00:03.07 | *** join/#asterisk jksM (jks@193.189.93.254) |
00:16.46 | *** join/#asterisk jakent (~john@soleil.johnkent.mooo.com) |
00:18.42 | *** join/#asterisk Xetrov` (~xetrov@unaffiliated/xetrov/x-827361) |
00:31.55 | *** join/#asterisk tdjacobs (~tiagoj@host002.ht2p.net) |
00:32.00 | *** join/#asterisk Torrieri (~Torrieri@nelug/crew/torrieri) |
00:32.34 | tdjacobs | hello guys, does asterisk have the capability to mix voices on the server? |
00:33.11 | tdjacobs | I want to integrate that with a existing voice talking solution.. I want to mix all voices on server to reduce bandwith usage... |
00:33.30 | tdjacobs | was thinking about using asterisk for that, is it possible? |
00:34.00 | tdjacobs | I have RAW audio, want to send it to somewhere (possible asterisk), and get merged/mixed audio in raw back |
00:39.43 | voipmonk | mix all voices? |
00:39.46 | voipmonk | explain |
00:39.51 | voipmonk | where do these voices come from? |
00:39.54 | voipmonk | conference? |
00:44.55 | tdjacobs | voipmonk: yes... |
00:45.01 | tdjacobs | in fact, its for e-learning |
00:45.38 | tdjacobs | I know that we can just understand one people talking at same time.. but I really want to mix that sounds togheter providing a "ROOM" feeling... |
00:46.07 | voipmonk | for recording? |
00:46.17 | voipmonk | you want to record the audio from a conference room? |
00:47.03 | *** join/#asterisk shimizu (~shimizu@li142-120.members.linode.com) |
00:48.47 | shimizu | is it possible for asterisk to grab sip user password from external source, say some script backend? |
00:48.59 | tdjacobs | no |
00:49.10 | voipmonk | what do you mean shimizu ? |
00:49.18 | tdjacobs | I want to grab it real time and reencode to my existing software... |
00:49.28 | tdjacobs | I think asterisk is much bigger than the thing I need |
00:49.33 | *** join/#asterisk Sedorox (brandon@smartserv/cna/Sedorox) |
00:49.41 | tdjacobs | does it uses a software mixer? |
00:49.50 | voipmonk | u want to grab a password from where and reencode the password into your existing software? |
00:49.53 | bmoraca_work | shimizu: it can get SIP authentication parameters in a database (realtime), but it cannot do it otherwise. |
00:50.02 | shimizu | i have an database where passwords are stored with salt + sha1 |
00:50.11 | tdjacobs | I can't use hardware mixer because of many "rooms" at same time on same server |
00:50.29 | shimizu | how can i authenticate user's to that database? |
00:50.34 | voipmonk | ahhh |
00:51.00 | ChannelZ | tdjacobs: asterisk does mix in software yes - it's all data |
00:51.02 | bmoraca_work | shimizu: you probably cannot. you would have to code that yourself |
00:51.09 | voipmonk | you can do that with asterisk "REALTIME" functions or the MYSQL application from the asterisk-addons package, or a php script |
00:51.26 | tdjacobs | ChannelZ: its internal asterisk code or 3rd part? |
00:51.29 | voipmonk | salt + sha1 will need to be dealt with before hand tho |
00:51.36 | ChannelZ | tdjacobs: it's internal |
00:51.39 | tdjacobs | ChannelZ: any API? |
00:51.42 | tdjacobs | or doc? |
00:51.48 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
00:51.49 | shimizu | voipmonk: i have api |
00:52.16 | shimizu | say some function check_password(username,plaintextpassword) |
00:52.41 | ChannelZ | tdjacobs: do you just want to record a conference? |
00:53.12 | shimizu | voipmonk: php script ? |
00:53.42 | bmoraca_work | ChannelZ: no, he wants asterisk to mix "room" sounds with his audio and then pass it back to another system to go out to the audience |
00:53.50 | tdjacobs | ChannelZ: no, I am not a asterisk user yet... I have another solution (red5), and I want to MIX the sound of all people in a room (via software). |
00:54.45 | tdjacobs | so, I have decoded RAW samples and I want to pass some channels of audio to a Software based Mixer and get the resulting channel back, encode and send to application |
00:55.01 | tdjacobs | I am looking for a software based mixer... |
00:55.05 | bmoraca_work | shimizu: you cannot authenticate SIP users by default against anything but a plain text document or a SQL database. you can use MD5 encrypted shared secrets, but if you want SHA1, it's just not going to work. |
00:55.18 | bmoraca_work | shimizu: the capability has not been programmed in to asterisk. |
00:55.40 | bmoraca_work | shimizu: i suppose you could adapt the realtime modules to add that capability, but you will not be able to do it otherwise. |
00:55.59 | ChannelZ | well I suppose you could write your own softphone type app to act as a robot participant in the conference, and then you'd get an RTP stream you could do whatever you wanted with |
00:57.01 | shimizu | bmoraca_work: hmm, that's bad |
00:58.00 | tdjacobs | ChannelZ: good idea, its trivial to create a room on asterisk? any howto? I am downloading asteriskNow |
00:58.11 | shimizu | bmoraca_work: i guess freeradius can help me |
00:58.26 | bmoraca_work | shimizu: actually, now that I think about it, you probably can't. i believe that shared secrets in SIP messages are hashed with MD5. your SHA1 are one-way hashes, and the SIP hashes are one-way MD5. they're not compatible |
00:58.44 | bmoraca_work | shimizu: no, no it can't, because Asterisk cannot authenticate against a RADIUS server currently either. |
00:59.06 | bmoraca_work | shimizu: you could use openSER, though, as your registrar server. i believe it can authenticate against RADIUS |
00:59.21 | shimizu | bmoraca_work: yeah, it can |
00:59.44 | *** join/#asterisk icyValk77 (~icyValk77@gateway.ash.thebunker.net) |
01:00.04 | bmoraca_work | shimizu: that must be new (like, today), because last time I checked, asterisk could not authenticate sip peers from a RADIUS server |
01:00.34 | ChannelZ | tdjacobs: well doing conferences in * is not hard no, but it sounded like you already have a phone system in place? or are you wanting to replace what you have? |
01:00.40 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
01:00.49 | *** join/#asterisk ManxPower-work (~EWieling@216.186.151.147) |
01:01.01 | tdjacobs | ChannelZ: I need to go now, will back in 10 minutes |
01:01.05 | tdjacobs | thanks for the moment |
01:01.11 | ChannelZ | oh.. Red5 is just some streaming audio thing |
01:01.16 | ChannelZ | sure |
01:01.23 | tdjacobs | yes, and I am core developper of that ;D |
01:01.26 | tdjacobs | ttyl |
01:01.28 | *** part/#asterisk tdjacobs (~tiagoj@host002.ht2p.net) |
01:01.34 | *** join/#asterisk jmcdowell (~nooe@173-112-87-255.pools.spcsdns.net) |
01:01.39 | jmcdowell | hello all.. |
01:01.55 | jmcdowell | I am back begging for me help with this polycom.. Although I am getting close.. |
01:02.05 | bmoraca_work | best part of the work day = going home |
01:02.14 | ChannelZ | indeedy |
01:02.19 | ChannelZ | shuts down his laptop |
01:02.25 | shimizu | bmoraca_work: i guess I'll just generate new password based on sha1 hashes and give them to users |
01:02.52 | jmcdowell | lol |
01:03.12 | shimizu | bmoraca_work: or some random stuff |
01:03.38 | jmcdowell | You could help this begging lil pesant.. |
01:03.38 | jmcdowell | :D |
01:03.53 | bmoraca_work | shimizu: i'm not sure what you're trying to do, but sha1 hashes are not understood by asterisk. asterisk can use plain text passwords or MD5 hashes. neither way helps against brute force attacks. i usually use a 20-30 character random string. |
01:03.56 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
01:04.21 | bmoraca_work | MD5 hashes make it so that you can't understand the password by looking at the file or the database record. |
01:04.28 | bmoraca_work | that's all |
01:04.31 | bmoraca_work | anyway... |
01:04.35 | bmoraca_work | goes home |
01:04.45 | shimizu | bmoraca_work: thank you |
01:07.38 | jmcdowell | interesting, <mac>-phone1.cfg was over written with over rides.. |
01:10.00 | ManxPower-work | jmcdowell: are you sure it was not <mac>-phone.cfg? |
01:10.43 | ManxPower-work | That's the default overrides file. <mac>-phone1.cfg is a customary name for the file that contains the system admin settings for the phone (registrations, etc) |
01:11.09 | jmcdowell | They were symbolicly linked |
01:11.15 | jmcdowell | so I think that explains it.. |
01:11.22 | jmcdowell | I was confusered. |
01:12.08 | ManxPower-work | jmcdowell: Polycom has a section in the Admin manual and they have a technical paper about how provisioning works. |
01:12.16 | jmcdowell | I have read it 10 times |
01:12.24 | jmcdowell | I can't seem to get it to work |
01:12.26 | ManxPower-work | jmcdowell: the tech paper as well? |
01:12.28 | jmcdowell | But I am getting close.. |
01:12.38 | jmcdowell | I have read this.. |
01:13.12 | ManxPower-work | I just finished rebuilding our polycom provisioning server from scratch |
01:13.54 | ManxPower-work | It worked the first time I tried it in production, except for some very old phones. I did have a test server I used to get it all right first. |
01:13.55 | jmcdowell | lol |
01:14.01 | jmcdowell | That makes me wanna cry |
01:14.26 | jmcdowell | Perhaps I am just getting ahead of myself, I have a-lot of stress going on right now and am finding it harder and harder to focus.. |
01:14.31 | ManxPower-work | jmcdowell: I've been using Polycoms and setting up provisioning servers for them since like 2003. |
01:14.57 | jmcdowell | It provisions, it just doesn't do anything from ther.e |
01:15.09 | ManxPower-work | If you think the docs are crappy today...... |
01:15.21 | jmcdowell | It says "214" as it should, but dialing produces strange behavior on the server side. |
01:15.56 | p3nguin | shimizu: I use apg to create regular, hard-to-guess passwords. |
01:16.26 | *** join/#asterisk Kumbang (~kumbang@rusnas.paume.itb.ac.id) |
01:16.40 | jmcdowell | On the server it never registers, but it tries to dial and is rejected. |
01:16.42 | jmcdowell | It's strange |
01:17.16 | shimizu | p3nguin: thx, i still try to lookup smth on google |
01:17.19 | ManxPower-work | registraton has to do with asterisk sending calls TO the phone, not accepting calls from the phone. |
01:17.42 | p3nguin | shimizu: If you look for smith on google, you'll have a lot of results. Try something more relevant, too. |
01:18.17 | jmcdowell | Hmm.. My softphone registers, and I can see it.. |
01:23.04 | *** part/#asterisk paulc (~paulc@unaffiliated/paulc) |
01:24.14 | *** join/#asterisk etfonhomey (~etfonhome@74-131-159-160.dhcp.insightbb.com) |
01:27.49 | jmcdowell | Can you lay on me, what files, and what order.. Real simple.. |
01:28.08 | jmcdowell | That's what I can't get out of the white paper, because they say one thing, and then say .."Wait, we changed that.." |
01:29.25 | jmcdowell | And.. this http://www.polycom.com/global/documents/whitepapers/configuration_file_management_on_soundpoint_ip_phones.pdf is what I have been reading. |
01:35.40 | *** join/#asterisk jmcdowell (~nooe@173.154.185.70) |
01:35.49 | jmcdowell | wooops |
01:37.08 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
01:37.08 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
01:37.53 | jmcdowell | holds up a sign.. " |
01:38.08 | jmcdowell | holds up a sign.. "Will work for Polycom provisioning tricks.." |
01:41.43 | jmcdowell | folds his ears back, and gets out his big puppy dog eyes... |
01:41.52 | *** join/#asterisk icyValk77 (~icyValk77@gateway.ash.thebunker.net) |
01:43.07 | leifmadsen | jmcdowell: eh? |
01:43.15 | jmcdowell | LOL |
01:43.26 | jmcdowell | Just begging for Polycom provisioning help and tricks.. |
01:43.38 | jmcdowell | but I am reading the white paper again, and seeing it a little easier.. |
01:43.41 | *** join/#asterisk mattwj2002 (~Matt@wikisource/pdpc.active.mattwj2002) |
01:43.45 | mattwj2002 | hi guys |
01:43.52 | jmcdowell | I guess I will know if a few minutes if it works.. |
01:43.54 | jmcdowell | ;D |
01:43.56 | jaytee | jmcdowell, have you gotten a phone to download configs properly |
01:44.02 | *** join/#asterisk coppice (~chatzilla@172.169.232.220.dyn.pacific.net.hk) |
01:44.07 | mattwj2002 | I am looking to buy a Cisco 7960 for my asterisk server |
01:44.21 | mattwj2002 | where is the best place to buy used equipment |
01:44.23 | mattwj2002 | ebay? |
01:44.32 | mattwj2002 | looking for inexpensive |
01:44.33 | leifmadsen | mattwj2002: I have one for sale funny enough |
01:44.42 | jmcdowell | only 5k too.. |
01:44.43 | mattwj2002 | lol |
01:44.46 | mattwj2002 | :O |
01:44.48 | leifmadsen | 2.5k! |
01:45.16 | mattwj2002 | :O |
01:45.20 | leifmadsen | I have no idea how much 7960s go for |
01:45.33 | jmcdowell | I think it's a switch or a phone but dont know.. |
01:45.42 | leifmadsen | phone |
01:45.54 | p3nguin | What would you want for the phone? |
01:45.54 | mattwj2002 | how about .05k ? |
01:46.05 | jmcdowell | lol |
01:46.05 | mattwj2002 | :P |
01:46.08 | jmcdowell | 5 bucks? |
01:46.16 | mattwj2002 | that is actually $50 |
01:46.18 | mattwj2002 | :) |
01:46.18 | jmcdowell | I will sell my 2 mitels for 100 bucks all inclusive |
01:46.28 | jmcdowell | ; |
01:46.29 | jmcdowell | ? |
01:46.38 | jmcdowell | I never claimed to be good @ math. |
01:49.02 | p3nguin | shimizu: If you want a good password generator, use the following to create a single password: apg -a1 -n1 -m13 -x26 -MSNCL -E^[]{}\"? -s |
01:51.53 | leifmadsen | jmcdowell: why the 7960 if you don't mind me asking? |
01:52.05 | jmcdowell | That wasn' tme |
01:52.11 | jmcdowell | that was someone else.. |
01:52.12 | leifmadsen | oops |
01:52.19 | leifmadsen | mattwj2002: ^^^ |
01:52.53 | jmcdowell | Hmmmm.. |
01:53.10 | jmcdowell | How do I tell the phone the "secret" via the xml file.. |
01:53.22 | leifmadsen | which phone? |
01:53.26 | leifmadsen | should be "password" on polycoms |
01:53.36 | leifmadsen | in the <MAC_ADDRESS>-phone.cfg file |
01:53.51 | mattwj2002 | because I have worked with call manager before in a previous job |
01:53.57 | leifmadsen | gotcha |
01:54.04 | jmcdowell | Right, but there is no xml for it. |
01:54.06 | p3nguin | I wouldn't mind replacing my 7940G with a 7960G. |
01:54.09 | mattwj2002 | and I have always wanted to try to hook it up to an asterisk server |
01:54.10 | mattwj2002 | :) |
01:54.13 | leifmadsen | was just curious, because polycom's are pretty much the defacto :) |
01:54.19 | mattwj2002 | yup |
01:54.19 | jmcdowell | I have a procurve 2948 for sale.. |
01:54.20 | mattwj2002 | :) |
01:54.24 | leifmadsen | 7960 is easy -- 7961 not so much |
01:54.36 | mattwj2002 | yeah that is what my friend said |
01:54.45 | mattwj2002 | something about java getting in the way? |
01:55.30 | leifmadsen | well, basically the .xml files are normally generated, so they are crazy difficult to modify by hand to work with asterisk |
01:55.51 | p3nguin | Why would a 7961 not cooperate? It's pretty much the same at 7960, but with different config files for the phone. |
01:56.17 | jmcdowell | reg.1.auth.password="123456" <--- Is that it? |
01:56.18 | leifmadsen | read what I just said again |
01:56.24 | p3nguin | If you can use a text editor, you can edit the file. |
01:56.24 | leifmadsen | jmcdowell: looks like it |
01:56.35 | jmcdowell | The example has a variable in it. |
01:56.39 | jmcdowell | reg.1.auth.password="${SECRET}" |
01:56.55 | leifmadsen | my 7970 was a super pain in the ass -- it never read the configurations correctly, and there were pretty much no examples on the internet because they are generated, and I didn't have a CCM to generate a config from |
01:57.08 | leifmadsen | jmcdowell: OH -- you're using res_phoneprov |
01:57.23 | leifmadsen | jmcdowell: users.conf is where the phones are configured |
01:57.31 | p3nguin | Hmm, that's interesting. |
01:57.37 | leifmadsen | needs to write that blog post about how to use res_phoneprov |
01:57.59 | jmcdowell | I am using no such package |
01:58.05 | jmcdowell | I searched the web to get that VAR |
01:58.19 | jmcdowell | The line does not exist in any of my .cfg files.. |
01:58.28 | jmcdowell | http://svn.dd-wrt.com:8000/dd-wrt/browser/src/router/asterisk/phoneprov/polycom.xml?rev=11933 |
01:58.44 | leifmadsen | phoneprov... ding ding ding |
01:58.46 | p3nguin | I wouldn't expect it to be much different from how a 7912 is configured; you have to edit the plain text file and then run Cisco's config app against the text file to generate a binary config that the phone uses. |
01:58.54 | leifmadsen | jmcdowell: you ARE using such a thing |
01:59.16 | leifmadsen | p3nguin: I've seen the 7912 files -- they 7970s were much more involved from what I remember of it |
01:59.31 | leifmadsen | anyways, it wasn't trivial getting it all configured and the phone to accept all the values |
01:59.46 | jmcdowell | I am editing them manually |
01:59.51 | jmcdowell | I pulled that var from the web |
01:59.52 | *** join/#asterisk icyValk77 (~icyValk77@gateway.ash.thebunker.net) |
02:00.01 | p3nguin | I would like to get a model with a lighted display. |
02:00.06 | leifmadsen | p3nguin: amen |
02:00.20 | leifmadsen | jmcdowell: how did you install asterisk? you're reading from a dd-wrt forum |
02:00.29 | leifmadsen | which sounds like you're using some embedded version |
02:00.38 | coppice | p3nguin torches are quite cheap |
02:00.40 | jmcdowell | I googled for what I thought what I needed |
02:00.47 | jmcdowell | I am using asterisknow |
02:00.50 | leifmadsen | which is talking about res_phoneprov as you're talking about the res_phoneprov template |
02:00.56 | p3nguin | If I had a spare 7940 display, I would attempt to add backlighting to my phone. I just need to have a spare display in case I mess up. |
02:01.13 | leifmadsen | jmcdowell: normally polycom phones are configured via an FTP or HTTP server with the configurations obtained from polycom.com |
02:01.24 | jmcdowell | I am configuring them via FTP |
02:01.31 | jmcdowell | Well,I am trying to anyway |
02:01.37 | *** join/#asterisk f0urtyfive (~noone@75.150.130.121) |
02:01.41 | f0urtyfive | waves |
02:01.49 | f0urtyfive | Anyone have asterisk setup with a GSM modem? |
02:02.01 | f0urtyfive | I'm trying to be able to use the same GSM modem for SMS/Phone notifications |
02:02.59 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
02:04.06 | *** join/#asterisk rnp (~robertnpa@c-76-101-196-166.hsd1.fl.comcast.net) |
02:06.26 | leifmadsen | jmcdowell: well, the configuration you're looking at is incorrect if you're not using res_phoneprov -- you need to use the configuration from the polycom.com site and then upload it to the FTP server you're using |
02:06.44 | *** join/#asterisk rossand (~aross@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
02:07.15 | jmcdowell | I have been trying that, and there is no password statement in it. |
02:07.40 | leifmadsen | there is in the 00000000-phone.cfg file, which you should rename to match the MAC address of your device |
02:07.55 | jmcdowell | let me re-expand the archive and start over |
02:08.03 | jmcdowell | perhaps I have screwed something up. |
02:08.04 | leifmadsen | runs off to hang out with the g/f |
02:08.14 | leifmadsen | it's pretty straight forward :) make sure you get the right archive for your phone |
02:08.56 | jmcdowell | I used their matrix. |
02:11.51 | *** join/#asterisk pawz (~pawz@ppp118-208-82-201.lns20.bne4.internode.on.net) |
02:13.10 | jmcdowell | So the <mac>.cfg is supposed to POINT to all the over config files related to the phone? |
02:13.27 | *** join/#asterisk hluesea (~hulusikah@88.247.127.66) |
02:14.13 | etfonhomey | jmcdowell, yes. |
02:14.26 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
02:14.30 | p3nguin | I know very little about Polycoms, but with Cisco we have a SIPDefault.cfg with config information for all phones, then a SIP<mac>.cfg for all phone-specific settings. I'm sure Polycoms have a similar setup. |
02:15.41 | etfonhomey | Polycom's have a <MAC>.cfg which points to other related config files for the phone. I usually use phoneX.cfg and sip.cfg where sip.cfg contains companywide settings and phoneX.cfg has phone specific settings for phone X. |
02:15.56 | jmcdowell | So what I am asking.. |
02:16.03 | jmcdowell | Is the 003202323.cfg |
02:16.11 | jmcdowell | doesn't contain setting for the phone. |
02:16.16 | jmcdowell | but much like the 00000000000.cfg |
02:16.17 | etfonhomey | <mac>-phone.cfg is an overrides file that the phone uploads when you make changes to the settings on the phone itself. |
02:16.26 | jmcdowell | it contains pointers to the correct config files. |
02:16.35 | jmcdowell | Jesus, that's what I missed. |
02:16.48 | jmcdowell | I have been trying to use 0000000000.cfg |
02:16.56 | jmcdowell | to loadup all the phone... What an ass am i. |
02:17.36 | jaytee | jmcdowell, no you're not, your just learning a difficult but powerful method of setting up Polycoms |
02:17.43 | jmcdowell | Other than the password field, which may show up after I re-expand the archive.. This should be about licked |
02:17.55 | jaytee | that will give you more control in the long run than using the web gui and save lots of time |
02:20.24 | jmcdowell | http://pastebin.ca/1786642 |
02:20.29 | jmcdowell | is that what you are talking about? |
02:20.35 | *** join/#asterisk maxagaz (~maxagaz@soho2.i-xanadu.com) |
02:20.39 | etfonhomey | jmcdowell, jaytee is right. It took me quite a few tries to figure out the provisioning setup for Polycoms. |
02:21.12 | *** part/#asterisk mattwj2002 (~Matt@wikisource/pdpc.active.mattwj2002) |
02:21.12 | etfonhomey | jmcdowell, wait 'til a <mac>-phone.cfg overrides file gets in the mix to throw you for a loop. |
02:21.41 | jaytee | that's when I do a Format File System on the phone |
02:21.45 | jmcdowell | but is what I pasted in pastebin the correct format for a 00112233445566.cfg ? |
02:22.52 | etfonhomey | jmcdowell, why do you reference sip.cfg and sip_316.profiletech.cfg? |
02:23.15 | jmcdowell | I just didn't see them to take them out |
02:23.19 | *** join/#asterisk fofware (~chatzilla@host171.190-30-113.telecom.net.ar) |
02:23.23 | jmcdowell | I have been messing with this for a LONG time. |
02:23.24 | jmcdowell | :D |
02:23.36 | etfonhomey | jmcdowell, I would simplify it. |
02:24.05 | jmcdowell | So the first line, where it calls out configs, these are the MAIN configs, and the second time are the overriding options per phone, is that correct? |
02:25.11 | jaytee | jmcdowell, here's my master config file for my phone at work. http://pastebin.ca/1786647 |
02:25.59 | etfonhomey | jmcdowell, here's mine: http://pastebin.ca/1786649 I have everything in the same directory. I probably wouldn't if I had more than just a few phones. |
02:26.03 | *** join/#asterisk andresmujica (~andresmuj@ubuntu/member/andresmujica) |
02:26.28 | jmcdowell | So you are calling them out only once.. |
02:26.29 | jmcdowell | NICE.. |
02:26.31 | jmcdowell | hang on |
02:26.41 | jaytee | jmcdowell, I don't have a second line of "overrides" configs |
02:27.38 | etfonhomey | jmcdowell, I would just try to get only one phone working first and then go from there. |
02:27.50 | jmcdowell | That's what I am doing.. |
02:27.53 | etfonhomey | k |
02:27.56 | jmcdowell | Why are you calling sip.cfg ? |
02:28.02 | jmcdowell | Is that your "Master" config? |
02:28.08 | etfonhomey | jmcdowell, yes. |
02:28.13 | jmcdowell | The precvious is your "overrides" ? |
02:28.16 | jaytee | All phones need to use these files so they must be listed in the CONFIG_FILES |
02:28.16 | jaytee | list. For example, the master configuration file for the phone with Ethernet |
02:28.16 | jaytee | address of 0004f2000607 is called 0004f2000607.cfg and looks like this: |
02:28.48 | jaytee | jmcdowell, the following is VERY important to understand |
02:28.50 | jaytee | Since the files are processed left to right, any parameter which appears in |
02:28.51 | jaytee | 0004f2000607-user.cfg will override the same parameter in phone1.cfg. |
02:28.51 | jaytee | Similarly any parameter in local-settings.cfg will override the same parameter |
02:28.51 | jaytee | in sip.cfg. |
02:28.57 | carrar | I wouldn't change the sip.ld |
02:29.11 | carrar | the phone will know what file to get |
02:29.12 | jaytee | i use the one for my specific model |
02:29.23 | etfonhomey | jmcdowell, what did you mean by "previous is your overrides"? |
02:29.50 | carrar | 601 will always ask for 345-11605-001.sip.ld |
02:29.51 | jmcdowell | Meaning, phone specific settings. |
02:29.54 | carrar | 2345-11605-001.sip.ld |
02:30.00 | carrar | and 2345-11605-001.bootrom.ld |
02:30.07 | jaytee | this is the line in my master config file for my phone that loads the appropriate sip.ld file <APPLICATION APP_FILE_PATH="2345-12200-001.sip.ld" |
02:30.19 | jmcdowell | Username, password, extension, display name etc. |
02:30.23 | carrar | no |
02:30.24 | jaytee | that is for the Polycom330 with SIP firmware 2.2 |
02:30.30 | carrar | APP_FILE_PATH="sip.ld" |
02:30.36 | etfonhomey | jmcdowell, phoneX.cfg is where you configure the line appearances for each phone. |
02:30.48 | carrar | phone will automaticall add 2345-11605-001. to the front |
02:30.54 | carrar | if sip.ld doesn't exist |
02:31.06 | carrar | 2345-11605-001. is specific to the model of the phone |
02:31.07 | jmcdowell | what? |
02:31.14 | jmcdowell | And I thought I was really starting to understand. |
02:31.18 | carrar | heh |
02:31.33 | carrar | it's in the zip file you got from polycom |
02:31.36 | etfonhomey | jmcdowell, don't worry too much about the sip.ld and bootrom.ld too much. |
02:31.44 | jmcdowell | Man, I was really getting excited, I am about to throw these things in the trash. |
02:31.48 | etfonhomey | jmcdowell, at least for now. |
02:32.02 | jmcdowell | I already upgraded the firmware and the bootrom on this phone to the max I could go up to. |
02:32.19 | jmcdowell | With the firware came a whole slew of config files etc. |
02:32.27 | jaytee | jmcdowell, the sip.ld file contains config info for ALL phones. 2345-11605-001.ld is for only the model 601 so it is smaller, uses less memory storage and downloads quicker and loads faster |
02:32.29 | carrar | You need to start with those |
02:32.32 | etfonhomey | jmcdowell, then don't worry about it. Just make sure you're using the sip.cfg/phone.cfg from the zip file that you got the firmware from. |
02:32.47 | jmcdowell | k |
02:33.10 | carrar | quicker, faster, better and CHICS DIG IT |
02:33.44 | etfonhomey | jmcdowell, you can get away with not touching the sip.cfg file for your first "hello world" setup. |
02:33.55 | carrar | if I had nothing to do I could write a nice long wiki on how I setup polycoms |
02:34.06 | etfonhomey | jmcdowell, you can config everything you need for a basic line registration in the phonex.cfg. |
02:34.22 | jmcdowell | ok hang on |
02:34.29 | jmcdowell | I am trying to boot this things.. |
02:34.30 | jmcdowell | thing |
