:57.51 | ChannelZ | like if you have multiple DIDs but come across the same [my-itsp] peer, just make your DID extensions immediately jump via Goto to a separate context if that's what you want to do |
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| 21:03.16 | *** join/#asterisk mboehn (mathias@mboehn.alfa-skk.pr0jectX.net) |
| 21:04.53 | devmod | can I pass like an extra header and examine that in the dialplan? |
| 21:05.16 | [Outcast] | devmod: sip header? |
| 21:05.22 | devmod | yes |
| 21:05.44 | [TK]D-Fender | caller = CALLERID = do what you want with it |
| 21:06.38 | devmod | I need to mantain the caller id being passed through |
| 21:06.38 | [Outcast] | devmod: check out http://www.voip-info.org/wiki/view/Asterisk+func+sip_header |
| 21:07.17 | [TK]D-Fender | devmod: then do it. You have our permission |
| 21:07.28 | devmod | [TK]D-Fender: lol |
| 21:13.24 | fullstop | devmod: you don't have my permission. |
| 21:13.46 | *** join/#asterisk roni (~roni@190.196.71.206) |
| 21:14.13 | roni | hi everybody .. can somebody give me a little help about dial patterns |
| 21:14.24 | roni | where can i get some information |
| 21:14.30 | kfife | ~book |
| 21:14.31 | 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 |
| 21:15.16 | kfife | roni: its a good start |
| 21:16.14 | kfife | c/its/it's |
| 21:19.20 | kerx | [TK]D-Fender, you mentioned place them in an extension that doesn't have them do anything |
| 21:19.29 | kerx | what app should i run in that extension? |
| 21:19.52 | nightwalk | roni: Are you serious? Google 'asterisk dialplan howto'...tons of sites out there with useful howto's and such. Most likely, you'll end up at the same site I did, though: voip-info.org. Good stuff there |
| 21:21.27 | roni | thanks .. is that i dont know how to search .. my english is not good .. |
| 21:21.37 | [TK]D-Fender | kerx: maybe have them sit in MusicOnHold |
| 21:22.00 | [TK]D-Fender | roni: Chapter 5 <- |
| 21:22.02 | kerx | ahhh... why didn't i think of that :) |
| 21:23.54 | devmod | [Outcast]: the sip header did it. thanks! |
| 21:24.40 | fullstop | paulc: are you using g729? |
| 21:27.34 | *** join/#asterisk guilhermebr (~Guilherme@189.63.88.174) |
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| 21:28.32 | paulc | fullstop: nope.. ulaw all the way :-) |
| 21:29.37 | *** join/#asterisk mrchrisadams (~Adium@CPE-121-217-218-126.lnse3.cht.bigpond.net.au) |
| 21:30.09 | fullstop | paulc: are you using inband DTMF or the RFC2833? |
| 21:33.56 | paulc | fullstop: RFC2833 as far as I know.. that said, I didn't double check my sip.conf, but I'm pretty sure I set up a peer template to use RFC2833 |
| 21:34.36 | fullstop | paulc: inband would probably work for you since you are using ulaw. |
| 21:34.50 | *** join/#asterisk lost_soul (shackett@devio.us) |
| 21:38.27 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 21:41.33 | paulc | fullstop: yes, it probably would.. but we're mixing direct SIP phone calls in with PSTN via gateways and/or providers.. so if they're clamping DTMF and sending RFC2833 I'd have a problem. Prefer to just say "We're RFC2833" and be done with it really... |
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| 21:54.55 | *** join/#asterisk krdian (krdian@killer.radom.net) |
| 21:54.59 | krdian | hi |
| 21:55.20 | *** join/#asterisk SomethingISODD (~Dan@d205-206-133-103.abhsia.telus.net) |
| 21:55.39 | SomethingISODD | Hello i was wondering if this channel supports a2billing by chance? |
| 21:56.05 | *** join/#asterisk Fubard (~brawr@173-165-148-200-fresno.hfc.comcastbusiness.net) |
| 21:58.00 | Fubard | Is there a history of fax to email calls some place? Due to a DNS record error I was getting rejected from email servers and have all these .tif's but no idea who they were intended to go to or who sent them (other than individually reading cover pages, if they are there). |
| 21:58.56 | [TK]D-Fender | SomethingISODD: No. |
| 21:59.15 | [TK]D-Fender | Fubard: There is no such thing as a "fax to e-mail call". |
| 21:59.22 | [TK]D-Fender | Fubard: there are just "calls" |
| 21:59.50 | kfife | [TK]D-Fender: Fubard: It does sound like a FooBar though. :-) |
| 22:00.41 | Fubard | [TK]D-Fender: I am not finding any calls in the CDR for any of the DIDs for fax to email, in the UI I use. Am hoping like a little boy at christmas that there is some sort of record of these calls kept by asterisk some palce |
| 22:00.54 | [TK]D-Fender | kfife: Pathetic fallacy or self-fulfilling prophecy at its best |
| 22:01.12 | [TK]D-Fender | Fubard: Maybe your UI is a flaming piece of shit :) |
| 22:01.29 | Fubard | [TK]D-Fender: Agreed. |
| 22:01.53 | kfife | [TK]D-Fender: I love the self fulfilling prophecy refernece |
| 22:07.51 | SomethingISODD | [TK]D-Fender ok thanks do you know if there is any channels that supports a2b? |
| 22:13.54 | Fubard | exit |
| 22:13.59 | Fubard | ...nothing. |
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| 22:19.43 | boodu | hello |
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| 22:55.23 | *** join/#asterisk hhoffman (~hhoffman@pool-74-109-47-229.phlapa.east.verizon.net) |
| 22:56.12 | hhoffman | is it possible to have a menu for clients connecting via SIP? Or does it not work that way since they haven't dialed a extension like when calling a DID? |
| 22:56.54 | *** part/#asterisk oldhack (~jfincher@cpe-24-27-56-221.austin.res.rr.com) |
| 22:59.55 | pabelanger-lap | hhoffman: yes |
| 23:01.30 | hhoffman | pabelanger-lap: any pointers on docs on how to tell asterisk to provide the user with the menu? |
| 23:01.40 | pabelanger-lap | ~book |
| 23:01.41 | 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 |
| 23:01.49 | pabelanger-lap | hhoffman: ^^^ |
| 23:01.49 | hhoffman | danke |
| 23:02.29 | jaytee | hhoffman, checkout the IVR samples in the book and on www.voip-info.org. |
| 23:02.33 | hhoffman | hmm, maybe I have it setup right... is there something that would cause voice to work only sometimes? |
| 23:02.57 | hhoffman | yeah, I pulled the IVR auto-attendant script menus. it works really well |
| 23:03.00 | *** join/#asterisk jks (jks@193.189.93.254) |
| 23:03.08 | jaytee | hhoffman, are you using the Answer() command as the first priority? |
| 23:03.50 | hhoffman | yep |
| 23:04.10 | hhoffman | well, no... as the first priority I'm assigning a context... it's the second |
| 23:05.10 | jaytee | hhoffman, when you say voice only works sometimes do you mean on some calls and not others or the audio drops out during a call in progress? |
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