02:35.17 | jmcdowell | I can bring 1000+ node clusters back from the dead, and birth new ones.. |
02:35.19 | *** join/#asterisk jakent (~john@soleil.johnkent.mooo.com) |
02:35.24 | jmcdowell | but I can't setup a polycom./. |
02:35.26 | jmcdowell | :/ |
02:35.37 | carrar | and you learned clustering over night? |
02:35.55 | rnp | To All: if I use asterisk/VICI dial in conjunction with my online crm system, is there anyway I can have my clients login to my crm, and see for instance the list of businesses we are calling for their account and what their responses were? |
02:36.38 | jaytee | jmcdowell, for comparison purposes here's my "MAC"-phone.cfg file which is actually named 0004f21a0c98-5146.cfg http://pastebin.ca/1786661 |
02:37.01 | jmcdowell | clearly I did not |
02:37.06 | jaytee | my extension is 5146 and my sip account in sip.conf for asterisk is [5146] |
02:37.26 | etfonhomey | jmcdowell, here is a basic config snippet from my phoneX.cfg with the corresponding sip.conf entry: http://pastebin.ca/1786660 |
02:37.49 | etfonhomey | That is all I need in order to be able to register "station1" to Asterisk. |
02:38.26 | jmcdowell | cool |
02:38.26 | carrar | I actually use the whole sip & phone1 file that is sent from polycom |
02:38.31 | jmcdowell | almost there |
02:38.38 | carrar | with all my changes |
02:39.07 | rnp | does anyone here consult/setup new pbx asterisk based systems? |
02:39.15 | carrar | everyone does |
02:39.25 | rnp | lol |
02:39.36 | carrar | ask away |
02:39.51 | rnp | well I want to talk to someone priv about what I want and what it will cost to do |
02:39.54 | rnp | basically |
02:40.13 | Sedorox | you might want to state where you are |
02:40.23 | Kobaz | anyone know how i make a t1 card bind to a specific cpu |
02:40.24 | rnp | USA |
02:40.35 | Sedorox | state? |
02:40.36 | Kobaz | i have a sangoma card that randomly flips it's inturrupts from cpu0 to cpu1 |
02:40.47 | rnp | I run a remote operation in the philippines and I want to get my own system setup so I control all the calls they make |
02:40.54 | rnp | Florida, I don't need on site help |
02:40.59 | Sedorox | ah |
02:41.10 | carrar | isn't voip in the philippines to the outside illegal? |
02:41.18 | rnp | no |
02:41.38 | Kobaz | www.kobaz.net/misc/t1_port1_device_interupts-day.png |
02:42.09 | carrar | rnp, should be easy |
02:42.11 | Kobaz | green is cpu0 and blue is cpu1 |
02:42.23 | rnp | yeah, but there are multiple things that need to be integrated with my online crm |
02:42.24 | etfonhomey | carrar, I use them all as well, just trying to give jmcdowell a "hello world" setup. |
02:42.29 | rnp | that's what is most important to me |
02:42.50 | carrar | bbl, dinner |
02:43.00 | jmcdowell | Config file rror is 0x20 |
02:43.14 | rnp | Sedorox, do you do consulting work for this stuff? |
02:43.58 | Sedorox | for asterisk, rarely, especially what your looking for |
02:44.20 | rnp | i see |
02:44.40 | Sedorox | I'm more of an end user of asterisk, or can do basic setups |
02:44.41 | jaytee | The most common reported reason for that error, an unescaped ampersand somewhere in the file |
02:44.54 | *** join/#asterisk NicoleMun (~ssharma@pool-173-63-185-74.nwrknj.fios.verizon.net) |
02:46.03 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
02:46.34 | jmcdowell | Call.. VMI |
02:46.45 | jmcdowell | That's all I can tell ya, but they aren't all open source. |
02:47.00 | jmcdowell | Voice mail incorp |
02:48.54 | jmcdowell | Ahhh! |
02:49.07 | jmcdowell | I figured out the error, got the phone to boot, and it still won't dial |
02:49.24 | jaytee | jmcdowell, is it registering with asterisk? |
02:54.06 | jmcdowell | And I thought I was close.. |
02:55.01 | jaytee | maybe you are close |
02:55.06 | jaytee | what are you trying to dial? |
03:09.20 | *** join/#asterisk rossand (~aross@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
03:19.47 | *** part/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
03:20.07 | *** join/#asterisk jblack (~jblack@71.181.248.16) |
03:21.14 | jmcdowell | rolls his eyes.. |
03:21.26 | jmcdowell | the damn thing hasn't been getting anything this WHOLE time.. |
03:21.36 | jmcdowell | It downloads it, but it doesn't use it. |
03:27.16 | *** join/#asterisk Tech_Travis (~Administr@cpe-76-168-191-127.socal.res.rr.com) |
03:30.55 | *** join/#asterisk Torrieri (~Torrieri@nelug/crew/torrieri) |
03:31.29 | jmcdowell | throws 1 out of 14 polycoms against the wall. |
03:31.30 | jmcdowell | next.. |
03:32.04 | ChannelZ | perhaps there is a syntax error somewhere that causes it to reject the whole thing? |
03:34.17 | jmcdowell | Could be but this is from scratch again with the files from the sip firmware upgrade. |
03:34.47 | jmcdowell | Wait it is taking some part of it, as I see 3 line keys now |
03:35.02 | jmcdowell | but the phone name is not being taken, the register bits are not being taken. |
03:36.25 | jmcdowell | when i go into the phone settings on the phone from the phone the sip server is blank |
04:00.16 | *** join/#asterisk OrNix (~ornix@l151-249-47.static.cn.ru) |
04:03.34 | *** join/#asterisk jmcdowell (~nooe@173.154.185.70) |
04:03.58 | jmcdowell | What is a channel troll? |
04:05.00 | p3nguin | It's a person who goes in the channel looking to stir up trouble, usually by asking really ridiculous questions that don't really make sense and are completely off topic. |
04:05.26 | *** part/#asterisk hluesea (~hulusikah@88.247.127.66) |
04:05.42 | jmcdowell | ahhh |
04:05.44 | jmcdowell | well.. |
04:05.49 | russellb | like ... hey, i heard asterisk is so bad, it puts voice under the IP |
04:06.05 | jmcdowell | lol |
04:06.07 | p3nguin | Like a common one we get on a Linux channel is a guy saying he needs to uninstall Linux because he needs to use Windows again... |
04:06.11 | jblack | Dont' run it on a cpu bound system. :P |
04:06.59 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
04:07.50 | jmcdowell | ok so this damn phone isn't even trying to register.. |
04:07.54 | jblack | i.e. "cheap bastards will pay the price of being cheap". =) |
04:08.00 | jmcdowell | lol |
04:08.06 | p3nguin | He is told that you just delete the partition(s) where it is installed, and then install Windows again. Then he continues to ask for help installing Windows, which is not a Linux channel topic. When he can't get help, he keeps bitching that he needs to uninstall Linux. |
04:08.31 | jmcdowell | It's getting an ip taking the config file and labeling the lines.. |
04:08.34 | jmcdowell | but that's it.. |
04:08.45 | jmcdowell | By the way, my juicer is broken.. |
04:08.47 | jmcdowell | ;) |
04:11.34 | p3nguin | jblack: Speaking of trolls, did you know that we still get the "poop guy" every few days? |
04:12.12 | jblack | that sounds familiar. |
04:12.22 | *** join/#asterisk Micc (~quassel@c-98-225-57-96.hsd1.wa.comcast.net) |
04:12.29 | jblack | That's an actual person, right? |
04:12.37 | p3nguin | After all this time, he still comes around. |
04:12.46 | jblack | He's been doing that for what? 5 years now? |
04:12.59 | p3nguin | Yep, he sometimes will interact with people if he doesn't get kicked out promptly. |
04:13.13 | p3nguin | Probably around 5 that I can account for. |
04:13.14 | jmcdowell | Why not just ban him for good? |
04:13.20 | jblack | He must have it in his google calendar, right after "bathe" but right before "take pills" |
04:13.50 | jblack | Because he was harmless. he wanted to spend a moment each day talking about poop. |
04:14.07 | jmcdowell | Ah hah!!! |
04:14.12 | jblack | other than that 90 seconds each day, he never causes any trouble |
04:14.22 | jmcdowell | - No matching peer found |
04:14.30 | jmcdowell | It's trying now and getting rejected! |
04:14.34 | p3nguin | When you have a reasonable supply of dynamic IP addresses and can choose nearly any nick name you want, it is easier for you to evade permabans. |
04:15.24 | jmcdowell | Yeah, I could I suppose, mine changes EVERY time I reboot my router and they are way off the charts, not usually anything near the last. |
04:15.39 | jblack | Yeah. An I'm a jijitsu op. I believe pushing straight back is pointless. It's better to let people use their own inertia to get themselves out of the way. |
04:16.08 | jmcdowell | is there anyway to make asterisk spit out what the phone is trying to log in as? |
04:16.19 | jblack | you can do a sip debug and watch the packets |
04:16.23 | p3nguin | How often does a router really need rebooted? |
04:16.49 | p3nguin | <PROTECTED> |
04:16.53 | p3nguin | There's my router. |
04:17.06 | jmcdowell | Mine doesn't usually, but my aircard dies randomly |
04:17.12 | jmcdowell | I think they are acutally kicking me off |
04:17.22 | jmcdowell | because I use it and abuse it for what it's worth. |
04:18.39 | jblack | peng: o you do facebook? |
04:19.23 | jmcdowell | sip debug is showing everything but the failed log ins |
04:19.43 | *** join/#asterisk ReDNeQ (~ReDNeQ@70.114.229.58) |
04:20.32 | jmcdowell | it's getting it |
04:20.37 | jmcdowell | but I don't see the user name |
04:21.17 | jmcdowell | Ooooohhh.. |
04:22.39 | p3nguin | jblack: Negative, I don't do social networking. |
04:23.15 | jmcdowell | I think I nailed it |
04:23.33 | p3nguin | Not twitter, facebook, myspace |
04:23.38 | jmcdowell | I had put the pbx address in the field that it apparently and strangely uses as the user ID |
04:23.49 | p3nguin | Don't know what else there is, since I don't do those either. |
04:24.31 | jmcdowell | Sorry, I am thinking out loud |
04:24.44 | jmcdowell | in case anyone wants to correct mel;. |
04:24.45 | p3nguin | don't worry about that. |
04:24.46 | jmcdowell | me |
04:24.47 | jmcdowell | :DS |
04:25.02 | p3nguin | IRC is like the friend you don't have. |
04:25.19 | jmcdowell | so the reg string was pbxaddress.com@pbxaddress.com |
04:25.22 | jmcdowell | would won't work |
04:25.23 | p3nguin | You can talk to it all you want, and if it feels like talking back, it will. |
04:25.29 | jmcdowell | lol |
04:25.34 | *** join/#asterisk Carlos_PHX (~Carlos@ip68-99-199-10.ph.ph.cox.net) |
04:26.12 | jmcdowell | starts doing the funky chicken dane.. |
04:26.14 | jmcdowell | dance |
04:26.23 | jmcdowell | It's working.. Sorta.. Now I have dial plan problems.. |
04:26.51 | jmcdowell | Anyone in here care to help? It's stripping the 9 off when it send, which causes asterisk to reject the call. |
04:27.27 | p3nguin | Show me the dialplan, what you are dialing on the phone, and what you want to happen. |
04:28.09 | jmcdowell | <digitmap dialplan.digitmap="[2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2-9]xxxT" dialplan.digitmap.timeOut="3|3|3|3|3|3"/> |
04:28.12 | jmcdowell | is the current plan |
04:28.28 | dlynes_laptop | Anyone know what 'Unable to allocate AST channel structure for SIP channel' means? |
04:28.32 | dlynes_laptop | pastebin at: http://pastebin.com/m3ab852db |
04:28.33 | jmcdowell | when I dial 913143212222 it strips the 9 and send 13143212222 |
04:28.41 | dlynes_laptop | I'm guessing it means an out of memory condition? |
04:29.03 | russellb | dlynes_laptop: you're likely hitting an open file descriptor limit |
04:29.07 | russellb | ulimit |
04:29.28 | russellb | chan_sip is trying to allocate an Asterisk channel object, and it fails because it can't open a pipe |
04:29.36 | russellb | most common cause is you hit the open fd limit. |
04:29.54 | p3nguin | 314, as in MO, USA? |
04:29.56 | russellb | and so then it goes, :'-( |
04:30.07 | jmcdowell | Yes |
04:30.08 | jmcdowell | as in MO |
04:30.24 | p3nguin | I'm about 70 miles from STL. |
04:30.30 | dlynes_laptop | russellb, and those would be listed in 'netstat -anp | grep "^unix"'? |
04:30.33 | jmcdowell | I would like to not have to dial the 1 and not have the 9 strip the 9 off.. |
04:30.40 | jmcdowell | What direction? |
04:30.44 | p3nguin | east |
04:30.45 | russellb | dlynes_laptop: i dunno |
04:30.48 | jmcdowell | IL |
04:30.51 | p3nguin | yep |
04:31.01 | dlynes_laptop | russellb, and fwiw, my ulimit is unlimited |
04:31.01 | jmcdowell | We'll have to have a nerd party some time.. |
04:31.02 | jmcdowell | :D |
04:31.02 | p3nguin | near Mt. Vernon |
04:31.07 | russellb | dlynes_laptop: orly! |
04:31.11 | jmcdowell | I am vaugley familiar with IL |
04:31.12 | russellb | well then. |
04:31.12 | dlynes_laptop | russellb, really |
04:31.30 | russellb | any hung channels? core show channels, sip show channels ... |
04:31.40 | dlynes_laptop | russellb, don't know...I couldn't connect |
04:31.47 | dlynes_laptop | russellb, i had to kill the pid |
04:31.58 | russellb | huh. it explodified. |
04:32.04 | russellb | i suggest a hammer |
04:32.05 | dlynes_laptop | russellb, root 1913 3.2 12.2 142108 125848 ? Ssl Jan19 706:49 /usr/sbin/asterisk |
04:32.15 | dlynes_laptop | russellb, using 'ps auxffww' |
04:32.17 | russellb | and that's the best I can suggest in my near unconcious state, heh |
04:32.36 | p3nguin | jmcdowell: But as for the dial plan... what extension does asterisk think you are dialing? |
04:32.46 | jmcdowell | 9 to get out |
04:32.51 | dlynes_laptop | russellb, if it makes any difference, it's asterisk 1.6.1.8 |
04:32.53 | jmcdowell | and the extensions are all 1xx |
04:33.04 | jmcdowell | the extensions can change if they need to |
04:33.29 | russellb | dlynes_laptop: hmm, lots of fixes since then, even one directly related to a problem that led to symptoms like this |
04:33.29 | p3nguin | But I need to know what * thinks you are dialing, because your phone's dialplan looks normal to me. |
04:33.39 | jmcdowell | Uhhh |
04:33.40 | russellb | you should consider an upgrade, I suppose. |
04:33.40 | jmcdowell | hang on |
04:33.47 | russellb | passes out ... gnight |
04:33.47 | dlynes_laptop | russellb, ok, thanks |
04:33.49 | russellb | np |
04:33.52 | dlynes_laptop | russellb, g'night |
04:34.25 | jmcdowell | The outbound route is 9|. |
04:34.31 | jmcdowell | I know stop the laughter |
04:34.39 | jmcdowell | i don't understand these dial plans |
04:34.39 | p3nguin | I don't even know what an "outbound route" is. |
04:34.42 | p3nguin | Sounds made up. |
04:34.51 | p3nguin | Must be an unsupported FreePBX. |
04:35.02 | jmcdowell | No.. |
04:35.06 | jmcdowell | The outbound route |
04:35.21 | jmcdowell | It's the route that gets to the SIP provider |
04:35.27 | jmcdowell | and 9|. picks it up |
04:35.44 | p3nguin | Right now, the phone is accepting 4-digit extens starting with 2-9. Also 10 and 11 digit numbers. |
04:35.45 | jmcdowell | of anything matches 9|. it goes to the sip provider |
04:35.53 | p3nguin | According to whom? |
04:36.01 | jmcdowell | hang on |
04:36.04 | jmcdowell | let me find something. |
04:36.23 | p3nguin | The phone certainly isn't doing that in its dialplan. |
04:36.43 | jmcdowell | http://blogs.elastix.org/en/wp-content/uploads/2009/11/melbviamelbpbx.png |
04:36.49 | jmcdowell | that is the screen I see |
04:37.08 | p3nguin | Dial 13143212222 or 3143212222 and it'll match the dialplan on the phone. |
04:37.14 | jmcdowell | That's the problem, I am dialign 12 digits.. |
04:37.17 | p3nguin | stop |
04:37.31 | jmcdowell | if I dial 1314xxxxxxxx the outbound route will not pick it up. |
04:37.35 | p3nguin | Dial it correctly. 10 or 11 digits. |
04:37.51 | p3nguin | The "outbound route" will do whatever you tell it to do. |
04:38.00 | jmcdowell | Your call cannot be completed as dialed |
04:38.03 | jmcdowell | * is telling me that |
04:38.12 | p3nguin | Then you haven't configured your extens correctly. |
04:38.18 | *** join/#asterisk pawz (~pawz@ppp118-208-82-201.lns20.bne4.internode.on.net) |
04:38.40 | jmcdowell | if I changed the outbound route to accept 1|. it would fly |
04:39.13 | jmcdowell | i can dial extension to extension |
04:39.43 | p3nguin | You mean device to device. |
04:39.53 | jmcdowell | Yes |
04:40.10 | p3nguin | exten => _1NXXNXXXXX,1,Dial(SIP/${EXTEN}@yourprovidename) |
04:40.17 | p3nguin | That's for 11 digit dial. |
04:40.21 | jmcdowell | So if I change the dial plan in the trunk to 1|. it gets to my provider but fails to dial because it gets the one stripped off. |
04:40.24 | p3nguin | exten => _NXXNXXXXX,1,Dial(SIP/1${EXTEN}@yourprovidename) |
04:40.30 | p3nguin | There's 10 digit dial. |
04:40.44 | dlynes_laptop | russellb, btw...if you're still here...it also resulted in SIP 500's and SIP 403's being sent back to any sip users that were attempting to send calls to the box |
04:41.12 | p3nguin | Want 7 digit dial? exten => _NXXXXXX,1,Dial(SIP/1314${EXTEN}@yourprovidename) |
04:41.24 | p3nguin | I think I missed an X in the first two. |
04:42.09 | p3nguin | jmcdowell: You really should stop playing with the dialplan on the phone and accept that it was fine the way you showed it to me. |
04:42.21 | jmcdowell | Then I have to figure out why I can't get out. |
04:42.25 | jmcdowell | hang on |
04:42.26 | p3nguin | It is asterisk's dialplan that needs work. |
04:42.35 | p3nguin | extensions.conf |
04:42.52 | p3nguin | The phone dialplan you showed earlier looked fine to me. Leave it alone. |
04:42.59 | jmcdowell | Ok......... |
04:43.02 | jmcdowell | hmmmmm |
04:43.08 | dlynes_laptop | heh...gotta love cunningpike's quit message :) |
04:43.19 | ManxPower-work | jmcdowell: dial, then press the DIAL button, that will bypass the phone dialplan by default |
04:43.41 | ManxPower-work | i.e. dial digits first |
04:43.52 | p3nguin | But the phone's dialplan is fine, so what's the use? |
04:43.55 | jmcdowell | I figured it out.. |
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04:44.01 | jmcdowell | I have to change the dial plan in * |
04:44.08 | p3nguin | No kidding? |
04:44.08 | jmcdowell | 1NXXNXXXXXX |
04:44.10 | jmcdowell | NXXNXXXXXX |
04:44.10 | jmcdowell | NXXXXXX |
04:44.15 | p3nguin | Do you read what I type? |
04:44.16 | dlynes_laptop | jmcdowell, isn't that what p3nguin just finished telling you? |
04:44.28 | ManxPower-work | jmcdowell: A unified, designed, well thought out dialplan will save you countless hours. |
04:44.31 | jmcdowell | I was wacking @ this problem.. |
04:44.36 | jmcdowell | But yes that is what he basically said |
04:44.39 | Corydon76-dig | jmcdowell: you're missing the leading underscore |
04:44.58 | p3nguin | I did typo my patterns, though, Missed one X in the first two. |
04:45.05 | jmcdowell | Hmmmm |
04:45.16 | ManxPower-work | sings "Sqlite3, and cookies, and PHP, and Polycom, Oh my!" |
04:45.26 | jmcdowell | The last thing I have left to grasp is the "lines" in the config.. |
04:45.27 | dlynes_laptop | cringes. |
04:45.34 | Corydon76-dig | If it doesn't start with an underscore, it ISN'T a pattern |
04:45.35 | jmcdowell | I have a 3 line setup from my sip provider... |
04:45.44 | voipmonk | How does that tune go? |
04:46.03 | jmcdowell | Thanks for all the help though, it has been very helpful.. |
04:46.07 | ManxPower-work | Lions, tigers, and Republicans, oh my! |
04:46.08 | jmcdowell | I am still astarded... |
04:46.14 | jmcdowell | I am still astertarded... |
04:46.38 | p3nguin | If you'll just pay more attention here it'll be much easier to receive help HERE. |
04:46.47 | ManxPower-work | jmcdowell: this isn't something easy like designing a national data network -- telecom is hard. |
04:47.04 | jmcdowell | I feel it. |
04:47.58 | dlynes_laptop | It's definitely more difficult than writing VB code |
04:49.38 | jmcdowell | lol |
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05:05.17 | etfonhomey | jmcdowell, did you finally get that phone registered to Asterisk? |
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05:10.08 | jmcdowell | yes |
05:10.15 | jmcdowell | now I am trying to learn the dialing patterns |
05:10.29 | jmcdowell | not the command line dialing patterns, the freepbx dial patterns. |
05:10.41 | p3nguin | ~freepbx |
05:10.41 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
05:11.11 | p3nguin | Do it in vanilla asterisk, and I'll help. |
05:11.39 | Micc | is there a way to not have the h extension jumped to after calling Hangup? |
05:11.51 | Micc | Like, should I use Return instead of Hangup? |
05:12.24 | Micc | I'm trying to do a .call file and if it fails and jumps to failed, I don't want it to goto h after that. |
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05:19.09 | ManxPower-work | Micc: you don't have much choice. |
05:19.42 | ManxPower-work | You can try Return, but I would not expect much. |
05:23.32 | Micc | How can I jump over a bunch of stuff in h? I guess I could do a gotoif |
05:25.01 | dlynes_laptop | p3nguin, he's trying to make it easier than vb for himself, but more difficult than acupuncture for you |
05:25.35 | p3nguin | dialplan, generally speaking, is very easy. |
05:27.24 | p3nguin | It's those quirky things like I am dealing with right now that make me yell and scream. Using a call file, connect to a channel, then run an extension... auto-outbound,s,1 tries to run and fails twice. |
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05:28.52 | p3nguin | Ah, I think I found the error. The context is auto-outbound, but it is auto-outgoing in the call file! |
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05:32.29 | p3nguin | Yep, that solved the issue. |
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05:41.34 | ZxCv47 | anyone know where i can find info on the different error messages produced by fax for asterisk? |
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05:47.51 | sbrath | this is probably a dumb question, but I have a call coming in on a DAHDI trunk, and in the CDR logs, I have the Caller-id-name, but when I do a dumpChan on the call caller-id-name is not set, |
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05:54.13 | jmcdowell | Hrrrmmm.. |
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05:58.09 | sbrath | I see: CallerIDName= (N/A) |
05:58.16 | sbrath | in my DumpChan(10) |
05:59.01 | sbrath | this isn't a complicated setup, just a PRI coming in from the telco that's delivering the Cid-name. Is their a DAHDI paramater that I need to let that name flow thru? |
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06:02.59 | brunner | Is there any hardware that I can plug into a phone jack in my home that has enough current to power the rest of my home phones? |
06:04.03 | brunner | could I just run a telephone cable from any FXS port to a telephone jack to do that? |
06:04.17 | brunner | (assuming AT&T isn't connected at the demarcation point) |
06:06.29 | sbrath | I guess as long as the demarc to at&t is disconnected, it should work. |
06:06.48 | sbrath | I'm not sure how many phones it will power tho, do you have any FXS card now? |
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06:20.08 | rnp | anyone on now that does asterisk implementation? |
06:24.04 | florz | anyone on now that does specific questions? |
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06:29.59 | p3nguin | ha |
06:30.08 | ChannelZ | no |
06:32.13 | rnp | lol |
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06:35.23 | kaldemar | sbrath: do you have usecallerid=yes and callerid=asreceived in chan_dahdi.conf? feel free to pastebin the cli output of a call. |
06:37.21 | rnp | Do you think 4k is a lot to get asterisk setup with custom integration with an online crm? |
06:37.56 | carrar | You better be supplying the hardware for that |
06:38.29 | rnp | really |
06:39.11 | rnp | carrar do you setup astrisk for others? |
06:39.15 | carrar | yeah |
06:39.45 | rnp | alright, let me hit you with an im maybe you can help |
06:39.59 | carrar | I'm not interested however, but thanks |
06:40.26 | rnp | oh, alright |
06:40.48 | carrar | What CRM do you want it to use? |
06:40.55 | carrar | and how |
06:41.00 | rnp | a custom one I had made from scratch |
06:41.06 | rnp | uses mysql though |
06:41.16 | carrar | So you would be writting that part? |
06:42.08 | carrar | You need to write out a proposal of exactly what functionality to exist in the server |
06:42.25 | carrar | otherwise if you expected call parking and didn't get it, well you need to write it out |
06:42.38 | carrar | as a example |
06:43.22 | rnp | ok |
06:43.30 | rnp | let me outline what is needed |
06:43.42 | rnp | I'm looking for an app that runs on top of Asterisk that will query CRM for a client account to work on which includes the following: |
06:43.48 | rnp | 1. A set of numbers (with names) to call. |
06:43.54 | carrar | What is needed from the phones perspective, PBX and what you want the PBX to do with the CRM |
06:43.54 | rnp | The number of appointments to achieve for the client |
06:44.24 | rnp | I want these numbers (1 above) to be dialed automatically for call agents residing in the Phillipines. Then, at the termination of each call, the agent should be able to specify if a successful appointment was achieved (a yes or no). You would need the app to post back to the CRM the result of the call and then start the process again with a new call. |
06:45.55 | carrar | might be better to build a web app around that |
06:46.16 | rnp | what kind of cost am I looking at to get something like this setup |
06:46.36 | carrar | agent might be walking away to the bathroom and you got this automatic dialer going crazy |
06:47.15 | carrar | depends how you have to get the numbers from mysql |
06:47.44 | carrar | but that alone is a customized app |
06:47.53 | carrar | not even including you need the PBX part too |
06:48.11 | rnp | well I said 4k and you said better be including the hardware for that too |
06:48.22 | rnp | so what kind of price do you think is real for what I want |
06:48.57 | carrar | thats one 1 part |
06:49.02 | carrar | You need a complete request |
06:49.07 | sbrath | kaldemar: my config for the dahdi is split between users.conf and dahdi-channels.conf |
06:49.18 | carrar | You might want something in the system that isn't customized |
06:49.23 | carrar | err that is |
06:49.36 | rnp | I see |
06:49.51 | carrar | You are writting a RFP |
06:50.28 | rnp | what if the business list was loaded onto asterisk, and then only appointments were transmitted to the crm ? |
06:50.31 | rnp | is that a lot easier? |
06:50.48 | carrar | but what you describe is certainly doable |
06:51.04 | rnp | well carrar, what I describe, what do you think it should cost $$ ? |
06:51.06 | kaldemar | sbrath: dahdi-channels.conf is something that chan_dahdi.conf includes into itself when the configuration script is used. the PRI should be defined in dahdi-channels.conf then. |
06:51.16 | carrar | I think you are leaving a lot out |
06:51.24 | carrar | and again you need to write a complete RFP |
06:51.34 | carrar | if it's not written out you don't get it |
06:51.36 | nix8n82 | depends on who you ask and how many hours of custom programming you want |
06:52.09 | sbrath | kaldemar: it is also configured in dahdi-channels.conf I don't have the usecallerid=yes in there, that is listed under the span config in users.conf, but I'll add it to dahdi-channels. |
06:52.11 | rnp | what's a reasonable $ per hour for this kind of programming? |
06:52.32 | carrar | depends who you ask |
06:52.39 | carrar | 80-$250/hr |
06:52.43 | rnp | shew |
06:52.45 | carrar | and how good they are |
06:53.08 | carrar | maybe less |
06:53.11 | nix8n82 | do you have the hardware? |
06:53.13 | rnp | based on the limited info i've given you, how many hours do you think this would take? |
06:53.29 | carrar | based on the limited info I can't say really |
06:53.34 | rnp | no hardware, was just going to get a server on slicehost.com |
06:53.45 | kaldemar | sbrath: don't mix users.conf and chan_dahdi.conf. actually, don't use users.conf at all for the PRI. |
06:54.43 | sbrath | kaldemar: I'm trying to get out of users.conf, but shouldn't the settings work the same? |
06:54.52 | carrar | JUST for that single application of dialing numbers by polling a table in mysql you looking at a day |
06:55.08 | carrar | at tops |
06:55.23 | carrar | but I suspect you want more out of that app |
06:55.27 | kaldemar | sbrath: you don't have the same settings in users.conf and chan_dahdi.conf, and users.conf is not meant for configuring trunks. |
06:56.06 | kaldemar | sbrath: do yourself a favor and configure the PRI in dahdi-channels.conf only. |
06:56.08 | rnp | well one concern i have is that once we set the appropriate number of appointments for 1 given area then we need to stop dialing and move on toa new area |
06:56.39 | carrar | if you don't completely spec out what you want someone to write you aren't gonna get what you asked for |
06:56.52 | carrar | repeat, repeat , repeat |
06:56.57 | rnp | I understand, spec it out |
06:57.11 | rnp | so what do you do carrar? |
06:57.19 | carrar | everything |
06:57.23 | rnp | big IT honcho position |
06:58.49 | sbrath | kaldemar: I'm pulling it from users.conf, but even thou I have group=2 in dahdi-trunks for the 2nd pri, both pri's are ending up in group 1?? |
06:59.09 | kaldemar | sbrath: pastebin it |
07:00.02 | sbrath | http://pastebin.com/m57f36fe5 |
07:01.15 | kaldemar | sbrath: group definitions must be above channel lines. now you have them both below. |
07:01.22 | sbrath | ok |
07:01.41 | carrar | rnp, 20 years of unix and programing, 15 years of network engineering, 9 years of asterisk |
07:02.02 | kaldemar | sbrath: all parameters apply for channel lines below them, until otherwise defined. |
07:02.28 | sbrath | kaldemar: very tricky.... |
07:02.38 | sbrath | ok, I have 2 groups now. |
07:03.04 | sbrath | still no caller-id-name |
07:03.06 | kaldemar | sbrath: you have a typo here: "callerid=asrecieved" |
07:03.33 | carrar | 8 years of asterisk, sorry |
07:03.40 | carrar | 8-9 |
07:03.43 | carrar | somewhere in there |
07:03.45 | carrar | heh |
07:03.47 | kaldemar | sbrath: and the caller id parameters were below "channel => 25-47" |
07:04.08 | kaldemar | carrar: it's the ninth year that makes the difference. :) |
07:04.13 | carrar | haha |
07:04.23 | carrar | gah, missed it by a hour |
07:04.29 | sbrath | so does every paramater in dahdi-channels then behave that way, they all colelct up, and then apply to channels when the channel line hits? |
07:04.39 | kaldemar | sbrath: yes |
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07:06.15 | sbrath | well that would explain most of my dahdi issues then... :) |
07:06.20 | sbrath | let me try it again. |
07:07.09 | sbrath | dahdi restart should reload all that, or do I need to do another reload to purge the users.conf stuff. |
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07:09.30 | kaldemar | sbrath: i'd restart whole asterisk to be sure. :P |
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07:11.50 | sbrath | kaldemar: restarted whole server. |
07:12.18 | sbrath | kaldemar: just wondering, are other configs in asterisk order dependant ? |
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07:13.49 | sbrath | carrar: but the real question is how many programming languages do you know :) |
07:14.05 | kaldemar | sbrath: only dahdi, others follow the context structure. |
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07:14.39 | sbrath | kaldemar: so I'm restarted, and I still see callerid-name in the cdr logs, but still N/A on a DumpChan. |
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07:17.07 | carrar | 5 'programing langauges' |
07:17.14 | kaldemar | sbrath: feel free to pastebin a cli output of a call. add NoOp(${CALLERID(name)}) into the extension aswell. |
07:18.10 | carrar | I don't consider shell scripting, cgi, html, SQL etc..programming language |
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07:20.18 | sbrath | carrar: so C++, Smalltalk, Pascal, Fortran, MASM, etc are "Programming Languages" |
07:20.26 | sbrath | only 5 :) |
07:20.29 | carrar | haha |
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07:21.03 | carrar | I wish I knew C++ |
07:21.37 | florz | SQL is a programming language nontheless ;-) |
07:21.43 | carrar | then 6 |
07:21.52 | kaldemar | sbrath: what is the name in cdr? |
07:22.22 | florz | Well, if you don't consider SQL a programming language, chances are you don't know SQL ;-) |
07:22.36 | sbrath | ok, I don't have the 9 years, of asterisk, but I'm close on the rest of your numbers :) |
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07:23.43 | sbrath | If you want to call SQL a language, you need to know more than just Select * from users :) |
07:24.05 | florz | that's basically what I mean :-) |
07:25.22 | sbrath | if you can write me a full 20 line "program" in SQL using full ANSI syntax, and inner and outer joins in the same query.... Then you know SQL .. |
07:26.16 | sbrath | kaldemar: did you get my msg? |
07:26.48 | florz | and some joins with unions of aggregating subqueries, or somesuch, yeah |
07:26.50 | sbrath | carrar: Do you know any other OO languages? |
07:27.00 | carrar | no |
07:27.15 | carrar | no C++ |
07:27.25 | sbrath | we have one SQL query in a reporting app that has 34 pages of SQL!!!!! I wish I could slap the developer. |
07:27.38 | sbrath | carrar: any java or smalltalk? |
07:27.41 | carrar | no |
07:27.43 | florz | basically, when you do something that could reasonable be called computation, and not just selection |
07:27.52 | florz | erm, reasonably |
07:28.13 | florz | 34 pages? that' |
07:28.18 | florz | 34 pages? that's really a lot for SQL :-) |
07:28.36 | sbrath | Amazing the DB dosent eat itself when that runs... |
07:28.45 | florz | pretty much for any definition of "page" ;-) |
07:29.10 | carrar | with comments you can make a selec count take half a page |
07:29.16 | carrar | heh |
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07:29.23 | sbrath | It's like about 20 seperate queries for different business metrics data sets all unioned and minus's together, with a crap load of temp-tables... |
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07:30.05 | sbrath | 34pages of sql is un-forgivable, when you try to write something in SQL that belonged in Java or the calling app, the developer should be slapped just from the point of Maintenance. |
07:30.39 | florz | well, it really depends on the task, I guess |
07:31.07 | sbrath | I can code stored procedures, but when you spend 20 lines of SQL to work around the fact that it's not a "language" it makes the code hard to support when the 3rd person after you inherits it. |
07:31.43 | florz | after all, data aggregation and correlation often can be expressed far more succinctly in SQL |
07:31.51 | sbrath | Most of the 34 page SQL is more of a "Hey look how conveluded I can make a SQL to do everything you asked for but nobody else will ever figure this out unless they break it all down line-by-line " |
07:32.02 | sbrath | agreed. |
07:32.27 | sbrath | but back to my asterisk problem :) |
07:32.30 | florz | well, yeah, if you start writing loops in SQL despite heving the option to use an application language for that .. then probably something is wrong ;-) |
07:32.53 | sbrath | florz: which flavor of SQL do you do? Oracle/MySql/MSSQL ? |
07:33.03 | florz | Pg? =:-) |
07:33.09 | sbrath | postgres. |
07:33.14 | florz | yeah |
07:33.18 | florz | mostly, at least |
07:34.09 | sbrath | I've only dabbled at postgres, I had a telco switch once "Broadsoft" that was based in Postgres, and I had to tweak the DB, and do backups. but never "programmed" it... Seems like it's a step up from MySql with stored procedures... |
07:34.32 | sbrath | kaldemar: Did I loose you? |
07:35.12 | florz | it's ... like ... up a high mountain from MySQL? ;-) |
07:35.39 | sbrath | without all the "Oracle" ownership behind it :) |
07:35.57 | sbrath | at least for the InnoDB stuff. |
07:35.58 | florz | well, Oracle now does endorse Pg, too! |
07:36.19 | florz | http://postgresql.blogg.se/2010/february/sun-oracle-postgresql.html <- ! |
07:36.24 | sbrath | but Oracle Bought InnoDB engine that MySql uses for transactions |
07:36.47 | florz | yeah, and with Sun they also bought the endorsement of Pg ;-) |
07:37.08 | florz | but they don't "own" any of it, that's true |
07:38.40 | kaldemar | sbrath: strange. the name part must come from somewhere. use PRI debug on an incoming call to see where the information is. |
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07:42.05 | sbrath | I see this: Ext: 1 Progress Description: Inband information or appropriate pattern now available. (8) ] |
07:42.13 | sbrath | After the DumpChan |
07:43.12 | sbrath | and a few more lines lower this: < Facility (len=31, codeset=0) [ [Feb 4 01:40:33] VERBOSE[5029] chan_dahdi.c: 0x9F, 0x8B, 0x01, 0x00, 0xA1, 0x17, 0x02, 0x01, 0x01, 0x02, 0x01, 0x00, 0x80, 0x0F, 'BRATH SHANE |
07:43.44 | sbrath | Is the call getting the name info to late? |
07:44.21 | sbrath | and then: Received simple calling name 'BRATH SHANE ' |
07:44.57 | sbrath | But all the happened "After" the call was connected to my SIP phone. |
07:45.18 | sbrath | so the CDR being written down after the call is done has the "late" caller Name info. |
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07:49.15 | kaldemar | sbrath: well, you'll have a hard time getting the information to your phone. you could try to Answer() the call before dialing the SIP phone, but that's a bit ugly. |
07:50.04 | sbrath | Is this just a case of the telco takes 15 ms to lookup the Cid, and the call is routed to quickly? |
07:51.42 | sbrath | I did a wait2, answer, and now I get the name... |
07:51.45 | sbrath | Geesh. |
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07:52.22 | kaldemar | maybe, don't know. if the progress message is not tied to an answer, a mere Wait might be worth a try. |
07:53.01 | sbrath | ok, off to bed... night and thanks for the help. |
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08:15.54 | qjb | I have a problem with playing custom numbers using sayDigits. |
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08:18.24 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
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08:32.49 | qjb | I am using Asterisk 1.6 |
08:35.18 | kaldemar | qjb: what is the problem? and which 1.6? |
08:38.17 | qjb | We use 1.6.2.0-beta2 |
08:38.53 | qjb | The problem is I would like to play custom sound digits with sayDigits. However when I set the language it keeps playing the english version |
08:39.13 | *** join/#asterisk eggers (~eggers@cpe-70-124-59-214.austin.res.rr.com) |
08:39.24 | qjb | I put the sound files under /var/lib/asterisk/sounds/za/digits/[0-9].au |
08:39.47 | kaldemar | how do you set the language? and btw, 1.6.2 is on 1.6.2.2 already, that's a really old beta. |
08:39.51 | qjb | And use Set(language=za) to set the language |
08:40.18 | qjb | (Beta is old but I cannot change it at the moment, need to retest everything first) |
08:40.33 | qjb | I check the language using: NoOp(LANGUAGE ${LANGUAGE}) |
08:42.03 | qjb | That reflects the setting but then it happily using 'digits/9.gsm' (language 'en') |
08:48.33 | kaldemar | you're setting it wrong, use Set(CHANNEL(language)=en) |
08:48.56 | kaldemar | language=.. is used in the configuration files. |
08:50.02 | kaldemar | and, check it with ${CHANNEL(language)} |
08:51.10 | *** join/#asterisk c0rnoTa (~c0rnoTa@178.176.198.228) |
08:51.12 | kaldemar | of course replace "en" in the Set with what you want. |
08:52.14 | qjb | Changed it but it makes no difference |
08:53.02 | kaldemar | what does the language check say? where are your sound files? |
08:53.13 | *** join/#asterisk Da-Geek (~Da-Geek@62.189.17.99) |
08:53.47 | qjb | The check reflects the language I set. The sound files are in /var/lib/asterisk/sounds/za/digits |
08:54.08 | qjb | In /var/lib/asterisk/sounds/za I only have the 10 digits |
08:54.20 | qjb | In .au format |
08:54.31 | c0rnoTa | Hello, everybody. Can anybody tell me how i can count channels on specific extension at moment. For example, i have extension _9XXXXXXX and i want to know how much channels now are on that extension? |
08:54.35 | *** join/#asterisk mlarsen (~mlarsen@212.37.141.188) |
08:55.40 | plundra | How does penalty on a member in a queue actually work? I mean, what happens with a value of 1? what about 5? |
08:55.41 | kaldemar | c0rnoTa: in shell, do asterisk -rx 'core show channels' and then grep it as you wish. |
08:56.12 | plundra | I want to see it as kind of a tripping-point, n-tries on lower priority memebers have to be tried first. |
08:56.23 | plundra | -e |
08:56.26 | qjb | (I actually copied the en digits to za digits to be sure the format was not the problem) |
08:59.37 | *** join/#asterisk pawz (~pawz@ppp118-208-100-34.lns20.bne4.internode.on.net) |
08:59.54 | kaldemar | qjb: do you have languageprefix=yes in asterisk.conf? |
09:00.19 | c0rnoTa | kaldemar: Ok, but 'core show channels' show extens like 91234567 or 97654321 not "_9XXXXXXX". And i couldn't select which of them use "_9XXXXXXX" and which "_97.". Only using analysing process in my mind i could divide it. So, i want to divide it automaticly in AGI script. |
09:00.39 | qjb | kaldemar: that line is commented |
09:01.03 | kaldemar | qjb: uncomment, restart, try again |
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09:01.45 | *** part/#asterisk Tech_Travis (~Administr@cpe-76-168-191-127.socal.res.rr.com) |
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09:03.01 | qjb | kaldemar: I used core restart gracefully after the change but it did not fix the issue |
09:03.15 | qjb | still playing Playing 'digits/9.gsm' (language 'en') |
09:03.39 | kaldemar | feel free to show a cli output of the whole call. |
09:03.49 | markwaters | anyone got any ideas for getting asterisk to send me messages via irc for things like incoming callerid , i do it with jabber atm but would prefer irc |
09:03.51 | kaldemar | use a pastebin |
09:03.53 | c0rnoTa | kaldemar: speaking in global of my task, i want to limit abilities of outgoing calls through trunks on each exten. |
09:03.59 | mlarsen | Can anyone confirm that the following line is supposed to goto ext-queues,200,1 if the date is within january 5th to febuary 5th, and if the day of week is within monday to thursday and the time of day is with in 08:00 to 18:00, because that ain't happening? |
09:04.00 | mlarsen | exten => 82,1,GotoIfTime(08:00-18:00|mon-thu|5|jan-feb?ext-queues,200,1) |
09:04.51 | dandre | Hello, |
09:05.30 | c0rnoTa | mlarsen: it will go to ext-queues only in 05.01 or 5.02 |
09:05.37 | c0rnoTa | not 5.01-5.02 |
09:06.30 | c0rnoTa | i think :) |
09:07.51 | *** join/#asterisk icyValk77 (~icyValk77@cl-670.lon-02.gb.sixxs.net) |
09:08.09 | dandre | I am using originate command from the manager interface. Everything works fine but I need toset some variable when the first phone is dial and make this variable available when the party is dialed. I have tried to do something like set(__foo="bar") but ${foo} remains empty on the second leg of the call. |
09:08.09 | dandre | How could I do? |
09:09.05 | kaldemar | mlarsen: use two lines. one for january with days as 5-31 and one for february with 1-5. |
09:09.14 | Amorsen | Is there an easy way to generate a unique file name for ReceiveFax? |
09:09.26 | mlarsen | Thank you for your replies |
09:09.29 | Amorsen | I can't think of a way which is guaranteed to work without race conditions |
09:10.03 | c0rnoTa | Amorsen: ${UNIQUEID} ;) |
09:10.25 | Amorsen | c0rnoTa: Clever |
09:11.14 | *** join/#asterisk pentanol (~pentanol@77-35-1-202.pppoe.primorye.net.ru) |
09:11.20 | pentanol | hello |
09:11.53 | c0rnoTa | Amorsen: it's the easiest why, i think. Anyway, I'm always use this way. |
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09:13.27 | c0rnoTa | Amorsen: you can get epoch timestamp too |
09:16.05 | pentanol | if I would use asterisk +cdr, I should do trigger in the database so as to synchronize sip accounts? |
09:16.05 | Amorsen | Next question: How do you run System on untrusted data without having problems with people adding e.g. quote characters? |
09:18.00 | c0rnoTa | Amorsen: only to use AGi before System to prepare untrusted data (remove quote, or 'commenting' it like \"). |
09:18.08 | Amorsen | Ouch. |
09:18.28 | c0rnoTa | only to work with the string before it sill be passed to System |
09:19.20 | *** part/#asterisk markwaters (~markwater@weloveit.info) |
09:19.35 | c0rnoTa | or check it in the field, where it should be typed. For ex., in PHP form |
09:20.04 | Amorsen | c0rnoTa: I'm trying to pass REMOTESTATIONID from ReceiveFax |
09:20.32 | Amorsen | It would suck to be 0wned by a fax |
09:20.48 | Amorsen | If somewhat unusual |
09:22.02 | Amorsen | So the solution is to use an AGI instead of System |
09:22.36 | c0rnoTa | i think so |
09:24.49 | c0rnoTa | don't use REMOTESTATIONID as argument for AGI. get it from $AGI->get_variable |
09:24.56 | Amorsen | Right |
09:25.16 | Amorsen | Next challenge is to ensure that I actually get to the AGI; the sender might hang up on me |
09:26.57 | c0rnoTa | use AGI on 'h' extension like DeadAGI |
09:27.13 | Amorsen | Right |
09:27.23 | c0rnoTa | RceiveFax should be last exten in fax receive logic |
09:27.58 | c0rnoTa | post processing only on 'h' extension. |
09:29.48 | Amorsen | Does the h extension get called even when my side does a Hangup()? |
09:30.25 | c0rnoTa | yes |
09:30.28 | TommyBotten | How can i set the callerid for analog lines in asterisk? Callerid(num) does not seem to have any effect. |
09:31.14 | Amorsen | TommyBotten: Are you trying to set it towards your service provider or towards your phones? |
09:34.26 | c0rnoTa | Amorsen: "h" exten always be called. it does not matter who makes Hangup. Another trouble is to destroy channels after call ends, and precess 'h' logic on ZOMBIES channels. If your AGI script takes a long time to work, and bridged channel don't want hangup, you did 'exten => s,n,Hangup', and 'h' logic starts, but channels are connected, and billing seconds still rise. |
09:35.20 | c0rnoTa | that's the problem i still can not resolve. |
09:36.21 | TommyBotten | Amorsen: Towards my provider... but it seems that it's not the callerid after all. When sending a fax from an analog device via asterisk and sip to my vendor, the fax origin lists as a different number than the callerid |
09:36.30 | TommyBotten | In fact a number that is not in our series at all. |
09:37.25 | c0rnoTa | TommyBotten: i think, is telco issue, not your's |
09:39.05 | *** join/#asterisk mbrevda (~mbrevda@unaffiliated/mbrevda) |
09:39.30 | mbrevda | can you set bindaddr on a per trunk/peer basis? |
09:41.01 | kaldemar | mbrevda: no |
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09:42.07 | Amorsen | TommyBotten: You can't send callerid towards your provider with analog |
09:42.10 | c0rnoTa | another question, can i set one externip for first peer and another for second |
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09:42.17 | c0rnoTa | & |
09:42.18 | c0rnoTa | ? |
09:42.19 | Amorsen | That's hard coded per analog line |
09:42.27 | mbrevda | kaldemar: thnx |
09:43.06 | kaldemar | c0rnoTa: no |
09:43.40 | c0rnoTa | ok, thx |
09:50.31 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
09:53.11 | TommyBotten | c0rnoTa: It sounds like it. |
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10:11.50 | chasing`Sol | is there a way to disable call transfer to certain extensions? |
10:14.34 | krion | hi |
10:14.53 | krion | anyone using sevana product in order to measure MOS voip score ? |
10:16.44 | *** join/#asterisk jmls (~jmls@host217-36-208-155.in-addr.btopenworld.com) |
10:16.54 | jmls | morning all |
10:16.59 | jmls | a little question: |
10:17.56 | jmls | the manager originate command can take any number of variable:foo=baa constructs, and these are available to the dialplan (${foo}) |
10:18.27 | jmls | can you add variable:foo=baa to other ami commands, such as Redirect ? |
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10:25.00 | *** join/#asterisk Tim_Toady (~moi@194.219.240.156) |
10:25.13 | krion | jmls: you could try and see :) |
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10:27.10 | jmls | krion: yup. I could. |
10:27.30 | jmls | I could also write my own pbx from scratch ;) |
10:28.00 | krion | :) |
10:28.08 | jmls | just hoping someone else had already thought about this and tried. |
10:28.17 | jmls | nm. I'll check it out |
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10:31.10 | fiddur | jmls: Only originate, afaik... the documentation from "manager show command ..." should be accurate |
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10:50.44 | cavalier_work | Hi, I've setup an asterisk server (opensuse 11.2) and configured it with the web gui (diguim asterisk) i've added a few users for sip phones. I can call the demo extension and these work. If I call another phone it rings and i can pickup the phone, but I get no audio. |
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10:51.48 | cavalier_work | I enabled rtp set debug on and the last thing it says for rtp is Packet2Packet bridging SIP/6003-0000002b and SIP/6000-0000002c |
10:52.37 | cavalier_work | I searched for this but I couldn't find a solution |
10:55.42 | *** join/#asterisk ccesario (~ccesario@189-19-6-236.dsl.telesp.net.br) |
10:57.36 | cavalier_work | My bad, it seems to be a problem with the firewall |
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11:15.19 | TommyBotten | Is it possible to have a .call file do multiple comands - without it having to enter the dialplan |
11:17.45 | Akiraa | Does anyone have some experience with the Cisco SPA9000 IP PBX? |
11:19.18 | *** join/#asterisk hluesea (~hulusikah@88.247.127.66) |
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11:21.00 | hluesea | Hello channel, I try to configure a voicemail for the numbered extensions but i failt. My conf files are here http://pastebin.com/d4f242466. What should i do ? |
11:26.32 | garymc | anyone know what I need to set in Asterisk CLI to monitor a Analogue call? As im on PRI do i still use Pri debug span 1 ? |
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12:10.19 | kaldemar | garymc: just add verbosity and you'll see the call. if you want more accurate debug, use core set debug. |
12:10.52 | kaldemar | garymc: the pri commands only apply for PRI channels, not for analog channels. |
12:11.46 | garymc | right |
12:12.01 | garymc | so why is my analogue not displaying the DID i want it too? |
12:13.06 | kaldemar | be more specific. |
12:14.00 | kaldemar | start with what kind of channel we're talking about, what you have connected to the channel, what you expect it to do and show a cli output of a call. |
12:14.40 | kaldemar | hluesea: how did you fail? |
12:15.46 | *** join/#asterisk cuco (~Diego@local.xorcom.com) |
12:16.30 | garymc | ok. I have an A200 synced with A101D . The A200 is for FAX through my PRI. E1. I set the extension on my Analogue A200 to display DID 881049 like I do on my sip extensions. But it displays the default number 01514876699 . DEBUG here http://pastebin.mozilla.org/700984 |
12:18.21 | kaldemar | garymc: i suppose you have already asked folks at #freepbx? |
12:18.36 | garymc | i have |
12:18.52 | kaldemar | in asterisk terms, what do you mean by "set the extension on my Analogue A200 to display DID 881049"? |
12:19.29 | garymc | well thats a freepbx thing i think. |
12:19.31 | kaldemar | you don't have anything like that in the extension you pasted. |
12:19.40 | garymc | I know. its not showing it |
12:19.52 | garymc | its showing 01514876699 as the outgoing |
12:20.32 | kaldemar | "-- Executing [s@macro-outbound-callerid:12] ExecIf("DAHDI/32-1", "1|Set|CALLERID(all)=01514876699") in new stack" |
12:20.38 | garymc | yes |
12:20.49 | kaldemar | that's a freepbx issue if it's supposed to do something else. |
12:21.07 | kaldemar | go bug them some more. |
12:21.09 | hluesea | kaldemar : actually i hear the beep tone and i try to record somethings, but i can't found any records like prompt neither /var/spool/asterisk/voicemail/ nor /var/lib/asterisk/sounds/en |
12:21.29 | *** join/#asterisk grayhame (~miller@209.12.249.242) |
12:21.40 | garymc | ok |
12:23.30 | kaldemar | hluesea: this looks really strange: exten => 500,2,Record(prompt:gsm) |
12:23.33 | *** join/#asterisk Victor_Yure (~victor@unaffiliated/victoryure/x-837844) |
12:23.35 | kaldemar | hluesea: show a cli output of the call |
12:30.43 | TommyBotten | Is there a way to manually set hints? |
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13:07.39 | krion | i want to send a wav directly to a sip account, can you spare me time and tell me an FOSS who does it ? |
13:08.13 | Gido-E | krion ? |
13:08.29 | Gido-E | you can spare me time and fund my bankaccount |
13:08.40 | krion | :-) |
13:08.49 | krion | sorry if my english is not understandable |
13:09.16 | krion | i meaned to ask for help nicely, not rudely |
13:09.18 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
13:13.23 | kaldemar | krion: if you mean calling someone and playing a wav file, asterisk can do that for you. you can judge FOSS part yourself. :) |
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13:30.12 | ManxPower-work | TommyBotten: As I understand it, in 1.6 you can manually set the device state in the dialplan. |
13:31.43 | leifmadsen | DEVICE_STATE() function I think |
13:32.42 | TommyBotten | Thanks... I'll check it out |
13:34.57 | Kobaz | how would i send a hangup to a polycom phone to get rid of a call.... ie: a call comes in, asterisk crashes, and now there's a phone call ringing on the polycom that cannot be picked up |
13:36.01 | Gido-E | asterisk crashes? |
13:36.04 | Kobaz | and this is part of a ringall queue... so now you have to go around and hit the reject button on 25 phones |
13:36.07 | Gido-E | that souldn't happen |
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13:36.15 | Kobaz | Gido-E: it does... get used to it :P |
13:36.25 | Gido-E | it NEVER! crashes! |
13:36.28 | Kobaz | heh |
13:36.55 | Gido-E | Kobaz whe have multi tenant servers, never ever does a production environment crash |
13:37.01 | Kobaz | so the 90830983405834 crash reports on issues.asterisk.org are figments of my imagination... |
13:37.11 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
13:37.23 | Gido-E | Kobaz go regression test. use sipp |
13:37.30 | Kobaz | Gido-E: i have a 1.4 server that's been running for 1 year plus straght |
13:37.39 | Kobaz | Gido-E: not good enough... i have my own testing suite |
13:37.54 | Gido-E | Kobaz wat is nog good enough? SIPP? |
13:38.03 | Gido-E | What is nog good enough on it? |
13:38.06 | Kobaz | and i have other servers, running for weeks,months without crashes... and then I have this one system that crashes several times a day |
13:38.24 | Kobaz | Gido-E: i need lots of custom code to test my crap |
13:38.33 | Kobaz | but anyways |
13:38.44 | Gido-E | Kobaz on centos? |
13:38.51 | Kobaz | the issue here, is that, i need to be able to drop a call on a polycom, without asterisk actually knowing about the call |
13:38.54 | Kobaz | debian |
13:39.06 | Kobaz | 2.6.27.38 |
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13:39.54 | Kobaz | i wonder if i just store the sip call id, and then just fire off a spoofed cancel packet |
13:40.58 | Kobaz | Gido-E: i wish i could always reproduce the problem on a consistant basis.. then i can find and fix the bug, or if i can't,.. post the bug and the asterisk guys can fix it |
13:41.22 | Kobaz | but asterisk tends to crash when you're using lots of local channels... it'll crash on channel hangup |
13:42.20 | Gido-E | Kobaz ok. Try to have the same basic LIBC and GCC versions as centos. |
13:42.33 | Kobaz | i don't think it has anything to do with libc |
13:43.29 | Kobaz | there's either memory corruptio or buffer overflow in asterisk somewhere, or a referencing counting problem, or something like that |
13:43.45 | Kobaz | i need to write more tests |
13:44.14 | Gido-E | Kobaz ok, you did write your own RTP implementation for testing? |
13:45.31 | Kobaz | no, i have a whole slew of asterisk call drivers that can simulate just about anything you can do in asterisk |
13:46.41 | Gido-E | Kobaz? I have a dialplan for it. |
13:47.01 | Gido-E | I just use SIPP, to make and recieve the calls SIPP - Asterisk - SIPP |
13:47.27 | Kobaz | yeah |
13:47.31 | Kobaz | not good enough for my purpose |
13:47.46 | Gido-E | what can you test, i can't? |
13:47.48 | Kobaz | sipp just makes sip calls, and has limited call control |
13:48.33 | Gido-E | Kobaz ok. |
13:48.42 | Kobaz | i can wait for events, wait for speech/silence, send dtmf, run any dialplan |
13:49.26 | Gido-E | Kobaz why sould that not be able with my setup? The magic is done by asterisk. |
13:49.40 | Gido-E | SIPP is just a stupid caller or callee |
13:49.44 | Kobaz | i don't need a call blaster, i need scenerio testing |
13:49.54 | Kobaz | which i've already written |
13:50.24 | Kobaz | Gido-E: well the point is, you still need asterisk |
13:50.36 | Kobaz | Gido-E: sipp alone isn't a solution |
13:50.43 | dlynes_laptop | Kobaz, I surmise the polycom phones cannot be logged into with a text browser such as lynx or links, or using wget? |
13:50.53 | *** join/#asterisk Warp4 (~Robert@firewall-a.buf.ny.i-evolve.net) |
13:51.10 | Kobaz | dlynes_laptop: you can log into a polycom with a web browser, but you cant send it commands like 'hey hang up this call' |
13:51.12 | [TK]D-Fender | dlynes_laptop: You can... but it won't be pretty |
13:51.28 | dlynes_laptop | Kobaz, but if you reboot it, it'll hang up the call |
13:51.37 | [TK]D-Fender | dlynes_laptop: (links). WGET has no point. |
13:51.40 | dlynes_laptop | Kobaz, or you can't reboot a polycom from the web interface? |
13:51.42 | *** part/#asterisk cavalier_work (~cavalier_@ip4daa055d.direct-adsl.nl) |
13:51.46 | Kobaz | yeah but, hitting the reject button is 100 times faster than rebooting |
13:51.56 | Kobaz | you can reboot a polycom with a sip notify |
13:52.01 | Kobaz | you dont even have to hit the web interface |
13:52.06 | dlynes_laptop | ok |
13:52.27 | Kobaz | i think i'll experiment with spoofing a sip CANCEL |
13:53.08 | dlynes_laptop | Kobaz, incidentally, which version of asterisk are you running? |
13:53.19 | dlynes_laptop | I'm just curious |
13:53.25 | *** join/#asterisk MedicineMan (~jrodgers@75.87.82.200) |
13:53.36 | Kobaz | 1.6.0.19 |
13:53.40 | Kobaz | 1.6.0.20 crashes even more |
13:53.53 | dlynes_laptop | Yeah...I've heard the whole 1.6.0 series is quite unstable |
13:54.04 | *** part/#asterisk MedicineMan (~jrodgers@75.87.82.200) |
13:54.11 | Kobaz | it's great if all your doing is passing in and out sip calls |
13:54.14 | Kobaz | or t1, or whatever |
13:54.23 | Kobaz | if you're running lots of dialplan with local channels, not so good |
13:54.33 | dlynes_laptop | I've got my own issues with 1.6.1.8....just going to upgrade to 1.6.1.14 right now |
13:54.39 | dlynes_laptop | Had a whole machine go down last night |
13:54.47 | Kobaz | yeah i've had asterisk kill a box |
13:54.54 | Kobaz | sucks up 100% cpu, and all available io |
13:54.54 | dlynes_laptop | well |
13:54.58 | dlynes_laptop | lemme restate that |
13:55.01 | Kobaz | the only thing you can do is hit the reset button |
13:55.11 | dlynes_laptop | Asterisk locked up...not the whole box |
13:55.14 | Kobaz | yeah |
13:55.16 | Gido-E | dlynes_laptop which distro? |
13:55.25 | [TK]D-Fender | 1.6.0.22 is out.... |
13:55.26 | Kobaz | i've found that asterisk -r is bad |
13:55.30 | dlynes_laptop | Gido-E, it doesn't matter...the distro has nothing to do with it |
13:55.36 | Gido-E | dlynes_laptop it has |
13:55.40 | Kobaz | not using realtime solved a lot of my lockup issues |
13:55.41 | dlynes_laptop | Gido-E, how so? |
13:55.47 | Gido-E | But i am not going to argue about that. |
13:55.49 | dlynes_laptop | Kobaz, i'm not using real time |
13:55.53 | Gido-E | I am experience on that point. |
13:56.07 | dlynes_laptop | Gido-E, it's Debian Lenny |
13:56.13 | Gido-E | dlynes_laptop ok, try centos |
13:56.15 | dlynes_laptop | no |
13:56.18 | Kobaz | haha |
13:56.19 | Gido-E | i think it will be more stable. |
13:56.36 | TommyBotten | leifmadsen / ManxPower-work: function_devstate does mostly that. Is there a way to manipulate the actual device state, and not a custom one? |
13:56.36 | Kobaz | Gido-E: distro has very little bearing |
13:56.51 | Kobaz | Gido-E: bugs in asterisk are bugs in asterisk |
13:57.06 | dlynes_laptop | Gido-E, wtf's your problem? Centos and Debian are both running the exact same operating system |
13:57.08 | Gido-E | ok, and i am on 1.4 |
13:57.10 | Kobaz | yes, if you had a buggy kernel or libc, you'll be having problems, but that's more rare |
13:57.27 | dlynes_laptop | Gido-E, I trust Debian a lot more than Centos |
13:57.33 | Kobaz | hah, yeah |
13:57.40 | ManxPower-work | dlynes_laptop: Chances they are running different kernel, gcc, etc versions. You're right, the distro has little bearing on this. |
13:57.40 | Kobaz | i <3 my debian |
13:58.02 | Kobaz | ManxPower-work: i can run centos... and build my own kernel, upgrade gcc, put on a broken libc |
13:58.21 | dlynes_laptop | ManxPower-work, yeah, but Debian tends to test their libc/gcc into the ground before they release, so they're usually several versions behind Centos |
13:58.32 | Kobaz | ManxPower-work: the distro just by itself is almost meaningless.. what matters are the versions of these particular packages that are used as part of the distro |
13:59.12 | Kobaz | using "centos" rather than "debian" or "fedora" is just religion |
13:59.22 | Gido-E | Kobaz you have to learn a lot. |
13:59.27 | Kobaz | Gido-E: ? |
13:59.28 | dlynes_laptop | Gido-E, btw...fwiw, my Debian boxes have less problems than my Centos boxes, but stupid software like FreePBX is very Centos-centric |
13:59.37 | Kobaz | Gido-E: i've been in this industry for almost 20 years |
13:59.49 | Gido-E | Kobaz so? |
13:59.58 | Gido-E | My bank account is bigger! |
14:00.13 | Kobaz | Gido-E: from my experience, i have a good idea of what actually matters |
14:00.38 | [TK]D-Fender | dlynes_laptop: How is FreePBX distro-centric? |
14:00.55 | dlynes_laptop | [TK]D-Fender, it's completely broken for an install on anything other than Centos |
14:01.08 | dlynes_laptop | [TK]D-Fender, You have to run a special script to get it to work properly on Debian |
14:01.14 | [TK]D-Fender | dlynes_laptop: BS... tons of people running it on Ubuntu, Debian, Slackware, etc |
14:01.21 | dlynes_laptop | [TK]D-Fender, and even then, the script doesn't work 100% |
14:01.31 | dlynes_laptop | [TK]D-Fender, i didn't say it couldn't run on other distros |
14:01.37 | Kobaz | anyways, off to breakfast |
14:02.14 | dlynes_laptop | [TK]D-Fender, Have you tried installing it on Debian? |
14:02.18 | krion | kaldemar: yes, i want to do it from a client side, like ekiga or something |
14:02.19 | *** join/#asterisk _omer (~omer@119.152.140.100) |
14:02.24 | ManxPower-work | Um. FreePBX doesn't work 100% |
14:02.47 | krion | kaldemar: it's in the purpose of measuring MOS score |
14:02.49 | Kobaz | Gido-E: i agree there's always more to learn, i do need to learn tons of stuff that I know I don't know... but most likly it's not the things you think |
14:02.57 | _omer | compiling asterisk-1.4.29 in CENTOS 5.4 .... error message... http://www.pastebin.org/86220 any help ? |
14:03.56 | [TK]D-Fender | _omer: ther is no error message in there. |
14:04.09 | dlynes_laptop | _omer, where's the rest? |
14:04.43 | ManxPower-work | I thought I was the only one that didn't see an error. |
14:04.44 | Amorsen | Hmm, ReceiveFax is fine and dandy but unfortunately I can kill Asterisk 1.6.0.21 with it |
14:04.49 | _omer | let me paste once again please |
14:05.08 | krion | maybe if i call someone with ekiga then i run my wav from my computer the called would hear it |
14:05.14 | dlynes_laptop | Amorsen, might help to let peeps know if you're using ffa, or the built in fax module |
14:05.20 | _omer | there is not any error message..but compilation is stopped .. |
14:05.31 | *** join/#asterisk benngard (~benngard@213.88.138.230) |
14:05.32 | Amorsen | dlynes_laptop: Built-in |
14:05.50 | dlynes_laptop | _omer, there must be something, because in the pastebin you pasted, it mentions a problem from previous |
14:05.54 | _omer | http://www.pastebin.org/86229 |
14:05.59 | Amorsen | I guess I have won another rebuild, this time with lock debugging enabled |
14:05.59 | [TK]D-Fender | _omer: maybe its just FINISHED. |
14:06.23 | ManxPower-work | _omer: What is the command you are running? |
14:06.24 | [TK]D-Fender | _omer: "make clean" != compile. MAKE = COMPILE |
14:06.29 | *** join/#asterisk muiro (~muiro@unaffiliated/muiro) |
14:06.44 | Gido-E | _omer it tels you it went ok |
14:06.45 | dlynes_laptop | _omer, what i see in your second pastebin is that it worked, with no errors |
14:06.49 | _omer | oh.. |
14:06.55 | Gido-E | why do yuo think it went wrong? |
14:07.06 | [TK]D-Fender | Gido-E: clue[-1] |
14:07.09 | Gido-E | You issue, a make clean |
14:07.10 | dlynes_laptop | _omer, i thought those other two lines were output by the makefile, but i guess they were just your comments |
14:07.22 | _omer | I think I need to do make install too ... oops first time in linux and asterisk |
14:07.39 | Gido-E | _omer dutch? |
14:07.57 | [TK]D-Fender | _omer: "make clean", "make", "make install" |
14:08.04 | dlynes_laptop | _omer, make clean ; ./configure && make menuconfig && make && make install |
14:08.08 | Gido-E | dont forget: make menuselect |
14:08.24 | _omer | ok |
14:08.25 | dlynes_laptop | erm menuselect...getting it mixed up with the kernel |
14:08.29 | [TK]D-Fender | yeah... the ./configure would help... |
14:08.31 | *** join/#asterisk nicknick (~administr@host213-123-201-13.in-addr.btopenworld.com) |
14:08.37 | Amorsen | My my, it appears I'm hitting AST-2010-001 |
14:09.18 | _omer | **** The configure script must be executed before running 'make'. |
14:09.18 | _omer | **** Please run "./configure". |
14:09.23 | dlynes_laptop | Amorsen, 1.6.0.22 is out...don't know if it solves that issue |
14:09.25 | _omer | i did ./configure |
14:09.30 | dlynes_laptop | _omer, what did I tell you? |
14:09.31 | _omer | and then make and make install... |
14:09.34 | _omer | return remains the same |
14:09.46 | dlynes_laptop | _omer, you didn't do './configure', or it wouldn't be telling you there |
14:09.48 | dlynes_laptop | _omer, you didn't do './configure', or it wouldn't be telling you that |
14:09.51 | kaldemar | krion: or just make an extension that originates a new call |
14:10.34 | dlynes_laptop | _omer, perhaps you're trying to do make before configure? |
14:10.39 | dlynes_laptop | _omer, not the other way around? |
14:11.10 | Gido-E | _omer, after running "make clean" you can start over again. |
14:11.27 | Amorsen | dlynes_laptop: It says it does, I'm compiling now |
14:11.31 | _omer | here is the result of ./configure ... http://www.pastebin.org/86232 |
14:12.15 | Gido-E | _omer on centos? |
14:12.23 | _omer | yes Centos 5.4 |
14:12.26 | Gido-E | yum install gcc |
14:12.36 | Gido-E | yum install ncurses-devel |
14:12.55 | _omer | Package gcc-4.1.2-46.el5_4.2.i386 already installed and latest version |
14:12.55 | _omer | Nothing to do |
14:13.16 | _omer | Package ncurses-devel-5.5-24.20060715.i386 already installed and latest version |
14:13.16 | _omer | Nothing to do |
14:13.19 | [TK]D-Fender | _omer: G++ <------------- |
14:13.38 | [TK]D-Fender | _omer: Go read the INSTructiONS which tell you the pile of packages you'll need... |
14:13.42 | Katty | hi. |
14:13.44 | [TK]D-Fender | _omer: You don't seem to have done this |
14:13.44 | dlynes_laptop | [TK]D-Fender, cpp is the c preprocessor, not the c++ compiler |
14:13.54 | _omer | G++ is not a package.. |
14:14.42 | Katty | your mom's a package |
14:14.42 | Gido-E | yum install gcc-c++ |
14:14.42 | Amorsen | gcc-c++... |
14:14.42 | dlynes_laptop | Katty, aren't you chipper this morning? :) |
14:14.42 | Katty | mostly asleep |
14:14.42 | Gido-E | maybe, i am not into centos that much |
14:14.43 | Katty | no caffeine today |
14:15.10 | _omer | gcc-c++ .. updating something |
14:15.20 | Gido-E | try again... |
14:15.30 | dlynes_laptop | Why is it calling '/lib/cpp' a c++ preprocessor? very weird.... |
14:15.52 | Warp4 | dlynes_laptop, because it is |
14:16.03 | Gido-E | it needs to :-) |
14:16.13 | Warp4 | IIRC , g++ handles the actual C++ compiling |
14:16.13 | dlynes_laptop | Warp4, hrm...could've sworn it was a C preprocessor in gcc 2.95 |
14:16.39 | Warp4 | dlynes_laptop, things change with newer versions, i suppose :) |
14:16.59 | dlynes_laptop | Warp4, so asterisk actually has c++ code now? |
14:17.09 | Gido-E | Yep, seems to... |
14:17.18 | Gido-E | or just one of the wrappers for easy install... |
14:17.48 | Katty | zonks out |
14:18.04 | Gido-E | _omer there is repo for centos asterisk, check out asterisk.org |
14:18.19 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
14:18.25 | _omer | repo? |
14:18.34 | _omer | ok let me check |
14:19.28 | dlynes_laptop | Katty, you've been up all night? |
14:19.40 | *** join/#asterisk knctrnl (~aembrey@76.164.169.130) |
14:20.12 | Gido-E | _omer http://www.asterisk.org/downloads/yum |
14:21.06 | Katty | no )= |
14:21.15 | Katty | i slept rather well. |
14:21.27 | _omer | ok |
14:21.56 | *** join/#asterisk minotaur01 (~minotaur0@24.215.3.50) |
14:22.09 | dlynes_laptop | Katty, ah...I guess you just need your timmy's, dunkin' donuts, starbucks, or whatever it is you drink in missouri :) |
14:22.39 | *** join/#asterisk andres833 (~andres833@190.144.75.22) |
14:22.40 | Katty | ehhh well |
14:22.43 | Katty | i'm giving up soda, again |
14:22.44 | _omer | Gido-E .. thanks .. |
14:22.50 | Katty | which is what i usually have in the AM |
14:23.00 | dlynes_laptop | Katty, ah...soda's not good....way too much sugar |
14:23.07 | Katty | nods |
14:23.09 | _omer | but I think yum install gcc-c++ solved the problem |
14:23.15 | dlynes_laptop | Katty, or that nasty aspartame crap |
14:23.23 | Katty | ugah, i can't drink that stuff |
14:23.28 | Katty | it makes me hungryhungryhungry |
14:23.29 | _omer | so what is the sequence? ./configure make make clean ? |
14:23.50 | *** join/#asterisk xpot-mobile (~xpot@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
14:23.52 | _omer | ./configure finished WELL .... gave me ASTERISK BIG LOGO |
14:24.06 | [TK]D-Fender | _omer: read the INSTRUCTIONS int he tarball ---------- |
14:24.07 | dlynes_laptop | Katty, i won't even touch it...it fools your body into thinking it's sugar, your body processes it like sugar (or at least attempts to do so), and you get mutated cells as a result |
14:24.10 | Gido-E | ./configure && make menuselect && make && make install |
14:24.19 | dlynes_laptop | Katty, it shouldn't even be legal to sell it to anyone but diabetics |
14:24.19 | _omer | ok....thanks |
14:24.28 | _omer | I will read the instructions now... |
14:24.29 | Katty | dlynes_laptop: ehh i used to drink it back in the day |
14:24.31 | _omer | thanks all |
14:24.33 | Gido-E | make clean, whill just make a clean environment, as if you just downloaded it. |
14:24.35 | Katty | dlynes_laptop: and i drank it for years and years and years |
14:24.45 | Katty | dlynes_laptop: and wondered why i was starving all the time |
14:24.52 | Katty | dlynes_laptop: course i cooked with splenda. |
14:25.02 | Katty | dlynes_laptop: and bought all sugar-free items when i grocery shopped |
14:25.05 | dlynes_laptop | Katty, splenda's way better for you than splenda |
14:25.10 | dlynes_laptop | erm nutrasweet i mean |
14:25.11 | Katty | dlynes_laptop: orly |
14:25.15 | Katty | dlynes_laptop: is that so ;> |
14:25.21 | Katty | idk what nutrasweet is |
14:25.29 | Katty | aspartame? saccarin? |
14:25.31 | dlynes_laptop | Katty, splenda is sucralose, which is just sugar with a molecule removed |
14:25.45 | Katty | still makes me hungry |
14:25.58 | dlynes_laptop | Katty, aspartame and saccharine are both chemicals, and both quite nasty at that |
14:25.58 | Katty | how would you like to feel hungry all day long |
14:26.11 | *** join/#asterisk basty (~basty@212.218.65.131) |
14:26.13 | basty | Hi |
14:26.14 | Katty | you could eat a bigmac combo meal |
14:26.19 | Katty | and then an hour later, be hungry |
14:26.24 | dlynes_laptop | Katty, there's also stevia (a natural sweetener), but apparently it's not very healthy either |
14:26.25 | *** join/#asterisk e4 (~e4@rrcs-76-79-59-194.west.biz.rr.com) |
14:26.35 | Katty | i'll stick with sugar cane |
14:26.45 | *** join/#asterisk voipmonk (~shido6@dsl-67-204-40-42.acanac.net) |
14:26.55 | Katty | i think i'd rather have that then shovel creepy processed food items in me |
14:27.02 | dlynes_laptop | Katty, we try to buy everything with cane sugar, or honey, or chinese herbal sweeteners |
14:27.05 | Katty | hello daddy voipmonk |
14:27.14 | Katty | mmmm, honey |
14:27.18 | Katty | good with pb :> |
14:27.24 | dlynes_laptop | heh |
14:27.24 | dlynes_laptop | yeah |
14:27.30 | *** part/#asterisk Zeeek (~Zeeek@pdpc/supporter/active/zeeek) |
14:27.50 | dlynes_laptop | Katty, if you go into a chinese supply store, pretty much all the sugar in there is cane sugar |
14:27.58 | dlynes_laptop | Katty, not as sweet as white sugar, and less processed |
14:28.01 | basty | I have a strange problem. I just installed asterisk 1.4.29 on an ubuntu 64bit system. Everything works well. After I restarted the machine, I noticed that the CPU Load for the Asterisk is 100%. So I stopped the daemon and started the asterisk manual. The cpu load was gone. So I stopped it again..and ran the init script again...and boom - asterisk uses 100% again. So there must be something weirdo with the init script I guess....anyone knows of that "problem" |
14:28.13 | Katty | dlynes_laptop: good stuff. |
14:28.29 | dlynes_laptop | Katty, but they also use chinese dates to sweeten the soup |
14:28.45 | coppice | I dated a few Chinese |
14:28.56 | Amorsen | Nope 1.6.0.22 didn't fix it. Damn, I'm back to Hylafax for now. |
14:29.09 | dlynes_laptop | Katty, and some other thing (it looks like a brown ball, very light, feels like it's hollow, and sold in a plastic globe), that's a natural sweetener |
14:29.12 | Katty | don't think i've ever had a chinese date |
14:29.24 | dlynes_laptop | Katty, it's quite similar to a honey date |
14:29.27 | Katty | but i will put it on my grocery list |
14:29.40 | dlynes_laptop | Katty, they're always sold dried, not fresh |
14:29.40 | coppice | Katty: I think he means red dates |
14:29.42 | Katty | i found a recipe for this amazing looking SOUP |
14:30.08 | dlynes_laptop | coppice, they're red in color, yes...but afaik, they're only called chinese dates |
14:30.09 | Katty | must aquire groceriees to make it :> |
14:30.29 | Katty | digs through TOH website |
14:30.32 | dlynes_laptop | coppice, do you know what those brown balls are called, offhand? |
14:30.33 | coppice | dlynes_laptop: here in china we call them red dates |
14:30.38 | dlynes_laptop | ah |
14:30.50 | coppice | they make nice tea |
14:30.58 | dlynes_laptop | coppice, they're some kind of a fruit that's grown on trees, and they're used for sweetening soup |
14:31.12 | coppice | and tea |
14:31.13 | dlynes_laptop | coppice, they can also be processed to produce sugar |
14:31.39 | dlynes_laptop | but i guess you don't know the name |
14:31.48 | dlynes_laptop | never heard of anyone using them in tea, though |
14:32.51 | coppice | In Shenzhen Science Park there is an excellent SiChuan restaurant that serves red date tea to everyone |
14:33.36 | coppice | and they fill your cup in the really traditional way, with a kettle that has a super long spout |
14:34.32 | dlynes_laptop | coppice, oh...thought you were talking about the chinese honey dates |
14:34.40 | dlynes_laptop | coppice, not the brown balls |
14:36.15 | dlynes_laptop | coppice, sounds like those tea pots that you see in the old chinese dynasty films |
14:37.18 | Katty | http://www.recipezaar.com/Crock-Pot-Taco-Soup-40022 <- tell me that don't sound good |
14:38.20 | coppice | dlynes_laptop: that's the kind of thing. a brass kettle with a metre long spout |
14:38.22 | creativx | crack pot? |
14:38.44 | coppice | you have to store your crack somewhere |
14:39.33 | dlynes_laptop | Katty, now why would anyone drain the juices? that sounds just plain weird |
14:39.35 | [TK]D-Fender | would never do a drug named after a part of his ass |
14:39.43 | Katty | dlynes_laptop: some of us don't like grease |
14:39.46 | *** join/#asterisk brezular (~brezular@adsl-dyn-104.95-102-247.t-com.sk) |
14:40.09 | coppice | Katty: you must hate Shanghai food :-) |
14:40.15 | Katty | never had it. |
14:40.23 | Katty | but if it's overly greasy, probably make me sick |
14:40.24 | [TK]D-Fender | ^ Tragically white |
14:40.52 | Katty | some dishes aren't liquidy |
14:40.54 | [TK]D-Fender | Katty: Hot damn girl..... get some colour in ya! |
14:41.05 | eppigy | MONRING |
14:41.13 | Katty | hugs eppigy |
14:41.53 | eppigy | huggles Katty |
14:42.19 | Katty | eppigy: so did you have steak n potato for dinner last night |
14:42.33 | eppigy | yesh |
14:42.39 | Katty | did you cook it yourself? |
14:42.42 | eppigy | yesh |
14:42.45 | eppigy | rare |
14:42.45 | Katty | :>>> |
14:42.48 | eppigy | RARE |
14:42.49 | Katty | applauds |
14:42.52 | Katty | mooing? |
14:42.56 | eppigy | yes |
14:42.59 | Katty | k |
14:42.59 | eppigy | mooing for mercy |
14:43.08 | Katty | i like mine medium |
14:43.13 | Katty | and not oozing red stuff |
14:43.23 | Amorsen | Ah, I'm hitting issue 16374 |
14:43.26 | ManxPower-work | <-- not a fan of eating dead animals |
14:43.37 | Katty | oh nice, some person just went in my yard |
14:43.43 | Katty | he looks...suspiciously religious |
14:43.43 | dlynes_laptop | Katty, what does draining the juices have to do with grease? |
14:44.00 | Katty | dlynes_laptop: err |
14:44.04 | Katty | dlynes_laptop: when you cook ground beef |
14:44.09 | Katty | dlynes_laptop: all of the fat 'melts' into the pan |
14:44.16 | Katty | dlynes_laptop: and that's called grease |
14:44.20 | dlynes_laptop | Katty, they're talking about draining the juices out of canned tomatoes and the like |
14:44.29 | Katty | what? |
14:44.31 | Katty | looks at recipe |
14:44.34 | dlynes_laptop | Katty, in your recipe |
14:44.49 | Katty | 4 - do not drain cans |
14:44.55 | dlynes_laptop | Katty, exactly |
14:45.09 | Katty | can you repeat the question |
14:45.11 | Katty | rubs eyes |
14:45.21 | dlynes_laptop | Katty, now why would anyone drain the juices? that sounds just plain weird |
14:45.32 | Katty | it's a recipe |
14:45.34 | [TK]D-Fender | Katty: ... What fat is there to drain exactly? |
14:45.38 | Katty | some people don't know how to cook |
14:45.42 | *** join/#asterisk jaytee (~jforde@unaffiliated/jaytee) |
14:45.48 | Katty | hi jaytee |
14:45.52 | jaytee | hi Katty |
14:45.53 | Katty | [TK]D-Fender: your mom. |
14:45.56 | Katty | hugs jaytee |
14:46.06 | coppice | dlynes_laptop: surely you don't want any liquid in your soup |
14:46.09 | [TK]D-Fender | Katty: Well? |
14:46.13 | Katty | [TK]D-Fender: Deep Subject |
14:46.16 | dlynes_laptop | coppice, no...that would make no sense at all |
14:46.19 | [TK]D-Fender | Katty: Deep dish |
14:46.24 | Katty | [TK]D-Fender: pizza? |
14:46.28 | Katty | perks up |
14:46.28 | [TK]D-Fender | Katty: So what fat are we talking about there? |
14:46.37 | Katty | [TK]D-Fender: we're talking about fat? |
14:46.38 | [TK]D-Fender | Katty: I fail to see how it applies in this case |
14:46.42 | *** part/#asterisk ManxPower-work (~EWieling@216.186.151.147) |
14:46.43 | Katty | [TK]D-Fender: i thought we were talking about draining vegetables |
14:46.48 | dlynes_laptop | well...besides |
14:47.07 | dlynes_laptop | if you buy extra lean ground beef, there shouldn't be much fat, anyways |
14:47.08 | Katty | dlynes_laptop: i would probably drain and rinse the kidney beans |
14:47.14 | [TK]D-Fender | Katty: You brought up drining the juices to cut grease |
14:47.18 | [TK]D-Fender | draining* |
14:47.20 | Katty | dlynes_laptop: just because the stuff they pack it in is pretty .......ick |
14:47.36 | dlynes_laptop | Katty, they pack it in water |
14:47.47 | Katty | yeah but have you seen the water they're sitting in? |
14:47.51 | Katty | it looks awful |
14:47.56 | dlynes_laptop | Katty, yeah..same water it was cooked in |
14:48.00 | Katty | eww eww eww |
14:48.15 | dlynes_laptop | Katty, well, you're eating the kidney beans |
14:48.25 | Katty | yeah but they look normal :P |
14:48.26 | dlynes_laptop | Katty, and it was cooked in that water |
14:48.32 | Katty | not greyish brown liquid |
14:48.48 | Katty | that just looks nasty |
14:48.49 | dlynes_laptop | Katty, oh...it's not greyish brown liquid here |
14:48.53 | Katty | it is here |
14:48.54 | *** join/#asterisk fofware (~chatzilla@host171.190-30-113.telecom.net.ar) |
14:48.59 | dlynes_laptop | it's a much paler color than the color of the beans |
14:49.14 | Katty | maybe i'll try another brand then |
14:50.27 | Katty | crock pot soup is the best. |
14:50.41 | Katty | i loves it. |
14:50.54 | eppigy | sleeple |
14:51.03 | Katty | me too |
14:51.04 | eppigy | i want more caffeine |
14:51.07 | eppigy | but I shouldnt |
14:51.24 | Katty | why |
14:51.42 | dlynes_laptop | Katty, stokely van camp's is usually pretty good |
14:51.44 | Katty | you have the metabolism of a miniture greyhound |
14:52.09 | Katty | dlynes_laptop: k |
14:52.29 | Katty | darn religious people in my yard |
14:52.31 | Katty | scarin my critters |
14:52.33 | dlynes_laptop | Katty, the cheap brands are usually quite nasty |
14:52.54 | Katty | dlynes_laptop: yeah and they always put these crazy added ingredients in stuff |
14:53.29 | Katty | Major General is awake! |
14:54.27 | Katty | he nests in the big tree across the street :> |
14:55.31 | *** join/#asterisk Yedidya (~chatzilla@host86-142-22-34.range86-142.btcentralplus.com) |
14:56.03 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
14:56.03 | *** mode/#asterisk [+o putnopvut] by ChanServ |
14:56.23 | Katty | hi putnopvut |
14:56.34 | putnopvut | Hi Katty. What's happening? |
14:56.49 | Katty | not sure, but when i wake up i'll let ya know |
14:57.22 | Katty | so far, i have swapped around backup harddrives |
14:57.46 | putnopvut | awesome |
14:57.51 | putnopvut | I'm headed to Brussels today! |
14:58.52 | Katty | travel safely |
14:59.01 | Katty | and don't let the crazies get you |
15:01.45 | Katty | major general is stuffin his chompers this morning |
15:01.57 | Katty | p3nguin: idk if you heard, but they're callin for ice tonight |
15:02.56 | Katty | p3nguin: and then snow this weekend |
15:06.19 | *** join/#asterisk soman (~somnath@stargate.starnet.fi) |
15:06.38 | dlynes_laptop | Is it normal to have a channel id of '00000000'? |
15:07.29 | dlynes_laptop | eg. SIP/navigata-gw-out-00000001 is making progress passing it to SIP/MorganSwitch-00000000 |
15:07.30 | *** join/#asterisk Cuz (~plastik@mail.gradeatechs.com) |
15:07.48 | dlynes_laptop | In this case, both channel ids look abnormal to me |
15:08.15 | dlynes_laptop | It always seems to be a random 8 digit hex number, but in this case, they seem to be sequential |
15:09.38 | Cuz | Could anyone help me with a custom CDR problem I'm having? As I read it, all I need to do is set an accountcode= for extensions that i'd like to do custom outbound logging for, setting an "accountcode" (in my case i use "support" to designate the support queue), it makes a support.csv file in /var/log/asterisk/cdr-csv, for whatever reason my support.csv remains at 0 bytes consistently |
15:10.08 | Cuz | i mean the file gets created when i specify accountcode, but nothing seems to be getting logged to it, is there a switch i missed turning on in some conf file somewhere? |
15:10.30 | dlynes_laptop | Cuz, it should be creating it in /var/log/asterisk/cdr-custom/Master.csv |
15:10.38 | Cuz | i'm trying to create these custom CDR files to parse out for a queue statistics generator |
15:11.02 | Cuz | dlynes_laptop: no, thats where things go if you don't specify "accountcode", Master.csv is the default CDR log file |
15:11.23 | dlynes_laptop | Hrm....did asterisk 1.6.1.14 change everything over so that channel identifiers are now sequential? |
15:11.36 | dlynes_laptop | Cuz, well, you were saying you were creating a custom cdr |
15:11.50 | Cuz | yes, which is "support" |
15:11.50 | dlynes_laptop | Cuz, adding account code does not make the cdr custom |
15:11.51 | Katty | your mom's a custom cdr. |
15:12.15 | Cuz | dlynes_laptop: oh, how i read this, it sounded like it did |
15:12.20 | [TK]D-Fender | Cuz: accoundcode is a FIELD in the amster.csv |
15:12.25 | Cuz | 2. All outgoing calls are stored in CDR files. The files are placed in /var/log/asterisk/cdr-csv/ by default. The Queue Statistics application does not parse the Master.csv file, because in some cases it might be too large and heavy. That is why you have to create your own *.csv files for the groups you need. You may add 'accountcode' value for the accounts you use in sip.conf or iax.conf. |
15:12.46 | dlynes_laptop | Cuz, accountcode and userfield are both normal fields....they just aren't normally in the csv file unless you use them |
15:12.49 | [TK]D-Fender | Cuz: what "Queue Statistics application"? |
15:12.56 | Cuz | so I've done that, how do i populate the data? |
15:13.03 | Cuz | [TK]D-Fender: http://www.asteriskguru.com/tutorials/installation_guide.html |
15:13.34 | [TK]D-Fender | Cuz: Oh.. you men 3rd party stuff we don't directly support! |
15:13.36 | [TK]D-Fender | mean* |
15:13.49 | Cuz | i'm not asking about support on 3rd party products |
15:13.56 | Cuz | i'm asking how to create a custom cdr file :) |
15:14.06 | [TK]D-Fender | Cuz: Well accountcode is jsut a field in the main |
15:14.39 | Cuz | well, why does it create a 0 byte <accountcodename>.csv when i give any extension an accountcode then? |
15:14.54 | [TK]D-Fender | 'cuzThere is a CONF FILE for you to make a custom format for different purposes... but this is completely different |
15:14.55 | Cuz | in /var/log/asterisk/cdr-csv/ |
15:15.24 | dlynes_laptop | Cuz, what it's saying is that it'll still log to Master.csv, but when you use account codes, asterisk will automatically create an 'accountcode.csv' file for every account code used |
15:15.32 | Cuz | as i read it, i just need to create custom cdr files for all extensions i have in the queue for this thing to parse, so that's all I'm trying to do. |
15:15.41 | dlynes_laptop | Cuz, they're not custom |
15:15.55 | Cuz | sorry, not custom |
15:16.24 | Cuz | just so the data for the accounts specified either dumps logs into both Master and support.csv or just support.csv, is this possible? |
15:16.42 | dlynes_laptop | so say you've got a sip peer with an account code of 'joeblow', you'll get an additional file called 'joeblow.csv' in your cdr-csv directory |
15:17.00 | Cuz | i still don't follow |
15:17.00 | dlynes_laptop | all the calls for joeblow will end up in both joeblow.csv and Master.csv |
15:17.11 | Cuz | yes, that's what i want |
15:17.25 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
15:17.27 | Cuz | that's not what's happening now |
15:17.30 | dlynes_laptop | And it does it that way so that your parsing engine only has to parse joeblow.csv |
15:17.39 | Cuz | the "joeblow.csv" is staying at 0 bytes |
15:17.52 | dlynes_laptop | Cuz, are your calls showing up in Master.csv for your accountcode? |
15:18.12 | dlynes_laptop | Cuz, if not, it's because you haven't made any calls that were assigned to that accountcode yet |
15:18.50 | dlynes_laptop | Cuz, the way you're currently doing it, it'll only log outbound calls, using the accountcode, not inbound |
15:20.04 | Cuz | oh no, i think there's a bigger problem here actually |
15:20.20 | Cuz | I just realized Master.csv hasn't grown in size since November |
15:20.23 | dlynes_laptop | Cuz, accountcode= only affects the sip user |
15:20.27 | Cuz | so that's probably the issue |
15:20.33 | dlynes_laptop | Cuz, there ya go |
15:20.55 | dlynes_laptop | Cuz, check to make sure cdr_csv.so is loaded |
15:21.48 | TommyBotten | Is it so, that there is no way to retain the original caller id on attended transfer? |
15:22.03 | dlynes_laptop | TommyBotten, correct |
15:22.08 | Cuz | but the funny part is, November was when i started using these "accountcode" lines, and was when i originally started trying to get these queue stats to work |
15:22.10 | dlynes_laptop | TommyBotten, other than in the logs |
15:22.28 | dlynes_laptop | Cuz, that's not the reason it stopped working |
15:22.37 | dlynes_laptop | Cuz, you probably made other changes that affected it |
15:22.39 | Cuz | dlynes_laptop: how do i check to see if that module is loaded? |
15:23.11 | *** join/#asterisk n3hxs (~HAMming@static-151-196-93-200.balt.east.verizon.net) |
15:23.20 | dlynes_laptop | Cuz, module show |
15:23.44 | Cuz | ew 185 modules |
15:23.46 | dlynes_laptop | Cuz, anything that's listed is loaded |
15:23.47 | Cuz | how do i grep it? :P |
15:24.02 | dlynes_laptop | /usr/sbin/asterisk -rx "module show" | grep cdr_csv |
15:24.11 | Cuz | oh yah, that would work |
15:25.38 | *** join/#asterisk zorp75ck (~zorp75ck@pool-209-158-23-39.altnpa.east.verizon.net) |
15:26.58 | *** join/#asterisk NicoleMun (~ssharma@pool-173-63-185-74.nwrknj.fios.verizon.net) |
15:27.27 | Cuz | cdr_csv.so Comma Separated Values CDR Backend 0 |
15:27.32 | Cuz | okay, so it's loaded |
15:27.41 | Cuz | what else would cause the CDR's to stop? |
15:28.43 | Katty | thinks about making a comment |
15:29.55 | Cuz | whoa |
15:29.58 | Cuz | okay |
15:30.05 | Cuz | so i'm missing the cdr.conf file in /etc/asterisk |
15:30.10 | Cuz | though cdr.conf.sample is there |
15:30.21 | Cuz | that ain't normal, is it? :P |
15:30.44 | dlynes_laptop | Cuz, that ain't normal, cuz you're not normal :) |
15:31.05 | dlynes_laptop | Cuz, rename the cdr.conf.sample to cdr.conf and use a text editor and configure it |
15:31.06 | Cuz | is it, or isn't it? |
15:31.17 | Cuz | yea, and do a reload? or an amportal restart? |
15:31.58 | Katty | you do the hokeypokey |
15:32.10 | Katty | cause that's what it's all about |
15:32.20 | Cuz | so helpful |
15:32.24 | Katty | i know, aren't i? |
15:32.32 | Cuz | rolls his eyes |
15:32.33 | Katty | have a nice day. |
15:32.52 | dlynes_laptop | Cuz, ummmm |
15:32.56 | dlynes_laptop | ~gui |
15:32.57 | infobot | gui is, like, (Graphical User Interface) Any system that uses graphics to represent the functions of a program. All Windows operating systems are GUI. or point-and-drool, double-click, drag-and-drop, such as http://toastytech.com/guis/miscwin1xp.png. Of course Real Programmers use the command line interface. See cli |
15:33.00 | dlynes_laptop | ~freepbx |
15:33.01 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
15:33.17 | dlynes_laptop | Cuz, freepbx has all kinds of issues... |
15:33.19 | Cuz | yea? |
15:33.21 | Katty | dlynes_laptop: the hokeypokey! |
15:33.24 | Katty | dlynes_laptop: come on then! |
15:33.36 | Katty | also, my workstation keeps rebooting |
15:33.37 | Katty | and irritating me |
15:33.41 | Katty | and it is not the power supply |
15:33.44 | Katty | grumps, throws things |
15:33.54 | Cuz | what about freepbx? |
15:33.58 | Cuz | i never said i was using freepbx |
15:34.03 | Cuz | why would you make that assumption? |
15:34.13 | Katty | are you denying it? |
15:34.21 | Gido-E | :-) |
15:34.26 | Cuz | sure |
15:34.45 | Katty | that is not an appropriate response. |
15:34.50 | Katty | an appopriate response is Yes or No |
15:34.58 | *** join/#asterisk basty (~basty@212.218.65.131) |
15:35.00 | basty | hi |
15:35.00 | Cuz | does it matter? |
15:35.12 | Katty | basty: hi |
15:35.13 | [TK]D-Fender | [10:31]<Cuz>yea, and do a reload? or an amportal restart? <--- AMPORTAL = FreePBX |
15:35.24 | Cuz | koay fine |
15:35.28 | [TK]D-Fender | Cuz: Don't play us for dumb |
15:35.40 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
15:35.42 | Katty | plays [TK]D-Fender for dumn |
15:35.43 | basty | is there any way to get app_rxfax.c and app_txfax.c compiled on asterisk 1.4.29 ? We moved from 1.2.X to the latest 1.4. And we used that app to recive fax with spandsp. |
15:35.44 | Katty | oh |
15:35.46 | Cuz | so do i do a /etc/rc.d/init.d/asterisk restart? |
15:35.51 | Katty | plays [TK]D-Fender for DUN DUN DUN DUN |
15:35.54 | Cuz | ... |
15:35.56 | Katty | hi tony |
15:36.05 | [TK]D-Fender | curestart * however appropriate to how you're running your install |
15:36.11 | [TK]D-Fender | Cuz: restart * however appropriate to how you're running your install |
15:36.56 | Cuz | i know how borked freepbx is, i still have to use it, because i can't afford to train 3 other guys on how to use asterisk console |
15:37.27 | [TK]D-Fender | Cuz: Less story, more action |
15:38.28 | *** join/#asterisk guax (~guax@unaffiliated/guaxinim) |
15:38.55 | Cuz | does the CDR module really require the .conf file to be sitting in /etc/asterisk though? i mean most of the lines in that file are commented out anyways |
15:39.02 | Cuz | in the sample |
15:39.14 | Cuz | and the defaults are fine for what i want to do |
15:39.24 | guax | just remove and test it |
15:39.55 | Katty | guax: he's using freepbx. there's no telling what his system is doing |
15:39.56 | Cuz | i'm trying |
15:40.00 | Cuz | i run 23 lines off this box |
15:40.04 | [TK]D-Fender | Cuz: You want custom CDR then you have to DEFINE IT |
15:40.05 | Cuz | i need to wait till everyone's done their calls |
15:40.14 | guax | you must have a backend, so prehaps the conf file just need to say that cdr is active, the rest stays on cdr_odbc and so on |
15:40.21 | guax | Katty, oh |
15:40.38 | Cuz | [TK]D-Fender: i don't think i want a "custom cdr" per-se, i just want to take the data that normally gets logged in Master.csv and put it in support.csv |
15:40.39 | Cuz | or both |
15:40.58 | [TK]D-Fender | Cuz: Symlink it then. |
15:41.17 | Cuz | no no, i mean for specific extensions with the accountcode "support" |
15:41.30 | guax | no support for that on cdr tough |
15:42.58 | Cuz | what do you mean? |
15:43.41 | guax | do you want to keep the rest of data on Master.csv and just record for suport? |
15:43.49 | guax | support* |
15:43.52 | Cuz | yes |
15:44.14 | Cuz | my problem right now is though nothing is going to either log file |
15:44.21 | guax | a pipe on cron can help you but i dont see native support in asterisk for that |
15:45.15 | [TK]D-Fender | Cuz: I see no mechanism for * to split where it saves CDR records to. Custom CDR only selects the line formatting, not the selection criteria |
15:45.45 | [TK]D-Fender | Cuz: perhaps you could make a script to filter the master into sub-files. |
15:45.55 | Cuz | why does it create a new .csv file when an accountcode is specified for an extension then? |
15:45.58 | [TK]D-Fender | Cuz: You'd have to trigger this on some regular basis |
15:46.13 | russellb | or use a database, and a simple SELECT will get you only the data you want. |
15:46.54 | russellb | i would much prefer that over text file mangling, personally. |
15:46.59 | guax | cat Master.csv | grep "support" > support.csv on crontab :) |
15:47.23 | guax | russellb, then you loose the style :D |
15:47.29 | russellb | sends a bunch of calls with callerid name of "support" |
15:47.31 | [TK]D-Fender | russellb: I'mt hinking his 3rd party reporting tool isn't flexible for that |
15:47.42 | Cuz | i understand what you guys are saying |
15:47.55 | Cuz | [TK]D-Fender: do you use queues? |
15:47.59 | [TK]D-Fender | Cuz: Yes |
15:48.04 | russellb | You know, I wrote a CDR module, cdr_python a long time ago that would have been nice for this |
15:48.08 | Cuz | do you generate statistics for said queues? |
15:48.14 | russellb | it just gave you CDRs as a python object that you could do whatever with |
15:48.16 | Katty | FOR SAID QUEUES |
15:48.18 | [TK]D-Fender | Cuz: No, ASTERISk generates them :) |
15:48.20 | Katty | said queues. |
15:48.26 | Katty | Qs. |
15:48.30 | russellb | it would be a 2 line script ... if accountcode == "support": writeitoverhere... |
15:48.32 | Katty | For, Said Qs. |
15:48.44 | russellb | should get that merged. |
15:48.46 | guax | russellb, you are cheating, the suggested kludge works here, hehe. |
15:48.50 | Cuz | yes, i understand how to manipulate text data on a cronjob |
15:48.55 | Katty | Kludge? |
15:48.55 | Cuz | i'm pretty good with awk, sed, grep |
15:49.02 | Katty | guax: define Kludge. |
15:49.06 | [TK]D-Fender | that's what she sed :p |
15:49.12 | Cuz | i'm wondering if asterisk has any build in mechanisms to accomplish my goal |
15:49.23 | [TK]D-Fender | Cuz: Clearly looking like "no". |
15:49.24 | guax | Katty, http://en.wikipedia.org/wiki/Kludge |
15:49.39 | Cuz | [TK]D-Fender: asterisk does not generate anything usable, such as pie charts/graphs/etc for the suits |
15:49.41 | [TK]D-Fender | Cuz: Shitty CDR tool to use CSV where every other tool uses SQL |
15:49.47 | n3hxs | Kludge... a software Rube Goldberg? |
15:49.48 | Cuz | i mean the raw data, yes |
15:49.58 | Cuz | [TK]D-Fender: okay |
15:50.05 | Cuz | so maybe i started out with the wrong question |
15:50.06 | [TK]D-Fender | Cuz: * isn't a GUi reporting tool, its a bunch of engine parts <- |
15:50.16 | [TK]D-Fender | Cuz: Get out of your Happy-Happy GUI FreePBX world. |
15:50.22 | Katty | guax: ohisee, i will have to add that to my fun list of words. |
15:50.24 | [TK]D-Fender | Cuz: You don't seem to realize where you are and what * is |
15:50.25 | Cuz | what would you guys use to generate queue statistics that you can give to suits so they can go oooooh ahhh |
15:50.50 | [TK]D-Fender | Cuz: Your 3rd party tool was based on a dumb idea. Complain to them. My tools work GREAT |
15:51.03 | Cuz | right, what tool are you using then? |
15:51.13 | [TK]D-Fender | Cuz: ScopServ |
15:51.15 | Cuz | this should have been my question to begin with |
15:51.20 | guax | Katty, thats all the theory behind it: klumsy, lame, ugly, dumb, but good enough |
15:51.32 | Cuz | oh if Katty's trying to talk to me, sorry... |
15:51.34 | *** join/#asterisk ManxPower-work (~EWieling@216.186.151.147) |
15:51.38 | Cuz | can't see |
15:51.45 | [TK]D-Fender | Cuz: No, asking to find a way to use what you have better is a good start. Realizing that yours is a DEAD END is a good thing too. |
15:51.50 | Katty | guax: there's a similiar word. |
15:51.55 | Cuz | yep |
15:51.56 | Katty | guax: MonkeyRigging |
15:52.00 | Katty | guax: but it has racist undertones |
15:52.02 | Katty | guax: :<<< |
15:52.05 | Cuz | i hated this piece of crap the last time i tried to set it up |
15:52.27 | Cuz | i found a good commericalware package, but the boss wants me to test out free ones first |
15:52.29 | *** join/#asterisk jpeeler (~jpeeler@asterisk/digium-software-dev/jpeeler) |
15:52.34 | [TK]D-Fender | sets the PR team for "Primate Equality" on Katty |
15:53.05 | Katty | What the razzleberries are you talking about |
15:53.16 | ManxPower-work | We're here! We're primates! Get used to it! |
15:53.26 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
15:53.26 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
15:53.38 | Katty | hji mister blitzrage |
15:53.40 | [TK]D-Fender | monkey see, monkey doo-doo... flungat your face! |
15:53.41 | Katty | -j. |
15:53.51 | leifmadsen | Katty: yo! |
15:53.57 | [TK]D-Fender | leifmadsen: I don't want to meet your mom! |
15:54.01 | leifmadsen | I just want... |
15:54.03 | [TK]D-Fender | leifmadsen: ! ! ! |
15:54.05 | [TK]D-Fender | \o/ |
15:54.08 | Katty | leifmadsen: cougar |
15:54.09 | leifmadsen | I sure do :) |
15:54.10 | Katty | oh |
15:54.12 | Katty | wrong person |
15:54.14 | Katty | [TK]D-Fender: cougar |
15:54.15 | leifmadsen | haha |
15:54.16 | Katty | [TK]D-Fender: bate |
15:54.22 | Katty | [TK]D-Fender: in soviet russia... |
15:54.30 | [TK]D-Fender | "bait" <- and yes.... this was my previous demographic |
15:54.40 | Cuz | [TK]D-Fender: i have this working right now, and it's SQL based |
15:54.41 | Cuz | http://www.asternic.biz/ |
15:54.46 | Cuz | but it costs $$$ |
15:54.51 | Katty | oh noes, it costs $$$ |
15:54.57 | Katty | END OF THE WORLD |
15:55.09 | Katty | how dare my lunch cost money |
15:55.15 | Katty | it is unacceptable |
15:55.16 | [TK]D-Fender | Cuz: Don't go expecting that you are going to find a free tool you will like for every business goal you have <- |
15:55.28 | Katty | i demand lunch for free, on a silver platter, with everything i want. |
15:55.33 | Cuz | no, i'm 100% aware of this |
15:55.39 | Katty | NO SUBSTITUTES ACCEPTABLE |
15:55.40 | [TK]D-Fender | Cuz: Life sucks, but rarely swallows |
15:55.45 | ManxPower-work | Telecom wants ot be free! |
15:55.57 | Katty | ManxPower-work: freeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeee |
15:55.59 | Katty | ManxPower-work: as a bird |
15:56.07 | *** part/#asterisk benngard (~benngard@213.88.138.230) |
15:56.14 | Cuz | so do you know the name of any 1/2 decent queue stats parser i can look into, since this asteriskguru one is complete ass |
15:56.37 | Cuz | have this one going too |
15:56.37 | Cuz | http://www.orderlyq.com/ |
15:56.44 | Cuz | but again, it's commerical |
15:56.50 | Gido-E | Cuz :-) |
15:56.51 | Katty | i know one. |
15:56.59 | Katty | it's called WriteYerOwn, Inc. |
15:57.01 | ManxPower-work | Most queue and cdr stat applications will cost money. |
15:57.24 | Katty | does asterisk-stat have queue stuffs |
15:57.33 | Katty | don't think it does |
15:57.45 | Gido-E | Almost the latest Linux traffic shaping patches ready. In a short while you can see traffic per rule in the traffic shaper. |
15:58.09 | Cuz | ManxPower-work: Yea, that's becomming more and more obvious the harder i look |
15:58.38 | Cuz | but my question is, is anyone in here generating queue stats with any free software they could recommend to me :P |
15:58.50 | ManxPower-work | Cuz: Most people seem to write their own. |
15:59.05 | Cuz | uh yea |
15:59.09 | *** join/#asterisk Deeewayne (~dwayne@75.76.254.162) |
15:59.10 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
15:59.10 | Gido-E | Cuz push it to snmp? :-) |
15:59.16 | Cuz | lol |
16:02.02 | *** join/#asterisk Geminizer (~whoami@cpe-76-180-27-4.buffalo.res.rr.com) |
16:02.16 | Katty | hugs Deeewayne |
16:02.17 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
16:02.29 | Katty | Deeewayne: youj're missing some awesome squirrely action this morning |
16:02.37 | Deeewayne | O.O |
16:02.41 | Deeewayne | Katty, good morning! |
16:02.47 | Deeewayne | checks the squirrels |
16:04.13 | Geminizer | Hello all. Is there an exclusion operator that can be used with trunk settings (e.g. basically allow any number of the form NXXNXXXXXX except 110NXXXXXX)? |
16:04.48 | [TK]D-Fender | Geminizer: No. |
16:05.11 | Gido-E | Geminizer but asterisk wil do the exact match for the other one |
16:05.20 | [TK]D-Fender | Geminizer: Check the # in that extension, or make another exten with the other pattern so that it catches it instead |
16:05.20 | Gido-E | You can add a noop or hangup |
16:05.22 | Deeewayne | Katty, awwww |
16:07.04 | Gido-E | Deeewayne it is also 5pm here end of the normal workday |
16:07.22 | Gido-E | s/ also// |
16:07.37 | Deeewayne | Gido-E, yay |
16:07.37 | Katty | Deeewayne: much face stuffing going on. |
16:10.47 | Geminizer | thanks |
16:11.05 | *** join/#asterisk Alfio (~amunoz@75.112.88.200.m.sta.codetel.net.do) |
16:13.05 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
16:13.23 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
16:14.49 | *** join/#asterisk IBC_JKENNEY (~jkenney@ip65-44-169-66.z169-44-65.customer.algx.net) |
16:15.40 | IBC_JKENNEY | Hey has anyone had any issues where you cannot dial a specific number. We are using asterisk with PRI cards on XO IP-FLEX service |
16:15.51 | IBC_JKENNEY | they say we are using non-standard signaling |
16:16.09 | IBC_JKENNEY | has anyone else had this problem |
16:16.16 | ManxPower-work | IBC_JKENNEY: That non-standard signalling might be? |
16:16.34 | ManxPower-work | IBC_JKENNEY: you understand that PRIs don't play busy messages or number disconnected messages, right? |
16:16.49 | IBC_JKENNEY | Yes |
16:17.00 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
16:17.01 | IBC_JKENNEY | And i get a message |
16:17.02 | ManxPower-work | IBC_JKENNEY: what is the HANGUPCAUSE of the failed call? |
16:17.09 | IBC_JKENNEY | the call doesn't fail |
16:17.17 | IBC_JKENNEY | the call is a ATT conference bridge |
16:17.34 | IBC_JKENNEY | when i call from traditional pots (outside XO) or my cell it works fine |
16:17.54 | ManxPower-work | So you CAN call that number. |
16:17.55 | *** join/#asterisk moy (~moy@74.12.123.169) |
16:18.00 | IBC_JKENNEY | if i call from IP-flex it rings 5 or six times and states that it could not access the bridge |
16:18.09 | IBC_JKENNEY | Yes but not from my asterisk pbx |
16:18.11 | IBC_JKENNEY | or PRI's |
16:18.18 | IBC_JKENNEY | on the IP Flex |
16:18.37 | IBC_JKENNEY | we have another PRI that is broke out traditionally that works fine |
16:18.45 | ManxPower-work | IBC_JKENNEY: IF it's a PRI we don't care what the underlying transport is (except for cases of fax and modem) |
16:18.46 | IBC_JKENNEY | for our modem pools so i stole a line from there and it works |
16:18.57 | ManxPower-work | Maybe you are sending out bad callerid? |
16:19.17 | IBC_JKENNEY | I thought of that however XO has us set if our callerid is invalid or wrong we can't call out |
16:19.32 | ManxPower-work | IBC_JKENNEY: then I have no further suggestions |
16:19.37 | [TK]D-Fender | IBC_JKENNEY: got debug to show us? |
16:19.40 | thansen | anyone have experience with nortel NTDU91 and asterisk (sip)? or nortel in general? |
16:19.48 | n3hxs[away] | is now away - Reason : Auto-Away after 30 minutes |
16:19.51 | [TK]D-Fender | IBC_JKENNEY: because so far you haven't given us any details. |
16:20.40 | IBC_JKENNEY | not for today i am running testing tonight once business closes and call traffic is slower |
16:21.08 | IBC_JKENNEY | however when XO plugged the PRI into a test set the call worked but they plugged it in before it went into the cisco router |
16:21.55 | IBC_JKENNEY | so there are 3 T1's that plug into the cisco and 2 of them plug into the asterisk pbx on a digium PRI card |
16:21.56 | *** join/#asterisk wcselby (~wcselby@216.110.88.194) |
16:21.56 | Katty | considers some lunch |
16:21.59 | wcselby | o/ |
16:22.04 | Katty | hi wcselby |
16:22.05 | IBC_JKENNEY | after the router |
16:22.14 | wcselby | howdy |
16:22.21 | *** part/#asterisk guax (~guax@unaffiliated/guaxinim) |
16:22.49 | IBC_JKENNEY | I just wanted to know if asterisk could even be the kulprit. Since if i was having problems wouldn't i have problems on all numbers. |
16:23.46 | c0rnoTa | thansen: asterisk has specific channel for nortel phones |
16:24.46 | *** join/#asterisk cuco (~cuco@local.xorcom.com) |
16:24.47 | c0rnoTa | thansen: chan_unistim |
16:24.57 | c0rnoTa | use google for more information :)) |
16:25.35 | thansen | c0rnoTa: ok, I am...this particular model looks like it has sip. I'm just looking for general attitude toward nortel phones though |
16:25.54 | *** join/#asterisk Zambezi (Zulu@unaffiliated/zambezi) |
16:26.19 | *** part/#asterisk Yedidya (~chatzilla@host86-142-22-34.range86-142.btcentralplus.com) |
16:26.32 | c0rnoTa | i have seen in real life that Nortel phones works with asterisk :) |
16:26.56 | Katty | hmm. lunch. |
16:27.00 | Katty | suggestions? recommendations? |
16:27.27 | c0rnoTa | Tea + cake |
16:27.44 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
16:28.00 | Katty | how about something with protein |
16:28.04 | wcselby | pizza buffet |
16:28.08 | wcselby | chinese buffet |
16:28.18 | wcselby | thai food |
16:28.29 | c0rnoTa | hey, it's just a lunch |
16:28.30 | wcselby | ^^ all the places around today's client |
16:28.42 | c0rnoTa | wcselby: :)) |
16:28.43 | wcselby | ooh, and tex-mex |
16:28.55 | Katty | okay how about something HEALTHY |
16:28.57 | Katty | with protein |
16:29.04 | Katty | which totally cuts out buffets >.< |
16:29.15 | wcselby | i think there's a salad bar nearby in the grocery stor |
16:29.18 | wcselby | store* |
16:29.26 | wcselby | which is sort of like a buffet :P |
16:29.47 | [TK]D-Fender | Katty: Shish-taouk |
16:29.49 | wcselby | so you didn't bring anything with you today? |
16:29.55 | [TK]D-Fender | Katty: clearly time for Lebanese |
16:30.34 | c0rnoTa | O_o it's just a lunch, guys. Eat something and go for a work |
16:30.38 | c0rnoTa | :))))) |
16:30.51 | Katty | ooooh, salad |
16:30.53 | Katty | OoOOooh |
16:30.56 | Katty | plots |
16:31.44 | [TK]D-Fender | Katty: Lots of great little Lebanese places here... $10 feast... |
16:31.46 | *** join/#asterisk Woody2143 (~Woody2143@machine76.Level3.com) |
16:32.34 | c0rnoTa | and i'm waiting for a dinner |
16:32.52 | c0rnoTa | it's 7:32PM |
16:33.16 | Katty | waiting for a dinner? |
16:33.30 | c0rnoTa | yeah.. |
16:33.31 | Katty | you mean you're waiting for it to get done cooking? |
16:33.51 | Katty | what are you cooking? |
16:34.16 | *** join/#asterisk lanning (~lanning@208.87.235.224) |
16:34.17 | c0rnoTa | no. just wait for a time, when i'll be at home and eat something :) |
16:34.39 | Katty | ahh |
16:35.59 | c0rnoTa | all what i'm cooking now is just a SESSION cookie |
16:38.46 | c0rnoTa | So, can anybody now answer on my question? :) 7 hours ago there was no answer couse you all sleeped |
16:38.57 | c0rnoTa | an anybody tell me how i can count channels on specific extension at moment. For example, i have extension _9XXXXXXX and i want to know how much channels now are on that extension? |
16:38.59 | wcselby | what was teh question? |
16:39.25 | c0rnoTa | 'core show channels' show extens like 91234567 or 97654321 not "_9XXXXXXX". And i couldn't select which of them use "_9XXXXXXX" and which "_97.". Only using analysing process in my mind i could divide it. So, i want to divide it automaticly in AGI script. |
16:39.40 | c0rnoTa | speaking in global of my task, i want to limit abilities of outgoing calls through trunks on each exten. |
16:40.17 | wcselby | use COUNT()? |
16:41.20 | wcselby | sorry, I think I mean GROUP |
16:41.42 | *** join/#asterisk cod3hax0r (~Boyperwis@121.97.56.48) |
16:41.45 | cod3hax0r | hi guys |
16:41.45 | wcselby | man |
16:41.48 | wcselby | my brain isn't work yet |
16:41.50 | cod3hax0r | good day to all |
16:42.00 | wcselby | i'll look c0rnoTa, I think I know what you need |
16:42.27 | cod3hax0r | i need help... im getting SIP/2.0 401 Unauthorized |
16:42.35 | [TK]D-Fender | c0rnoTa: "core show function GROUP_COUNT" |
16:42.36 | c0rnoTa | wcselby: so, i can set group in each channel dialed throu exten, then group_count before dial this exten again, right? |
16:43.28 | c0rnoTa | [TK]D-Fender, wcselby! Thanks, guys! |
16:43.31 | *** join/#asterisk mort_gib (~mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
16:43.39 | c0rnoTa | it should work :) |
16:43.51 | wcselby | c0rnoTa, yeah, create a process that first checks if the group_count is below your threshhold, if it is, allow the call and add to the group, then if it's above your threshhold, deny the call. |
16:44.10 | c0rnoTa | wcselby: thanks a lot :)) |
16:44.28 | c0rnoTa | at the end of the day, i'v found an anser :) |
16:44.31 | c0rnoTa | answer |
16:45.12 | c0rnoTa | and i have no ideas before, either global vars :) |
16:46.50 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
16:47.02 | Katty | stares at clock |
16:47.21 | ecrane | http://www.theregister.co.uk/2010/02/03/voip_hacker_guilty/ |
16:49.08 | eppigy | stares at Katty |
16:49.38 | *** join/#asterisk crazybyte (~crzp@unaffiliated/crazypenguin/x-000001) |
16:50.05 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.157) |
16:52.41 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
16:58.29 | wcselby | i love how the guy made millions, and only paid his accomplice 20,000 |
16:58.57 | wcselby | referring to ecrane's register article |
16:59.32 | *** join/#asterisk soman (~somnath@stargate.starnet.fi) |
17:02.46 | *** join/#asterisk comradeb14ck (~comradeb1@72.37.252.50) |
17:03.51 | *** join/#asterisk kalib (~lkhlui@osiris.aspec.com.br) |
17:04.28 | kalib | What does it means when I receive in my CLI: "Remote UNIX connection | Remote UNIX disconnected |
17:04.47 | kalib | ? |
17:04.57 | Chainsaw | kalib: It means a command was run using asterisk -rx (in most circumstances). |
17:05.10 | kalib | oh.. thanks ;] |
17:05.15 | Chainsaw | kalib: So if you have bash scripts or other magic running in the background, that's probably it :) |
17:06.22 | *** join/#asterisk tbenson (~tbenson@c-67-174-228-93.hsd1.ca.comcast.net) |
17:07.08 | *** join/#asterisk corretico (~laguilar@201.201.46.106) |
17:09.19 | tbenson | Is there a simple way to spawn a new channel ID from within the h extension of a context? I am attempting to dial out after a call hangs up. I had been using a call file created from within the dialplan but am not sure how to identify the channel asterisk creates for the call file so i can examine the CHANNELSTATUS to handle exceptions. |
17:11.41 | *** join/#asterisk jameswf (~james@unaffiliated/jameswf-home) |
17:12.28 | ManxPower-work | tbenson: you should not do dialing out in your 'h' exten. |
17:12.47 | voipmonk | heheh |
17:12.48 | ManxPower-work | tbenson: if you want to post process after the callEE has hung up see the "g" option to dial. |
17:13.12 | ManxPower-work | There's no point in dialing after the callER hangs up. |
17:14.02 | *** join/#asterisk yahh (~root@122.169.77.74) |
17:15.08 | yahh | hello guys.. |
17:15.13 | yahh | i have a query |
17:15.56 | comradeb14ck | sup |
17:16.13 | Corydon76-dig | ~ask |
17:16.14 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
17:17.01 | yahh | i am using 2 - quad processor |
17:17.06 | yahh | asterisk is running on it |
17:17.10 | black | ok |
17:17.26 | yahh | right now cpu usage is going upto 60% |
17:17.35 | black | ok |
17:17.38 | yahh | for asterisk |
17:17.39 | *** join/#asterisk bklang (~bklang@tesla.alkaloid.net) |
17:17.41 | c0rnoTa | it's ok |
17:17.53 | c0rnoTa | flood of qualify messages ? |
17:17.55 | yahh | so can i go to up to 800% as it is 2 - quad |
17:18.07 | black | yahh, what operating system are you using? |
17:18.14 | yahh | centos |
17:18.18 | black | 32 or 64 bit? |
17:18.22 | yahh | 64 |
17:18.40 | black | you won't get 800% cpu, asterisk doesnt support multiple cores (afaik) |
17:18.56 | p3nguin | Was there a question involved in this story? |
17:19.44 | yahh | ohh so max i can go upto 100% only is it? |
17:20.36 | Corydon76-dig | black: Uh, it doesn't? |
17:20.43 | c0rnoTa | black: is it only on 64bit systems |
17:20.43 | c0rnoTa | ? |
17:20.48 | cod3hax0r | hello |
17:20.59 | cod3hax0r | can anyone help me with sip unauthorized errors? |
17:21.00 | p3nguin | corydon76-dig: Is asterisk a multi-threaded application? |
17:21.14 | Corydon76-dig | p3nguin: Yes, it most certainly is |
17:21.31 | p3nguin | Then I guess he can have 800% CPU usage! |
17:21.51 | voipmonk | use the right password/username combination cod3hax0r :) - you have some debug to show ( use pastebin.ca ) |
17:22.09 | yahh | black: pls give your advise |
17:22.13 | yahh | i mean input |
17:22.20 | black | Corydon76-dig knows way more than I do. |
17:22.23 | black | He is a developer. |
17:22.36 | p3nguin | I used to run it on a quad core, but I never remember checking threads. I since moved to a single core, since I have such a low load -- no since in wasting a quad core system. |
17:22.38 | black | I didn't think that asterisk ran on multiple cores, but I could be wrong. |
17:22.50 | black | Also, just because somethig is multithreaded doesn't mean it will run on multiple cores. |
17:22.59 | Corydon76-dig | There is one place that I can recall where Asterisk won't use more than one core: a fairly old version of FreeBSD, but I'm fairly certain they've fixed that since |
17:23.26 | yahh | okay |
17:23.36 | p3nguin | Maybe that FreeBSD didn't have an SMP kernel. :) |
17:23.51 | Corydon76-dig | But that's a limitation of the OS kernel, not Asterisk |
17:24.04 | Qwell | black: it's a multi-threaded application.. why wouldn't it? |
17:24.06 | c0rnoTa | cod3hax0r: what's your problem? |
17:24.15 | Corydon76-dig | p3nguin: No, it was that FreeBSD did not allow the threads of a single process to execute on more than one core concurrently |
17:24.35 | Corydon76-dig | p3nguin: you could run multiple processes concurrently, just not the threads of a single process |
17:24.46 | p3nguin | That's kind of an interesting feature(?) to have, I guess. |
17:24.47 | Qwell | black: sounds like nonsense your former employer fed people. :) |
17:25.02 | black | Qwell, I said I didn't know. |
17:25.04 | black | Nobody told me this. |
17:25.16 | Qwell | fair enough ;) |
17:25.20 | black | Awesome if it does though, good work. |
17:25.25 | *** join/#asterisk ch1nawhyte (~dliu@aker.tobi-office.com) |
17:25.27 | Qwell | but yeah, any app that has multiple threads will run on multiple cores |
17:25.42 | yahh | correct |
17:25.45 | Corydon76-dig | black: Thank the kernel developers. Wasn't anything we did, other than to use proper mutexes |
17:25.48 | Qwell | if it didn't, we wouldn't need any locking |
17:25.48 | yahh | make sence |
17:26.11 | Corydon76-dig | Qwell: that's not completely true, either |
17:26.24 | Qwell | Corydon76-dig: generalization |
17:26.36 | Corydon76-dig | We wouldn't need locking if the kernel used cooperative multitasking, instead of preemptive multitasking |
17:27.01 | cod3hax0r | <c0rnoTa> cod3hax0r: what's your problem? --> im getting SIP 2.0/unathorized when i do sip debug |
17:27.28 | c0rnoTa | but you sent correct username/secret is it? |
17:27.40 | yahh | so i think summary is... asterisk can go upto 800% for 2 quad |
17:27.41 | c0rnoTa | cod3hax0r: but you sent correct username/secret is it? |
17:27.44 | yahh | right? |
17:28.01 | Corydon76-dig | yahh: I think you mean 8.0 load average, but yet |
17:28.04 | Corydon76-dig | yes |
17:28.07 | yahh | yes |
17:28.12 | c0rnoTa | yes |
17:28.49 | bmoraca_work | 8.0 load average? that's ridiculously high |
17:29.04 | cod3hax0r | <c0rnoTa> cod3hax0r: but you sent correct username/secret is it? --> yes sir did |
17:29.08 | bmoraca_work | that's 8 processes waiting every second for a chance to run their work...that's too much |
17:29.09 | *** join/#asterisk Victor_Yure (~victor@unaffiliated/victoryure/x-837844) |
17:29.11 | c0rnoTa | cod3hax0r: try to set pedantic=yes in general section of sip.conf. |
17:29.15 | yahh | if i look top., there is %CPU for asterisk |
17:29.18 | Corydon76-dig | bmoraca_work: that's maximum for 8 cores, without delaying any thread execution |
17:29.30 | yahh | that could reach upto 800% , isn't it |
17:29.48 | bmoraca_work | Corydon76-dig: ahhh...8 cores. that's somewhat ok, i suppose. |
17:30.24 | cod3hax0r | <c0rnoTa> cod3hax0r: try to set pedantic=yes in general section of sip.conf. --> just set it right now |
17:30.36 | Corydon76-dig | bmoraca_work: read up. :-) |
17:30.47 | cod3hax0r | <c0rnoTa> cod3hax0r: try to set pedantic=yes in general section of sip.conf. --> whats next? |
17:31.06 | c0rnoTa | cod3hax0r: sip reload in cli and go on |
17:31.10 | c0rnoTa | try to auth |
17:31.23 | bmoraca_work | Corydon76-dig: yeah, i suppose coming in to the middle of a conv. isn't exactly enlightening :P |
17:31.46 | p3nguin | Why would asterisk cause an 8.0 load average? If asterisk is using 800% CPU and you have 100 processes waiting to use the processor, the load average is going to be more than 8.0. |
17:32.04 | yahh | Corydon76-dig: is there any difference for 8.0 load and %CPU for asterisk process |
17:32.06 | yahh | ? |
17:32.12 | c0rnoTa | cod3hax0r: either i have no fast idea what it could be. need to see sip debug and look on SIP pockets exchange |
17:32.27 | yahh | it is similar to the 800% cpu right? |
17:32.34 | bmoraca_work | yahh: load average is impossible to equate directly to % cpu usage |
17:32.35 | Corydon76-dig | yahh: Yeah, one is an average over a period of time, and the other is a snapshot in time |
17:33.15 | Corydon76-dig | Well, % cpu usage is kind of a misnomer. |
17:33.22 | tzafrir_laptop | yahh, yeah, there is |
17:33.57 | tzafrir_laptop | also, %cpu is for a shorter period. load avarage is for the last minute |
17:33.58 | Madkiss | I have a problem with asterisk 1.6. Somebody is coming inside from an official IP, and actually I see the packets flying in, but: asterisk doesn't care at all. |
17:34.01 | *** join/#asterisk albertoandrade (~albertoan@189.58.23.130.dynamic.adsl.gvt.net.br) |
17:34.28 | Corydon76-dig | CPUs are only either in use or not... You can't really use 50% of a CPU... |
17:35.01 | yahh | hmm.. |
17:35.07 | p3nguin | You can certainly use the CPU 50% of the time, though. |
17:35.08 | *** join/#asterisk gelpg (~chatzilla@94-21-99-19.pool.digikabel.hu) |
17:35.11 | Madkiss | asterisk definetely listens on the corresponding port. |
17:35.45 | yahh | okay.. so in other words |
17:36.11 | yahh | asterisk can use 800% time with 2 quad |
17:36.24 | bmoraca_work | Madkiss: perhaps if you had a debug log of the failed call...maybe we could help you. but from your description of the problem, there's nothing we can do, because the description of the problem doesn't make any sense |
17:36.35 | cod3hax0r | still the same |
17:36.49 | mort_gib | Madkiss: /etc/init.d/iptables stop |
17:36.56 | Madkiss | bmoraca_work: it's not about the call. it's about registering. registering fails. |
17:37.05 | Madkiss | mort_gib: iptables it not even installed on this machine. |
17:37.11 | bmoraca_work | Madkiss: ok, so show a failed registration debug. |
17:37.14 | p3nguin | You need not register to make a call. |
17:37.37 | mort_gib | Madkiss: 99% of those probs are related to a firewall.... |
17:37.44 | Madkiss | SIP/2.0 401 Unauthorized |
17:38.02 | Madkiss | hu. |
17:38.08 | bmoraca_work | Madkiss: that's not a debug. pastebin an ENTIRE debug from the failed registration. |
17:38.16 | yahh | asterisk can use 800% time with 2 quad (if we assume that no other porcess is not running)if am not wrong |
17:38.31 | Warp4 | anyone have any luck doing any php scripting to communicate with the AMI? |
17:38.31 | yahh | ? |
17:38.51 | bmoraca_work | yahh: it can use 100% of each core. that's not 800%, though. |
17:39.10 | mort_gib | Madkiss: Then you might want to rephrase the "But asterisk dosen't care at all" |
17:39.10 | p3nguin | 8 cores, some "monitor" apps will report 800%. |
17:39.30 | mort_gib | Madkiss: Asterisk cares, that phone/user is just not authenticated.... |
17:39.35 | gelpg | hi |
17:39.37 | gelpg | I need some help to install the asterisk with misdn driver |
17:39.42 | gelpg | i already installed the mISDN and mISDNuser drivers and tools |
17:39.44 | yahh | okay |
17:39.50 | gelpg | the misdn_info shows the card (Digium B410P) |
17:39.55 | gelpg | but I cannot configure the asterisk to enable the misdn channel |
17:40.05 | mort_gib | gelpg: Why misdn?? |
17:40.19 | gelpg | because with dahdi it doesn't work |
17:40.19 | *** part/#asterisk c0rnoTa (~c0rnoTa@178.176.198.228) |
17:40.28 | *** join/#asterisk c0rnoTa (~c0rnoTa@178.176.198.228) |
17:40.35 | gelpg | it says the bri signalling doesn't supported |
17:40.36 | mort_gib | gelpg: Not correct |
17:40.44 | c0rnoTa | cod3hax0r: use sip debug to find answer |
17:40.56 | mort_gib | gelpg: That's different, you need to install libpri |
17:41.05 | mort_gib | before dahdi |
17:41.19 | gelpg | I already did it |
17:41.44 | mort_gib | Ok, I have it running with 2 different clients.... |
17:41.52 | mort_gib | So |
17:41.56 | Madkiss | Via: SIP/2.0/UDP 192.168.92.55:2 <= can somebody tell me where that 192-IP is actually coming from? |
17:42.25 | mort_gib | point-to-multipoint |
17:42.32 | cod3hax0r | still getting unauthorized using sip debug |
17:42.40 | cod3hax0r | im behind a cisco 800 router |
17:43.33 | bmoraca_work | Madkiss: it's probably the internal IP address of the device trying to register with you, BUT WITHOUT A FULL SIP DEBUG, IT IS IMPOSSIBLE TO SAY. |
17:44.03 | gelpg | mort_gib: yes, that's what I need |
17:44.14 | Madkiss | bmoraca_work: take it easy on me. I am handling two totally crazy chicken right now yelling at me because they can not phone, thinking they are way smarter than me, I have more than enough shit to deal with ... ;) |
17:44.36 | mort_gib | Madkiss: He is right tho |
17:44.43 | Madkiss | I know he is |
17:44.54 | mort_gib | :-) |
17:44.55 | bmoraca_work | Madkiss: that's fine, but until you provide a FULL SIP debug of the failed registration, there is no possible way I can help you |
17:45.32 | mort_gib | Madkiss: Check very CAREFULLY usernames and passwords |
17:46.21 | Madkiss | http://paste.debian.net/58720/ |
17:46.23 | Madkiss | there you go. |
17:46.29 | ManxPower-work | cod3hax0r: you'll always get unauthorized, then the client will retry with auth info |
17:46.44 | bmoraca_work | Madkiss: I suspect that the issue is one of three things: 1) There is an ALG on the user's NAT router that is improperly configured to work with asterisk and the phone; 2) Asterisk is not set to look for a natted phone; 3) the phone is configured to look for asterisk as if it were not a NAT. |
17:46.44 | mort_gib | gelpg: So, latest version of Dahdi, and install libpri BEFORE dahi |
17:46.49 | bmoraca_work | Madkiss: what model is the phone and the user's firewall? |
17:48.19 | Madkiss | hold on for a second. |
17:49.48 | bmoraca_work | Madkiss: additionally, you did not show Asterisk's failure. you didn't give me the entire debug and you obfuscated certain parts that are pretty useful in debugging this. |
17:51.13 | bmoraca_work | Madkiss: other things which might be helpful would be the sip peer definition from sip.conf (with ONLY the secret obfuscated) |
17:54.06 | cod3hax0r | its a funny thing because i can register to another sip provider but not to my asterisk server |
17:54.30 | *** join/#asterisk xpot-mobile (~xpot@173-14-232-121-Utah.hfc.comcastbusiness.net) |
17:55.28 | *** join/#asterisk mnick86 (~mnick86@95-90-246-116-dynip.superkabel.de) |
17:56.50 | *** join/#asterisk SuPrSluG (~SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
17:57.00 | wcselby | Madkiss - what kind of phones, what's the setup like? |
17:57.39 | ManxPower-work | Maybe Madkiss should come back when he has time to concentrate on the problem he is having. |
17:57.49 | wcselby | i.e - phones at people's homes behind home nat / routers, asterisk at work behind firewall or on a public IP, etc? |
17:58.11 | wcselby | i may have missed all that, I came back in the middle of his issues |
17:58.36 | cod3hax0r | heres my sip debug output http://pastebin.com/m55d54ef1 |
17:58.59 | wcselby | cod3hax0r - that's only one transmission |
17:59.08 | wcselby | there's more in a failed auth attempt |
17:59.28 | wcselby | everytime you register a phone with a secret, you'll always get a 401 unauth first, then the phone is supposed to resubmit with the proper auth |
18:00.12 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.172.179.dsl.dyn.forthnet.gr) |
18:00.46 | cod3hax0r | what will i need? |
18:00.47 | Ad-Hoc | hi ppl |
18:00.50 | wcselby | if it's not sending the second auth with the correct auth info, either it's entered incorrectly in the phone itself, or it isn't receiving the 401 unauth message from asterisk (which can happen because of many issues) |
18:00.56 | ManxPower-work | The way digest auth (both SIP and in HTTP) works. Client connects, gets rejected with an auth required and is sent a key (called salt) to use to encrypt the response along with the password. |
18:02.12 | wcselby | cod3hax0r - you need to show us everything, from the first REGISTER statement (from the phone -> asterisk), the UNAUTH response from asterisk -> phone, and any response the phone may be sending back to asterisk. |
18:02.44 | wcselby | also, what's your network setup? is the phone or the server behind a firewall, on separate networks, the same, etc? |
18:03.22 | cod3hax0r | here is the other http://pastebin.com/m3c1877bc |
18:03.29 | cod3hax0r | my server is collocated |
18:03.38 | cod3hax0r | my zoiper softphone is behind a cisco 800 router |
18:04.18 | wcselby | what comes after the last sip message? |
18:04.34 | *** join/#asterisk etfonhomey (~etfonhome@74-143-192-74.static.insightbb.com) |
18:04.40 | wcselby | you've only showed the beginning of the sip negotiation |
18:04.49 | Katty | hi |
18:04.57 | wcselby | o/ kaldemar |
18:04.59 | wcselby | erm |
18:05.01 | wcselby | o/ Katty |
18:05.02 | wcselby | :) |
18:05.10 | Katty | that's what you get for tab completing |
18:05.22 | wcselby | it is indeed |
18:05.30 | Katty | ;> |
18:05.43 | ManxPower-work | cod3hax0r: have you done a "service iptables stop" |
18:05.51 | Katty | i acquired sparkling water on the way back |
18:05.59 | tbenson | ManxPower-Work firgure out a workaround (had it typed in not to bother, but took a call). just into lots of code for this application so was getting lost for a moment, just making my AGI create the call file dial from an alternate context that the hangup code and channelstatus will be accessible for the call in dialplan, rather then try to pull it back into the current dialplan. This h exten is also not in default, its a custom c |
18:06.00 | tbenson | ontext for alert notification to engineers. Thanks all. |
18:06.42 | Katty | tries sparkling water for the first time. |
18:06.51 | wcselby | i've never liked sparkling water |
18:06.52 | Katty | hmm. |
18:06.56 | Katty | needs ice. brb |
18:06.58 | wcselby | i don't mind sparkling grapejuice |
18:07.08 | wcselby | or sparkling wine, for that matter |
18:07.27 | Madkiss | Looking for 3946 in from-sip (domain 10.9.11.244) -- erm. thing is ... 3946 is in another domain, actually |
18:07.40 | cod3hax0r | yes ive done that service iptables stop |
18:07.54 | cod3hax0r | funny thing is i can connect from my other branch in canada |
18:07.57 | Madkiss | can I tell asterisk in sip.conf that 3948 is in another domain? |
18:07.58 | cod3hax0r | but i cant connect from here |
18:08.48 | Katty | tries again, with ice. |
18:08.52 | wcselby | cod3hax0r - did you understand what I meant about the last pastebin being incomplete? |
18:08.54 | [TK]D-Fender | Madkiss: * is NOT a PROXY |
18:09.28 | Katty | wcselby: not too shabby |
18:09.36 | Katty | wcselby: this has lemony flavors added |
18:09.49 | wcselby | ahh |
18:09.54 | wcselby | so more like a sprite or 7up? |
18:10.00 | Katty | not really |
18:10.04 | wcselby | lol |
18:10.04 | Katty | not that strong |
18:10.09 | Katty | just, lightly lemmonied |
18:10.10 | wcselby | i dunno, i may have to try it |
18:10.18 | Katty | it's called Perrier Lemon |
18:10.23 | wcselby | ahh |
18:10.26 | wcselby | never like perrier |
18:10.38 | wcselby | but then, i never tried their flavored versions either |
18:11.02 | Katty | it's different |
18:11.22 | Madkiss | uh. I see the problem. |
18:11.40 | Katty | checks on the critters |
18:11.43 | wcselby | was annouce-position added to queues.conf in 1.6.x? |
18:12.01 | wcselby | or is it supposed to be available in 1.4.x? |
18:12.56 | ManxPower-work | wcselby: what does queues.conf.sample say? |
18:13.23 | ManxPower-work | From YOUR Asterisk, not from some web site like voip-info.org |
18:13.31 | wcselby | i don't see it in there, i just see it on the wiki (don't shoot me please.... ;) ). that's why I thought I'd ask. |
18:13.38 | wcselby | haha |
18:13.46 | *** join/#asterisk lanning (~lanning@208.87.235.224) |
18:13.46 | wcselby | yeah, it's not listed in my queues.conf.sample |
18:13.58 | wcselby | which is a shame |
18:14.00 | Katty | everyone sure is hostile today |
18:14.43 | [TK]D-Fender | Katty: LIES. |
18:14.44 | ManxPower-work | "The Internet crashed today, plunging the nation into productivity." <-- The Onion |
18:15.19 | Katty | ^- hostility |
18:15.19 | [TK]D-Fender | Katty: And I'll &#^$ING kill anyone who says otherwise |
18:15.19 | wcselby | lol |
18:15.19 | Katty | ^- reeks of hostility |
18:15.37 | ManxPower-work | that's not a nice think to say about wcselby |
18:15.51 | wcselby | ;) |
18:16.04 | *** join/#asterisk benngard (~benngard@90-230-92-67-no148.tbcn.telia.com) |
18:16.07 | Katty | puts up No Free Radicals Allowed sign |
18:16.37 | Katty | even my lunch is being hostile |
18:16.43 | Katty | :< |
18:17.23 | bmoraca_work | mmm...lunch...too bad it's a distant 105 minutes away |
18:17.24 | Katty | something which feels similiar to heartburn |
18:17.34 | wcselby | my morning coffee is being pretty hostile to my stomach at the moment |
18:17.42 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
18:18.24 | Katty | :< |
18:18.46 | Katty | let's write today off as a bad idea and go home |
18:18.47 | bmoraca_work | way OT, but has anyone here ever done those edible arrangements? |
18:18.53 | Katty | yes |
18:19.06 | Katty | i got the chocolate dipped strawberry plant |
18:19.11 | cod3hax0r | here is the full log http://pastebin.com/m3b7d1dd5 |
18:19.11 | bmoraca_work | any good? |
18:19.15 | Katty | mhmm |
18:19.16 | Katty | spensive tho |
18:19.55 | Katty | i'm getting ryan a case of beer for valentines day |
18:20.39 | bmoraca_work | i'm looking for something for my wife...already cooking her filet mignon, but she'll want flowers or something...and i'm thinking edible arrangements could be both pretty and useful for other reasons |
18:20.40 | wcselby | cod3hax0r - looks like your phone isn't receiving the 401 Unauth response |
18:21.07 | wcselby | cod3hax0r - what's the network setup between * and phone? |
18:21.16 | Katty | For Other Reasons |
18:21.17 | ManxPower-work | <--- Transmitting (NAT) to 61.5.156.146:28659 ---> |
18:21.18 | Katty | (tm) |
18:21.20 | [TK]D-Fender | cod3hax0r: I don't see configs to match... |
18:21.21 | ManxPower-work | That concerns me |
18:21.28 | [TK]D-Fender | ManxPower-work: It shouldn't |
18:21.33 | Katty | ManxPower-work: you concern me. |
18:21.40 | Katty | ManxPower-work: how are you feeling? |
18:21.42 | bmoraca_work | bad thing is that they don't deliver to where we are |
18:21.51 | Katty | bmoraca_work: they allow pickup |
18:21.57 | ManxPower-work | Katty: coming from Squirrel Girl..... |
18:21.59 | bmoraca_work | yeah...12 miles away |
18:22.18 | Katty | ManxPower-work: razzleberries. |
18:22.36 | Katty | bmoraca_work: ahhhh |
18:22.45 | Katty | bmoraca_work: you could always make your own |
18:23.22 | bmoraca_work | Katty: when everyone else went to the ARTISTIC class in elementary school, I went to the AUTISTIC class. i can't even draw my name on a peice of paper :P |
18:23.32 | cod3hax0r | its zoiper softphone |
18:23.40 | cod3hax0r | network is behind a cisco 800 router |
18:23.42 | cod3hax0r | basic nat |
18:23.59 | Katty | bmoraca_work: if you can do steak on a stick |
18:24.04 | Katty | bmoraca_work: you can do edible arrangements |
18:24.12 | wcselby | hmmm, maybe i'm not seeing something obvious, but - is there a way to play a sound file to the caller when they enter a queue? before any hold music or anything like that? Or at the very least, let them know their position in queue and potential hold time? |
18:24.55 | Katty | http://www.ediblearrangements.com/fruit-bouquet-detail.aspx?ArrangementID=318&StoreID=0&OrderType=&SelectedDate=&AreaName=&set=true <- bmoraca_work |
18:25.16 | wcselby | cod3hax0r - what's the network like on the * side? is it behind a firewall / router? does it have a public IP address? |
18:25.32 | cod3hax0r | it has a public ip address |
18:25.33 | bmoraca_work | cod3hax0r: ManxPower-work is right...your phone is never receiving the 401 Unauthorized. Because of that, it never tries to register again with credentials (or it's not configured to). check the config of your softphone. it's not a NAT problem because it's not being retransmitted. |
18:26.08 | cod3hax0r | the server is on a colocated public ip address in the internet |
18:26.23 | Katty | bmoraca_work: just get some big long wooden skewer things and put a bunch of fruit chunks on it |
18:26.24 | bmoraca_work | katty: I kind of liked this one: http://www.ediblearrangements.com/fruit-bouquet-detail.aspx?ArrangementID=316&StoreID=579&OrderType=2&SelectedDate=02/14/2010&AreaName=95382&set=true |
18:26.24 | cod3hax0r | and the workstation is behind a t1 with a public ip address |
18:27.01 | Katty | it's still fruit on a stick |
18:27.07 | bmoraca_work | lol |
18:27.13 | Katty | take skewer |
18:27.14 | Katty | apply fruit |
18:27.21 | Katty | place in pretty colorful mug |
18:27.28 | Katty | or bowl or vase, whatever |
18:27.35 | Katty | drizzle with chocolate, chill in fridge |
18:27.42 | bmoraca_work | Katty: I also kindof liked this: http://products.proflowers.com/flowers/HugsandKisses-5395?viewpos=4&trackingpgroup=vca&ref=HomeNoRef&pagesplit= |
18:28.08 | Katty | hmm |
18:28.10 | bmoraca_work | i suspect she wants roses though...being that she is a girl and my wife |
18:28.23 | Katty | ryan's never gotten me flowers. |
18:28.25 | Katty | pouts |
18:30.10 | *** join/#asterisk Badrobot- (~badrobot@cpe-76-173-229-89.socal.res.rr.com) |
18:30.29 | bmoraca_work | screw it...a dozen roses it is. |
18:30.34 | Naikrovek | i got my wife flowers once |
18:30.38 | Naikrovek | cat ate them and died |
18:30.45 | Naikrovek | "you don't ever buy me flowers again" she said |
18:30.47 | Naikrovek | okay... |
18:30.47 | bmoraca_work | lol |
18:30.53 | bmoraca_work | easy way out |
18:30.56 | Katty | bmoraca_work: a reasonable choice |
18:31.17 | Katty | i'll have yet another year of....no valentines day |
18:31.18 | wcselby | Naikrovek - my wife has never, and will never, tell me anything even remotely like that |
18:31.44 | benngard | now i am lost... shouldnt ${CALLERID(DNID)} be set to callee number when u execute a "channel originate ..." command? |
18:31.46 | wcselby | the closest may be if I just simply get the wrong kind of flowers, she may say, don't get me THOSE kind of flowers again. but really, you can't go wrong with roses, at least for my wife |
18:32.30 | Naikrovek | yeah wives are all different, just like like regular people |
18:32.55 | Katty | maybe i'll send myself flowers this year |
18:32.58 | wcselby | lol |
18:33.01 | Naikrovek | aww |
18:33.01 | benngard | its empty both from cli and ami, and cdr post dst is empty to |
18:33.28 | corretico | hello people |
18:33.33 | Katty | and THEN i'll take myself out to dinner :> |
18:33.54 | Katty | have a few drinks, strike up a conversation |
18:35.09 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
18:35.09 | Naikrovek | kinda like Tyler Durden. forced against your own nature so much your Id becomes its own personality |
18:35.34 | wcselby | but hopefully without all the mass murder and psychosis and stuff |
18:35.41 | Naikrovek | he never killed anyone |
18:35.49 | Naikrovek | one guy died, but he was shot by cops |
18:35.58 | Naikrovek | but plenty of psychosis |
18:35.58 | wcselby | he planned on blowing up a bunch of buildings and stuff though |
18:36.15 | wcselby | i've read teh book, but it's been a LONG time.... |
18:36.16 | Naikrovek | wcselby: yes, and he did, but security and maintenance were all in on it, and made sure the buildings were celar |
18:36.27 | Naikrovek | s/celar/clear/ |
18:36.35 | Katty | what an awful job |
18:36.48 | Katty | get up in the morning, have a cup of coffee |
18:36.51 | Katty | disarm a building |
18:36.55 | Katty | lunch |
18:36.58 | Naikrovek | the book and movie are a bit different |
18:37.01 | *** join/#asterisk _omer (~omer@119.152.140.100) |
18:37.02 | Naikrovek | i recall Marla's mother in the book |
18:37.05 | Naikrovek | not in the movie |
18:37.15 | Katty | can you imagine hitler's diary? |
18:37.22 | Naikrovek | i can't imagine him keeping one |
18:37.24 | Katty | death death death death death |
18:37.26 | n3hxs | is no longer away : Gone for 2 hours 47 minutes 37 seconds |
18:37.28 | Katty | lunch |
18:37.30 | _omer | hi |
18:37.31 | Katty | death death death death death |
18:37.34 | Katty | shower |
18:37.39 | Nugget | hugs hugs hugs hugs hugs |
18:37.40 | jaytee | I saw the recent Internet Hitler rant about the iPad. funny as usual |
18:37.42 | Katty | death death death death death, painting |
18:37.43 | Naikrovek | 1pm: Manic. |
18:37.46 | dlynes_laptop | Katty, you're quite morbid and depressed this year? |
18:37.49 | Naikrovek | 2pm: depressed |
18:37.52 | Naikrovek | 3pm: manic |
18:37.53 | Naikrovek | etc. |
18:37.55 | dlynes_laptop | Katty, why aren't you having valentine's day? |
18:37.57 | Katty | Naikrovek: yesh |
18:38.06 | Katty | dlynes_laptop: because ryan doesn't want to |
18:38.14 | dlynes_laptop | Katty, that's a dumb reason |
18:38.15 | Katty | dlynes_laptop: he thinks it's all commericalized and stuff |
18:38.25 | jaytee | he's such a romantic |
18:38.28 | dlynes_laptop | no kidding |
18:38.30 | Katty | yeah, totally |
18:38.41 | dlynes_laptop | Katty, give him a kick :) |
18:39.10 | Katty | heh |
18:39.21 | dlynes_laptop | my sister-in-law can't pull that one off |
18:39.22 | Katty | that man does do a single thing he doesn't wanna do |
18:39.26 | wcselby | Katty - you need to explain to him that it doesn't matter what he thinks about Valentine's Day - it's what matters to YOU. |
18:39.30 | Katty | s/does/doesn't/ |
18:39.40 | [TK]D-Fender | Katty: He's right. Is he at least acceptably romantic acceptably often enougha s it is? |
18:39.49 | dlynes_laptop | She was dumb enough to get married on Valentine's day :0 |
18:39.50 | Katty | [TK]D-Fender: yes. |
18:40.01 | wcselby | i feel the same way, but that doesn't mean I can get away with not doing anything for my wife |
18:40.14 | Katty | wcselby: well i'm not your wife either |
18:40.16 | wcselby | although we usually schedule something on a day OTHER than vday |
18:40.18 | [TK]D-Fender | Katty: then good for that. He should still do a little SOMETHING... but that doesn't imply any extra special expense. |
18:41.16 | benngard | can some smrter than me explain whats wrong here: http://pastebin.com/d797a8309 ${CALLERID(DNID)} is empty but it dials my mobile |
18:41.46 | _omer | asterisk-addons help ... http://www.pastebin.org/86356 |
18:41.51 | thehar | Comcast to buy NGT. :( |
18:41.57 | Katty | wha'ts ngt |
18:42.11 | thehar | New global telecom |
18:42.14 | thehar | http://www.xchangemag.com/hotnews/exclusive-comcast-to-buy-ngt.html |
18:42.21 | thehar | a large large wholesaler |
18:42.29 | Katty | what's this mean |
18:42.59 | thehar | means comcast is on it's way to become the largest telcom provider |
18:43.19 | Katty | and this is somehow, bad, right? |
18:43.23 | thehar | comcast == devil |
18:43.24 | thehar | yes |
18:43.31 | Katty | ahh i see |
18:45.28 | [TK]D-Fender | benngard: exten => s,n,ExecIf($["${CALLERID(num)}"="0317998975"]?Set(CALLERID(all)=Magnus Benngard <317998975>)) |
18:45.39 | [TK]D-Fender | benngard: "all" != "dnid" |
18:45.58 | _omer | asterisk-addons help ... http://www.pastebin.org/86356 |
18:46.02 | _omer | ? |
18:46.34 | [TK]D-Fender | _omer: Clearly |
18:46.44 | wcselby | _omer did you install mysql after you ran ./configure? |
18:46.54 | wcselby | but before you ran make menuselect? |
18:47.20 | _omer | mysql was installed before asterisk-addons |
18:47.45 | _omer | even mysql was installed before I downloaded asterisk-addons |
18:48.24 | Katty | that's odd. |
18:48.29 | Katty | Major General is just sitting there. |
18:48.31 | Katty | staring. |
18:48.41 | Katty | and he's been sitting there, staring, for the better part of 5 minutes |
18:48.55 | *** join/#asterisk Victor_Yure (~victor@unaffiliated/victoryure/x-837844) |
18:49.07 | wcselby | _omer - did you install from source or using a package manager? |
18:49.19 | titter | Is there a way to force users to change their voicemail password during their first login? |
18:49.24 | _omer | I did yum install mysql |
18:49.36 | _omer | in CentOS 5.4 |
18:49.39 | wcselby | _omer - you also installed mysql-server and mysql-devel? |
18:49.50 | _omer | yes |
18:49.58 | _omer | but let me check mysql-devel |
18:50.30 | _omer | Package mysql-devel.i386 0:5.0.77-4.el5_4.1 set to be updated |
18:50.42 | _omer | yum is updating mysql-devel now |
18:50.51 | benngard | [TK]D-Fender: that not the problem, check what "exten => s,1,NoOp(${CALLERID(DNID)})" procudśe for output |
18:51.02 | benngard | produce* |
18:51.10 | [TK]D-Fender | benngard: that is looking at NUM, not DNID. NOT ThE SAME |
18:51.38 | [TK]D-Fender | exten => s,n,ExecIf($["${CALLERID(num)}"="0317998975"]?Set(CALLERID(all)=Magnus Benngard <317998975>)) <-- NOT DNID. |
18:51.51 | *** join/#asterisk davix (~davix@82.166.170.193) |
18:52.05 | wcselby | _omer - do a make distclean, make clean, then redo ./configure in your asterisk-addons source directory |
18:52.07 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
18:52.16 | _omer | ok |
18:52.20 | wcselby | _omer - after you've updated your mysql-devel, of course :) |
18:52.27 | _omer | yep :) |
18:52.30 | benngard | oki , will make it more simple and show u what i mean sec |
18:53.07 | [TK]D-Fender | benngard: You're talking about DNID, and the only ExecIf that matches isn't LOOKING at DNID |
18:56.15 | _omer | make distclean = DONE , make clean = DONE , ./configure = DONE |
18:56.26 | _omer | now ? |
18:56.32 | _omer | make menuselect? |
18:57.08 | wcselby | _omer - yes |
18:57.16 | wcselby | and let's see what it has to say about res_config_mysql |
18:57.20 | _omer | ok |
18:57.40 | benngard | http://pastebin.com/d3816f268 |
18:57.51 | _omer | It works!!! options are available |
18:57.56 | wcselby | :) |
18:58.01 | _omer | thanks alot !! :) |
18:58.02 | wcselby | glad I could help |
18:58.13 | _omer | :-D |
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18:59.07 | Alfio | hi, anyone here knows if there is some SIP or IAX2 cleints for blackberry? |
18:59.23 | wcselby | I know of sip clients for iPhone |
18:59.29 | wcselby | i'm pretty sure there are sip clients for bb |
19:00.19 | [TK]D-Fender | benngard: Still meaningless. This has nothingt o do with the other pastebin |
19:00.43 | knctrnl | There are no SiP or IAX2 clients for BB as of the last time I looked. |
19:00.50 | knctrnl | there are for Iphone and android |
19:00.56 | Alfio | ok |
19:01.02 | *** join/#asterisk opticy (~opticy@that.violates.us) |
19:01.22 | Alfio | i have two days looking into google and i cant see anything related to it |
19:01.30 | opticy | is it possible to route incoming calls based upon DFMTs entered by the calling party? |
19:02.11 | wcselby | opticy - you mean like an IVR? |
19:02.35 | benngard | [TK]D-Fender: i did "clean out" so it was easier to see the error |
19:03.09 | [TK]D-Fender | benngard: You are looking at DNID in one, and NUM in the other |
19:03.14 | [TK]D-Fender | benngard: NOT THE SAME |
19:03.20 | benngard | and the follow up is that if u originate a call from cli, u will get an emtp dst in cdr |
19:03.28 | opticy | wcselby: basically i only use asterisk for outgoing SIP/POTS calls; never for incoming POTS calls..i would like a wway that i could call my phone line from the outside via pots and have asterisk route the call only if it hears a sequence of DFMTs |
19:03.38 | [TK]D-Fender | beek: What don't you get? You show me a call where you PROVE what you are comparing just before doing the compare. |
19:03.58 | opticy | kinda like how it can detect a fax |
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19:04.55 | wcselby | opticy - once the call comes in, the users enter a DTMF code of some sort to route the call? |
19:04.58 | benngard | [TK]D-Fender: skip pastebin 1, look at pastebin 2, 2 calls 1 from cli, 1 from phone and they behave differnt |
19:05.07 | wcselby | route it within the system or back out on another line? |
19:05.38 | benngard | but both works, call from cli will not produce correct cdr, call from phone will do |
19:06.11 | [TK]D-Fender | benngard: Why are you even looking at DNID? |
19:06.31 | wcselby | opticy - if you're looking to password protect, check the Authenticate() app. If you're looking to route calls based on a menu presented to the user, check into making IVR's (Google). If you're looking to route a call back out on another line based on the DTMF input, check into DISA. |
19:06.31 | opticy | wcselby: basically if i were to call the line, and enter say "*22" while it was ringing, asterisk would answer the call and route it to somewhere (like to an IVR or something) |
19:06.33 | [TK]D-Fender | benngard: And an ORIGINATE-D channel is not a an inbound call |
19:06.46 | [TK]D-Fender | benngard: They didn't call IN... Asterisk called OUT |
19:07.00 | [TK]D-Fender | benngard: Apples 7 oranges |
19:07.05 | opticy | wcselby: authenticate sounds about right :) thx for the leads, i'll check them out |
19:07.11 | wcselby | _omer - opticy - asterisk has to answer the call first before it can handle the DTMF....unless you can do something with early media....but I've never worked with that |
19:07.20 | [TK]D-Fender | benngard: Feel free to update your records to correct this AFTER The FACT |
19:07.25 | wcselby | you know who I meant :) |
19:07.27 | n3hxs[away] | is now away - Reason : Auto-Away after 30 minutes |
19:07.40 | opticy | wcselby: on faxes, for instance, it can not answer it unless it detects a fax..right? |
19:08.01 | wcselby | opticy - my understanding is it won't detect that it's a fax until it's been answered. |
19:08.08 | opticy | ah ok |
19:08.09 | wcselby | so it has to answer it before it detects it's a fax. |
19:08.21 | opticy | perhaps that's where i'm a bit confused |
19:08.26 | wcselby | at least, that's how I've always done it. |
19:08.53 | *** join/#asterisk Micc (~quassel@c-98-225-57-96.hsd1.wa.comcast.net) |
19:09.07 | wcselby | like I said, I haven't ever played with early media. Or FAX over POTS, for that matter... FAX over PRI though, the call comes into specific numbers and is routed to fax destinations. we don't do asterisk's built-in fax detect. |
19:09.23 | wcselby | or I should say, none of my clients do built-in fax detection. they just setup extensions for fax. |
19:09.30 | wcselby | erm, DID's for fax. |
19:10.44 | benngard | [TK]D-Fender: i can make a call to another sip phone but the same error will occur, i cant trust the cdr.dst because dnid is not set |
19:11.07 | [TK]D-Fender | benngard: How can you have a Dialed Number ID when..... they didn't DIAL A NUMBER? |
19:12.16 | Katty | they used brain powers. |
19:12.19 | benngard | explain for me plz how cdr.dst will be empty, i did try to follow the code backward |
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19:12.53 | [TK]D-Fender | benngard: there is no DESTINATION. The call didn't come IN. * called OUT |
19:13.11 | [TK]D-Fender | benngard: there are no corners in a round room. |
19:13.14 | wcselby | is he looking for rdnis? |
19:13.43 | benngard | yes i se that in cdr, i see my 0317998975 as src as an empty dst |
19:14.15 | Katty | catches up on the news |
19:14.18 | [TK]D-Fender | benngard: What you are looking for does not exist. Nobody dialed that number to start the call. * jsut called out on its own. |
19:14.27 | rnp | Will a propperly provisioned asterisk system hooked up to an online crm to provide it leads / account information need maintenance? Or is it something that will just work? |
19:14.48 | [TK]D-Fender | rnp: HUH? |
19:15.24 | rnp | Fender: what I mean is if I have an asterisk system setup with custom scripting to work with my crm, will I have to worry about the asterisk server or will it just work without issue? |
19:15.40 | jksM | rnp, if you ask like that, you will have to worry |
19:16.05 | [TK]D-Fender | rnp: Its an app. Apps can lock-up, crash, fill your filesystem up with logs, have vulnerabilities that can be exploited, etc. |
19:16.06 | benngard | so it means if a wirite a small web app that let people make calls from ami they will not get recorded properly in cdr because of "nobody called" :( |
19:16.12 | [TK]D-Fender | rnp: Just like every other app out there. |
19:16.50 | [TK]D-Fender | [14:07]<[TK]D-Fender>benngard: Feel free to update your records to correct this AFTER The FACT <-------------------------- |
19:17.18 | rnp | Fender: most apps can run for months and years without having issues. I'm just worried that I'm getting into something too complex by having my own asterisk server |
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19:17.38 | [TK]D-Fender | rnp: as opposed to? |
19:18.07 | Katty | rnp: if you're that concerned, outsource it |
19:18.34 | Katty | rnp: let someone qualified manage it |
19:19.08 | rnp | katty: I wasn't concerned until someone trying to sell me the outsourced version struck fear into me |
19:19.51 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
19:19.51 | Katty | rnp: keyword being Qualified |
19:19.52 | *** join/#asterisk [netman] (~netman@147.Red-83-63-246.staticIP.rima-tde.net) |
19:20.43 | rnp | Katty: in your experience how much would it cost to have someone check in on an asterisk server once a week or so? |
19:20.53 | [TK]D-Fender | rnp: Well you haven't described what tasks * will be performing. What resources it will have, what devices it will talk to, etc |
19:21.30 | [TK]D-Fender | rnp: certaint hings require more monitoring than others. there are security considerations,e tc |
19:23.28 | rnp | Fender: it will work directly with my crm, getting leads from the crm to dial, then transmitting back successfull call data to the crm. for voip it will use a service like www.vitelity.net - as for the actual server i was going to use a slicehost.com virtual xen server |
19:23.34 | rnp | is that enough data? |
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19:25.03 | Katty | sighs. |
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19:26.03 | [TK]D-Fender | rnp: How many simultaneous calls? You will have keep up with updates on the software,etc. |
19:27.03 | wcselby | rnp - are you going to do the install / linkup with your CRM? or are you going to need someone else to do this for you? are you going to need anyone do custom dialplan programming, etc? |
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19:27.16 | rnp | Fender: maximum of 20 simultanious calls |
19:27.39 | wcselby | when you say someone should "check on it" once a week or so, what do you need them to check on? if it's working? if it needs updating? resource load, etc? |
19:27.46 | rnp | wcselby: I'm absolutely going to hire someone to set it up, do the link to the crm and custom programming. |
19:27.57 | Katty | rnp needs to sit down with a support company and have a bid worked out |
19:28.17 | rnp | wcselby: honestly, I don't even know if it's something that needs to be looked at once a week. My thought was someone should just check it out to make sure no problems were cropping up. |
19:28.28 | wcselby | wrnp - what I'm getting at is there a lot of factors involved, and no one can really give you an honest ball-park figure for what you're asking |
19:28.52 | wcselby | until you sit down with them and (as Katty said) work out a bid |
19:29.29 | rnp | Wcselby: can you give me a proper ballpark on this question, if the system is properly setup by a professional, should it need maintenance, or is it like a unix pc that can run for years without issue? |
19:29.46 | Katty | rnp: find a support company |
19:29.48 | Katty | rnp: request a bid |
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19:31.12 | Katty | wasn't there an irc support channel for this? |
19:31.17 | Katty | asterisk consultants or somethin |
19:32.21 | wcselby | as [TK]D-Fender said earlier, it will be like any other application - issues can crop up, security patches would need to be applied, etc. likely you'd work out a support contract guarenteeing x number of hours per month for a flat fee, with a negotiated rate for anything above that. i don't know what other people tend to charge for these kinds of things. |
19:32.30 | wcselby | Katty - I think there's an asterisk-biz mailing list |
19:33.01 | [TK]D-Fender | rnp: Have you calculate the bandwidth requireents, etc? |
19:34.29 | rnp | I figure 100k for 20 callers is enough |
19:35.40 | [TK]D-Fender | rnp: Unit of measure please. |
19:36.15 | rnp | Kilobyte |
19:36.21 | rnp | is that what you mean? |
19:36.29 | wcselby | lunch time |
19:36.29 | wcselby | :) |
19:37.10 | [TK]D-Fender | rnp: Try an say that in a more common UOM / multiple combination... |
19:39.05 | [TK]D-Fender | rnp: One hundred thousand kilo-bytes? Per? |
19:40.03 | [TK]D-Fender | wishes he had such a 100 MEGABYTE link (is that bi-directional?) |
19:40.43 | rnp | Fender: I'm not sure what you mean by VOM, but what I mean is each caller would use 5kilobytes of bandwith per second, and with 20 callers that would mean a throughput of 100kilobytes per second (about t1 speed if I'm not mistaken), that would be about 3gigs of bandwith for a 8 hour call shift |
19:41.47 | [TK]D-Fender | rnp: What codec? |
19:42.07 | [TK]D-Fender | rnp: is that 20 calls to the outside? Who is talking to who? |
19:43.12 | rnp | I don't know what codec, the 20 calls are to the outside, to businesses in the usa |
19:43.33 | [TK]D-Fender | rnp: Whats on the other side of the call? |
19:47.06 | rnp | a land line |
19:47.18 | rnp | well |
19:47.23 | rnp | we call businesses they have a land line |
19:47.26 | rnp | our end is just a computer with mic |
19:47.37 | rnp | transmitting to the asterisk server which uses voip |
19:49.07 | [TK]D-Fender | rnp: Local soft-phone on PC? Fine (Acutaqlly this is a SHIT expeience for the user, but i suspect you don't care) |
19:49.26 | [TK]D-Fender | rnp: Calculate about 85kbps PER CALL. |
19:51.01 | rnp | ok |
19:51.19 | rnp | 85 X 50 Calls an Hour X 8 Hours a Day X 20 Callers |
19:51.39 | rnp | 700megs a day |
19:51.40 | rnp | not bad |
19:52.17 | rnp | alright, i've got to run, thanks for the info |
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19:52.57 | Katty | scowls |
19:58.26 | gelpg | mort_gib: thanks the idea, know I can user the bri_cpe_pzmp signalling |
19:58.35 | p3nguin | How did he arrive at 700M per day? |
19:58.48 | nix8n82 | he thought each call last a second |
19:58.52 | raden_work | i was just thinking same thing |
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19:59.42 | raden_work | wow |
19:59.43 | [TK]D-Fender | Assumed call duration.... |
19:59.48 | [TK]D-Fender | Horrible math |
19:59.56 | [TK]D-Fender | SOSO |
20:00.19 | raden_work | lol |
20:00.20 | *** join/#asterisk oej (~olle@ns.webway.se) |
20:00.41 | [TK]D-Fender | ~soso |
20:00.42 | infobot | [~soso] Shoot-On-Sight Offense |
20:00.52 | p3nguin | I was coming up with like 300GB per day. |
20:01.29 | nix8n82 | I was wondering what that was |
20:02.30 | nix8n82 | I think I was the one that scared him last night. |
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20:10.37 | *** join/#asterisk niekvlessert (~niek@5ED25657.cable.ziggo.nl) |
20:12.28 | p3nguin | After recalculating, I'm thinking it's more like 3000GB per day. |
20:14.37 | raden_work | any opinion what might be more stable with asterisk 1.6 opensuse 11.2 or ubuntu server 8.04 LTS ? |
20:14.55 | raden_work | 85*50*360*8*20 |
20:15.01 | p3nguin | I would probably go with the LTS. |
20:15.11 | Corydon76-dig | raden_work: I don't think it makes much difference |
20:15.18 | *** join/#asterisk rtp4me (~447db8c1@gateway/web/freenode/x-pyeneglihnfkpech) |
20:15.29 | raden_work | Corydon76-dig, they are colo i want the least problems LOL |
20:15.39 | rtp4me | hey everyone |
20:16.08 | raden_work | 244gb a day ??? |
20:16.08 | Corydon76-dig | raden_work: as long as you aren't using VM in production... |
20:16.09 | raden_work | woa |
20:16.14 | niekvlessert | raden: don't think it'll make much of a difference |
20:16.20 | raden_work | ok thanks |
20:16.24 | raden_work | il lstick with what i know then |
20:16.29 | niekvlessert | when you compile yourselve, dunno about the default package quality |
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20:16.44 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
20:16.48 | p3nguin | raden_work: I'm coming up with 2391GB. |
20:16.54 | Corydon76-dig | If you use the MySQL ODBC connector, though, compile from source |
20:17.03 | *** join/#asterisk ktwilight (~keliew@254.63-240-81.adsl-dyn.isp.belgacom.be) |
20:17.07 | Corydon76-dig | You'll save yourself a TON of grief |
20:17.47 | niekvlessert | i like bri stuff... compiles very easy |
20:18.22 | rtp4me | I'm experiencing a possible bug in 1.4.26 x86_64 with app_queue. I have a queue set to ringall and autofill=no, but I am seeing autofill behavior (non FIFO), with latter callers ringing through before the oldest caller has been answered. Thoughts? |
20:19.26 | DotComStu | hello, is there a function that returns the last number to call an extension (similar to 1471 in uk) or should i add some astdb magic to my dialplan to save last caller id to an extension? |
20:19.51 | [TK]D-Fender | DotComStu: latter <- |
20:20.18 | rtp4me | DotComStu: I think the last number dialed is also stored in the asterisk database, you should be able to find a key for it |
20:20.20 | raden_work | i was wrong on my math |
20:20.29 | raden_work | 85 X 50 Calls an Hour X 8 Hours a Day X 20 Callers |
20:20.38 | raden_work | what does 50 calls a hour have todo with it |
20:20.48 | raden_work | 50 hours of calls a hour ? |
20:20.52 | rtp4me | Regarding my previous question, where should I head with it? Submit a bug? |
20:21.00 | DotComStu | [TK]D-Fender/rtp4me: ok thanks |
20:21.07 | p3nguin | dotcomstu: Set(DB(callerid/last)=${CALLERID(num)}) |
20:21.10 | rtp4me | DotComStu: no problem |
20:21.21 | [TK]D-Fender | raden_work: SHHH!!! |
20:21.34 | [TK]D-Fender | raden_work: Next you'll be expecting themt o be able to SPELL as well! |
20:21.42 | raden_work | LMAO |
20:21.43 | p3nguin | dotcomstu: SayDigits(${DB(callerid/last)}) |
20:22.17 | rtp4me | hello? |
20:22.32 | DotComStu | p3nguin: k, but i want to store last caller id per exten so i guess ${DB(callerid/last-${EXTEN})} |
20:23.00 | p3nguin | dotcomstu: I gave you the basic concept, so I know you'll be able to figure out the rest. |
20:23.08 | [TK]D-Fender | DotComStu: Better |
20:23.13 | DotComStu | thanks |
20:23.45 | p3nguin | dotcomstu: I just use it at home to check the last calling number and I even have the option to blacklist it. |
20:23.57 | rtp4me | When you guys encounter problems that seem to be bugs, do you submit a bug report? |
20:24.04 | rtp4me | Is anyone willing to respond to me on here? |
20:24.41 | [TK]D-Fender | rtp4me: We usually check it out with someone else first so we don't pst silly accidents as bugs |
20:25.08 | rtp4me | [TK]D-Fender: ok, totally agree. Who would you recommend I check it out with. It doesn't seem as though anyone here has any feedback. |
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20:25.27 | p3nguin | dotcomstu: http://pastebin.com/d1bc37e3e |
20:25.34 | niekvlessert | rtp4me: have you tried other strategies? just to make the strategy is the problem |
20:25.52 | niekvlessert | * sure |
20:26.47 | DotComStu | p3nguin: nice - didnt realise you could use *xx as an extension number :-) |
20:27.03 | rtp4me | niekvlessert: I haven't. That's going to be my next step. Either way, I need to get the ringall working though. It's like it's using autofill, but it's completely disabled. |
20:28.03 | gelpg | I try to dial out via a B410P |
20:28.27 | gelpg | exten => 1004,n,Dial(DAHDI/21/06204661111) |
20:28.36 | *** join/#asterisk raden_work (~jon@69-179-99-17.stat.centurytel.net) |
20:28.44 | gelpg | but my error answer in asterisk is the following |
20:28.46 | niekvlessert | rtp4me: weird stuff... |
20:28.49 | gelpg | http://pastebin.com/d18e72926 |
20:29.34 | gelpg | any hints, what did I setup wrong? |
20:29.41 | p3nguin | dotcomstu: Alternatively, I also have a blacklisting extension in a different context: http://pastebin.com/d7e1df6b8 |
20:30.00 | rtp4me | niekvlessert: Agreed. I've never seen it happen on any other systems. Of course, this one has enough volume for it to happen much more often. I dug through the logs and don't seen any strange errors or timeouts reached... it's acting like its on autofill for all I can tell. |
20:30.22 | rtp4me | niekvlessert: I suppose the course of action is to just submit a bug and change ring strategy in the meantime. |
20:30.47 | *** join/#asterisk bsdmail (~dig@67.228.177.47) |
20:30.56 | niekvlessert | gelpg: the word dahdi is not in your log |
20:31.02 | niekvlessert | is this the correct log? |
20:31.31 | niekvlessert | rtp4me: why don't you try to get the queue smaller? |
20:31.45 | rtp4me | niekvlessert: what do you mean>? |
20:32.15 | gelpg | niekvlessert: I think so, at least that's what asterisk shows when I try to call exten 1004 via a sip phone |
20:32.16 | DotComStu | p3nguin: thank you - got that working nicely, *++ |
20:32.28 | wcselby | ahh |
20:32.30 | wcselby | tex-mex |
20:32.33 | wcselby | chips and salsa |
20:32.36 | wcselby | yummy |
20:32.53 | p3nguin | dotcomstu: I guess the US's *69 is UK's 1471? |
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20:33.00 | niekvlessert | rtp4me: well just remove people from the queue to see if all the phones ring then. :) |
20:33.01 | DotComStu | <p3nguin> yes |
20:33.14 | niekvlessert | can be done from CLI |
20:33.24 | rtp4me | well, it's not a problem with some of the phones not ringing |
20:33.38 | Katty | peeks in |
20:33.45 | rtp4me | its a problem with newer callers ringing on the phones before the oldest one has been answered |
20:33.46 | *** join/#asterisk Amorsen (~Amorsen@94.127.50.7) |
20:34.03 | DotComStu | is it poss to provide a "broken" dial tone to indicate stuff like message waiting etc for fones without a msg waiting lamp |
20:34.20 | niekvlessert | gelpg: try dialing voicemail or something with your phone first to see if it works |
20:34.25 | dlynes_laptop | DotComStu, it's done automatically |
20:34.29 | rtp4me | DotComStu: I believe you want to look into a playtones function |
20:34.36 | dlynes_laptop | DotComStu, it's called a 'stutter tone' |
20:34.57 | DotComStu | ok |
20:35.06 | dlynes_laptop | DotComStu, all phones I'm aware of have that enabled by default (Aastra, Linksys, Aastra, Mediatrix) |
20:35.11 | niekvlessert | rtp4me: ah sorry. that's weird, i've seen that behaviour in the past (we have customers using queues a lot) but that was asterisk 1.2 |
20:35.16 | gelpg | niekvlessert: I can call another softphone |
20:35.21 | niekvlessert | and the load had to be high |
20:35.28 | dlynes_laptop | DotComStu, but if it's not enabled by default, it is enablable |
20:35.47 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
20:35.53 | DotComStu | <dlynes_laptop> will look into that |
20:35.54 | rtp4me | niekvlessert: no problem, thanks for the help. time for a bug submission! |
20:35.58 | niekvlessert | gelpg: context problem then maybe? u're not getting to the dahdi part at all |
20:35.59 | dlynes_laptop | DotComStu, oops...second aastra should have been sipura |
20:36.37 | sbrath | If I have a call come in on a PRI from another internal Switch I'm rewriting the CALLERID(all)="555-1212", but the CALLERID(num) was an actual number in our office, CDR is being layed down as the rewritten CID, how can I change the CID out to the PSTN, but still write the real originating # to CDR? |
20:37.40 | dlynes_laptop | sbrath, write the real originating # as the 'accountcode'? |
20:38.18 | dlynes_laptop | sbrath, Set(ACCOUNTCODE=555-1212) |
20:39.01 | dlynes_laptop | sbrath, accountcode will take alphanumeric, too, so you can give it an English name as well...doesn't have to be a phone number |
20:41.57 | sbrath | Thanks :) |
20:42.18 | bsdmail | can someone help-me with a little issue, basically i want to receive a call on my cellphone (with a call-out script) and then dial 9 + externalnumber, and put the dialed number on the conversation, and play a music for both when I press 1. whats wrong? http://pastebin.com/d3a7a678f |
20:43.13 | sbrath | Hmm, didn't work. |
20:43.32 | p3nguin | sbrath: After you set the accountcode, make sure you use CALLERID(num) to set the number before you Dial() out. |
20:43.48 | sbrath | Thanks p3nguin! |
20:43.58 | *** join/#asterisk Wildy (~simba@89.222.134.42) |
20:44.02 | Katty | i have a cabbage craving |
20:44.32 | Katty | cabbage, garlic, mushrooms, and crushed red peppers |
20:44.43 | Katty | and maybe garlic bread |
20:44.44 | p3nguin | spews all over the channel |
20:45.00 | Katty | weirdo. |
20:45.06 | sbrath | awesome! it works. |
20:45.14 | gelpg | niekvlessert: what do you mean by that it does not get to the dahdi part? |
20:45.18 | p3nguin | The only cabbage I will eat is purple cabbage in a salad. |
20:45.25 | niekvlessert | thinks katty is always hungry..... she should bring bread from home |
20:45.35 | Katty | i am always hungry |
20:45.37 | p3nguin | Mushrooms are a no-no. I don't care for fungus that much. |
20:45.44 | Katty | but if i ate all the time, i'd be huge probably |
20:46.17 | niekvlessert | gelpg: well, I don't see any dahdi stuff in the log, so i guess that line in the dialplan is never reached |
20:47.10 | niekvlessert | [Feb 4 21:27:19] WARNING[8154]: pbx.c:3675 pbx_extension_helper: No application '' for extension (phones, 1004, 1): not good gelpg |
20:47.43 | niekvlessert | dunno what that says, maybe someone in here does... :) |
20:47.49 | bsdmail | can someone help? |
20:48.17 | niekvlessert | bsdmail: what does work and what doesn't? |
20:48.17 | gelpg | niekvlessert: what do i misconfigure? what kind of output or config would help? |
20:48.22 | dlynes_laptop | bsdmail, state your problem...we're not mindreaders |
20:48.28 | bsdmail | dlynes_laptop |
20:48.29 | Katty | dlynes_laptop: speak for yourself |
20:48.29 | p3nguin | What does phones,1004,1 have on it for a command? |
20:48.32 | bsdmail | can someone help-me with a little issue, basically i want to receive a call on my cellphone (with a call-out script) and then dial 9 + externalnumber, and put the dialed number on the conversation, and play a music for both when I press 1. whats wrong? http://pastebin.com/d3a7a678f |
20:48.34 | Katty | dlynes_laptop: personally, i'm all sort of telepathic |
20:48.38 | [TK]D-Fender | niekvlessert: No Application as it says... |
20:49.06 | bsdmail | before i dial the 9 extension, i'm able to dial extension 1 and hear the music, but i want to play the music for the other person |
20:49.11 | niekvlessert | yeah i know but WHY doesn't it know.... |
20:49.45 | p3nguin | (1448.29) <p3nguin> What does phones,1004,1 have on it for a command? |
20:50.06 | p3nguin | Show us that, it'll likely show the problem. |
20:50.12 | bsdmail | niekvlessert did u understood? |
20:50.28 | gelpg | p3nguin: how can I do that? |
20:50.39 | p3nguin | gelpg: You can look at the dialplan. |
20:50.43 | niekvlessert | bsdmail, yeah, but u still not told me what does work and what doesn't |
20:50.54 | dlynes_laptop | bsdmail, you're talking about hitting '1' both before the 9xxx call _AND_ after it? |
20:50.56 | p3nguin | gelpg: dialplan show phones or visually look in extensions.conf |
20:50.58 | niekvlessert | is the music on hold the problem? |
20:51.16 | *** part/#asterisk ch1nawhyte (~dliu@aker.tobi-office.com) |
20:51.34 | bsdmail | yes, because i cant trigger the music on my side, and other side can't trigger too... |
20:51.44 | bsdmail | so the music doesn't starts |
20:51.45 | dlynes_laptop | bsdmail, also your pastebin does not illustrate what your problem is...it only illustrates that you don't have anywhere near enough dialplan to implement what it is you're talking about |
20:52.09 | gelpg | p3nguin: http://pastebin.com/d3c74ffac |
20:52.37 | bsdmail | so can u tell-me a site or tuto with the thing that i want to do? |
20:52.39 | niekvlessert | -2?? what is that |
20:52.40 | p3nguin | gelpg: extension 1004 is broken. |
20:53.08 | p3nguin | gelpg: Go look in extensions.conf and fix it. If you aren't sure what is wrong with it, paste it in the pastebin and I'll look at it. |
20:53.40 | niekvlessert | bsdmail: once again, where are you at now?? can you recieve the call on your mobile?? |
20:53.53 | bsdmail | yes |
20:54.05 | bsdmail | let me explain, step by step |
20:54.35 | gelpg | p3nguin: http://pastebin.com/d1ad78faa |
20:55.18 | p3nguin | ;exten => 1004,1,Verbose(1, Extension 1004) |
20:55.23 | niekvlessert | exten => 1004,n,Dial(DAHDI/21/06204661905) |
20:55.24 | p3nguin | exten => 1004,n,Dial(DAHDI/21/06204661905) |
20:55.32 | niekvlessert | this should be 1,Dial etc |
20:55.33 | p3nguin | YOu have no priority 1 for 1004. |
20:56.00 | niekvlessert | woohoo, my dcap is useful for something. :) |
20:56.05 | bsdmail | i have a sip softphone waiting for a call that is made with a asterisk call-out script, when i run the script, my sip phone rings and i answer. and i dial the extension: 9mycellphone , so my cellphone rings, and i answer, the next step is that i have to press 1 on my sip softphone to play a music to my cellphone hear and softphone itself too. but the problme is that the music doesn't plays. |
20:56.33 | p3nguin | gelpg: So you can either uncomment the first line or change the second line from n to 1. |
20:56.37 | niekvlessert | bsdmail: does asterisk register the 1 key? |
20:56.45 | bsdmail | yes |
20:56.46 | p3nguin | gelpg: then save/exit, then 'dialplan reload' |
20:56.52 | bsdmail | but, nothing happens |
20:57.15 | gelpg | p3nguin: I did, nothing has changed |
20:57.27 | p3nguin | gelpg: Then you did it wrong. |
20:57.33 | niekvlessert | gelpg: did you do reload? |
20:57.33 | bsdmail | i think that is some kind of linear problem |
20:57.49 | gelpg | yes, I did |
20:58.01 | wcselby | on asterisk 1.4 - if a call-limit is set to 1, and there's a call in progress on that sip channel, and another call comes in, why does asterisk throw an error message stating: ERROR[6011]: chan_sip.c:3358 update_call_counter: Call to peer '2625' rejected due to usage limit of 1 ? I mean, why is it considered an error, and not just the way it's supposed to be? |
20:58.01 | p3nguin | gelpg: Which did you do, change n to 1 or uncomment the first line? |
20:58.18 | niekvlessert | can you show us the show dialplan rule for 1004 again? |
20:58.30 | gelpg | p3nguin: uncomment the first line |
20:58.35 | *** join/#asterisk evil_gordita (~evilgordi@ip70-188-50-186.rn.hr.cox.net) |
20:58.35 | niekvlessert | bsdmail: no idea, I'm not that good yet |
20:58.43 | p3nguin | "dialplan show phones" again |
20:59.05 | *** join/#asterisk ruben23 (~AGENT@122.55.48.243) |
20:59.18 | gelpg | http://pastebin.com/d649cd5c1 |
20:59.37 | niekvlessert | and now paste the debuglog |
21:00.01 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
21:00.38 | gelpg | http://pastebin.com/d69efede2 |
21:01.44 | niekvlessert | rule 47 in your paste says it all |
21:01.51 | p3nguin | Unable to create channel of type 'DAHDI' |
21:02.03 | gelpg | yes |
21:02.24 | niekvlessert | does dahdi_tool show you the card is up and stuff? |
21:02.25 | gelpg | but dahdi show channels shows |
21:02.39 | niekvlessert | gelpg: tried channel1? |
21:02.47 | niekvlessert | or a group of channels? |
21:02.59 | gelpg | http://pastebin.com/d5f853cfd |
21:03.22 | gelpg | channel 1 is my first analog phone |
21:03.34 | niekvlessert | ok, i mean the first isdn 30 channel |
21:03.47 | niekvlessert | is your jumper in the right setting? :) |
21:04.11 | niekvlessert | takes a show first |
21:04.14 | niekvlessert | *shower |
21:04.24 | p3nguin | shower show? |
21:04.28 | bsdmail | niekvlessert and there is other problem, i can't hear on softphone what is said from my cellphone, i think is that asterisk isn't putting the calls togheter |
21:04.35 | gelpg | the card lights green |
21:04.36 | p3nguin | hopes you're a girl, otherwise will not be watching |
21:05.22 | bsdmail | there is a command to put after the answer to put the calls in a "conference" |
21:05.22 | gelpg | it is a simple NT |
21:05.23 | bsdmail | ? |
21:05.35 | gelpg | with 2 channels |
21:05.38 | p3nguin | MeetMe() |
21:06.13 | gelpg | the card is TE mode |
21:06.21 | DotComStu | how do you tell Dial() to use the dialplan instead of directly dialing the channel? eg Dial(SIP/123) is not reaching the macro for extension 123 even though its been included in the context |
21:06.43 | p3nguin | dotcomstu: local channels -- Dial(Local/123@context) |
21:06.46 | bmoraca_work | DotComStu: use the local channel |
21:07.00 | DotComStu | thanks - thats logical |
21:08.08 | *** part/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
21:08.40 | wcselby | hmmm |
21:08.52 | Wildy | hi there :) anyone tried to use res_snmp w/Zabbix? |
21:12.36 | bsdmail | app_meetme.c:774 build_conf: Unable to open pseudo device |
21:13.26 | ManxPower-work | bsdmail: you need zaptel/dahdi installed to use MeetMe |
21:14.59 | gelpg | p3nguin: here is my dahdi_cfg -vv output: http://pastebin.com/d659fdc78 |
21:15.44 | p3nguin | gelpg: Hopefully someone else can lend a hand here, as I do not use dahdi channels to know what they should look like. |
21:16.09 | gelpg | p3nguin: thanks anyway |
21:16.19 | ManxPower-work | we would need your dahdi config FILES as well |
21:20.15 | niekvlessert | bsdmail: ztdummy is enough |
21:20.18 | gelpg | my system.conf : http://pastebin.com/d2e21ffcb |
21:21.07 | niekvlessert | span=3,0,0,ccs,ami |
21:21.11 | niekvlessert | no timing? |
21:22.02 | gelpg | i read it from a sample config |
21:22.21 | gelpg | what shall I change? |
21:22.38 | ManxPower-work | gelpg: other than lack of echo canceler it looks good. |
21:23.08 | gelpg | my cha_dahdi.conf : http://pastebin.com/d1ab7d117 |
21:23.16 | benngard | how do u say in english? pest or kolera? |
21:23.52 | ManxPower-work | gelpg: what specific problem are you having? |
21:24.26 | gelpg | i try to dial oupt my B410P card |
21:24.47 | titter | Whats the best way to force users to change their voicemail password the first time they access their voicemail? |
21:24.56 | gelpg | but I cannot and I'm trying to find out what is the problem |
21:25.02 | ManxPower-work | titter: voicemail.conf.sample should have info on that. |
21:25.10 | titter | ManxPower-work: thanks |
21:25.11 | ManxPower-work | gelpg: pastebin the CLI output of a failed call |
21:25.36 | ManxPower-work | titter: you set their password to be the same as the voicemailbox and then enable some option in voicemail.conf |
21:26.02 | gelpg | http://pastebin.com/d22f3c64c |
21:26.08 | ManxPower-work | gelpg: also pastebin the output of "dahdi show channels" in the CLI |
21:26.38 | ManxPower-work | This indicates you have a DAHDI issue. [Feb  4 22:25:27] WARNING[8451]: app_dial.c:1745 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) |
21:26.44 | gelpg | http://pastebin.com/d1b30eb3e |
21:27.15 | gelpg | ManxPower-work: yes, that's my problem |
21:27.34 | gelpg | but I have no clue what do I wrong |
21:27.45 | niekvlessert | gelpg: span=3.1.0,ccs,ami |
21:27.54 | ManxPower-work | gelpg: I would contact Digium support. Give all the configs you gave us. |
21:27.55 | niekvlessert | span=3,1,0,ccs,ami |
21:28.12 | ManxPower-work | niekvlessert: commas, not . |
21:28.30 | niekvlessert | ManxPower-work: /me is kinda tired :) |
21:29.19 | gelpg | niekvlessert: I tried but nothing has changed |
21:29.30 | n3hxs | is no longer away : Gone for 2 hours 52 minutes 3 seconds |
21:29.46 | niekvlessert | hmm |
21:30.15 | niekvlessert | test |
21:30.20 | niekvlessert | ah ; |
21:30.23 | niekvlessert | :) |
21:30.34 | niekvlessert | i shouldn't start a line with /var/log/syslog |
21:30.42 | niekvlessert | check that gelpg |
21:30.52 | niekvlessert | or messages |
21:31.19 | dlynes_laptop | Has asterisk always used sequential channel names, and I just never noticed? |
21:31.38 | ecrane | Any ideas who this VoIP equipment provider is that tries to use 1.1.1.1? http://labs.ripe.net/content/pollution-18 |
21:32.03 | *** join/#asterisk superbeef (~lanej@74.84.194.4) |
21:32.45 | raden_work | running asterisk as a stand alone server with 300 extensions is there anything ishould do special for partitioning ? |
21:32.47 | superbeef | are there any CLI tricks for disabling forwarding on an extension? |
21:33.24 | nix8n82 | only if you want to do something special |
21:33.33 | superbeef | how special |
21:33.41 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
21:33.44 | ManxPower-work | superbeef: You must be using a GUI |
21:33.47 | nix8n82 | that is a good question |
21:34.14 | Warp4 | hi all, writing a short checking script in php for asterisk (at http://pastebin.mozilla.org/701058) and i am wanting it to pull back the status of a call. what do I need to add to this to get this to happen? |
21:34.51 | superbeef | ManxPower-work: yeah AMP or Freepbx manages my extensions..... I tried doing database show and looking for a flag on my ext but didnt see anything to set |
21:34.56 | gelpg | niekvlessert: http://pastebin.com/d395e23ea |
21:34.56 | niekvlessert | superbeef: use agi to get to a script that disables the forwarding by changing a field in de the db |
21:35.00 | ManxPower-work | superbeef: since Asterisk does not actually have a Call Forward feature. |
21:35.04 | ManxPower-work | ~freepbx |
21:35.05 | 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 |
21:35.18 | gelpg | niekvlessert: I mean I checked but nothing has happened |
21:35.39 | niekvlessert | gelpg: weird... can you give me shell access? |
21:36.05 | niekvlessert | gelpg: we can use screen so that you can see i won't pollute anything :) |
21:36.07 | p3nguin | n3hxs: I really hope you're going to stop doing that soon. |
21:36.12 | DotComStu | why doesnt BackGround(digits/12345) work? |
21:36.26 | p3nguin | n3hxs: You'll notice that none of the other 242 people here do that. |
21:37.06 | gelpg | niekvlessert: it's a littlebit difficult but I try to organize |
21:37.16 | niekvlessert | ssh portfowarding ftw |
21:37.31 | niekvlessert | do you guys know ssh -w any:any ? |
21:37.45 | nix8n82 | Warp4, I would recommend looking at phpAGI it has a manager class that makes things easier |
21:37.47 | niekvlessert | one can tunnel sip + rtp through an ssh tunnel |
21:37.49 | niekvlessert | with it |
21:37.53 | dlynes_laptop | ecrane, probably misconfigured...it's the example ip address usually given in Cisco examples |
21:38.00 | Warp4 | nix8n82, got a current URL for it? |
21:38.14 | DotComStu | is it possible to saydigits() in the style of Background() |
21:38.22 | raden_work | nix8n82, mainly just wondering how full mailboxes could get etc.... |
21:38.53 | n3hxs | p3nguin do what? |
21:39.27 | ManxPower-work | DotComStu: not that I'm aware of, but if you find out let me know |
21:39.40 | nix8n82 | google phpAGI |
21:39.51 | Warp4 | ~phpAGI |
21:39.52 | infobot | i heard phpagi is http://phpagi.sourceforge.net/ |
21:39.54 | dlynes_laptop | ecrane, it also seems to be an example ip address used by Huawei, too |
21:39.57 | DotComStu | ManxPower: http://lists.digium.com/pipermail/asterisk-users/2006-January/136692.html |
21:40.35 | niekvlessert | i will ask the question again i also asked yesterday |
21:40.58 | ManxPower-work | DotComStu: Ah. Clever. |
21:41.04 | p3nguin | n3hxs: I'm talking about the away/back announcements. |
21:41.27 | ManxPower-work | It won't really work for my needs, but it's still clever. |
21:41.32 | n3hxs | OH, sorry, I will fix that. |
21:41.42 | p3nguin | n3hxs: Thanks! |
21:41.42 | *** join/#asterisk Greek-Boy (~Monching@41.188.154.137) |
21:41.52 | *** join/#asterisk zafar_ (~IceChat7@91.144.34.220) |
21:41.56 | niekvlessert | is someone in here capable and willing to change the asterisk source so that sip headers can be sent by sip after a call has started? So i can change the display from a phone when doing direct call pickup for example... I'm willing to pay for it |
21:42.03 | *** join/#asterisk phix (~threat@123-243-44-131.tpgi.com.au) |
21:42.11 | ManxPower-work | DotComStu: you can also just merge the individual sound files if it's something that will be the same often |
21:42.36 | nix8n82 | raden_work, yeah I could see that as an issue, play it cautious man, maybe you could write a short cron job to alert you to when your mail is at critical size |
21:42.39 | *** join/#asterisk ttl- (~patrick@d5153A420.access.telenet.be) |
21:42.41 | *** join/#asterisk lupine_85 (~lupine_85@unaffiliated/lupine-85/x-7392152) |
21:42.57 | lupine_85 | scratches his head at chan_mobile |
21:43.01 | dlynes_laptop | niekvlessert, Are you using Polycom or Aastra phones? |
21:43.03 | raden_work | nix8n82, thanks ;) |
21:43.06 | DotComStu | ManxPower: its for random inbound callerids so i'mm sticking with the loop way of doing it |
21:44.05 | lupine_85 | so the module is failing to load, but not giving me any decent output as to why. that's with core set debug (big) and core set verbose (big), then module load chan_mobile. Anyone know how to coerce it into telling me what its problem is? |
21:44.11 | niekvlessert | dlynes_laptop: aastra |
21:44.29 | dlynes_laptop | niekvlessert, why not use xml? |
21:44.40 | niekvlessert | dlynes_laptop: slow and ugly |
21:45.10 | niekvlessert | i have to check through ami if a call has started and then decide what to push |
21:45.23 | niekvlessert | if the call ends i have to push it empty again |
21:45.59 | raden_work | brb |
21:46.54 | niekvlessert | dlynes_laptop: any opinions about it? it's an anoying problem.... my hopes are on sipsak right now |
21:47.30 | dlynes_laptop | niekvlessert, no idea...I don't know sip on a low enough level |
21:47.58 | niekvlessert | dlynes_laptop: same here... :( i'm willing to pay!! find me someone to fix it :) |
21:49.26 | Deeewayne | anyone here do any asterisk-java development? |
21:49.54 | ManxPower-work | Heh, Just got notification from one of our carriers that there will be an outage tonight to "clean the fiber" (I'm assuming fiber ends). |
21:50.17 | Deeewayne | offers ManxPower-work a bran muffin |
21:50.29 | dlynes_laptop | ManxPower-work, if it's inside of an airtight connection, why should they even need to clean it? |
21:51.09 | ManxPower-work | dlynes_laptop: You shouldn't, but that card is taking errors so they are looking at things like light levels, etc as well |
21:51.17 | dlynes_laptop | ah |
21:52.02 | zafar_ | how much r u people ready to pay for this, i think i have people who can do it |
21:52.29 | zafar_ | i need to talk to the guys though |
21:52.43 | *** join/#asterisk raden_work (~jon@69-179-99-17.stat.centurytel.net) |
21:52.57 | niekvlessert | zafar_: talking to me? |
21:53.06 | *** join/#asterisk came0 (~came0@rrcs-71-42-53-211.se.biz.rr.com) |
21:53.07 | niekvlessert | well we can discuss hourly rates |
21:53.14 | came0 | polycom 335 woot! :) |
21:53.20 | niekvlessert | and then make a guess on how much hours it will be |
21:53.31 | zafar_ | yes |
21:53.42 | zafar_ | i need to know the estimate so i can talk to them |
21:54.57 | *** join/#asterisk unspin (~unspin@96.49.129.159) |
21:55.07 | niekvlessert | estimate for what? |
21:55.22 | niekvlessert | i can write down what needs to be done? |
21:55.30 | niekvlessert | maybe they can decide how much hours that would be |
21:55.40 | zafar_ | like how much time can u give them and how much per hour |
21:58.46 | Katty | jeebus |
21:58.49 | Katty | i fell asleep |
21:58.50 | unspin | is it possible to apply a simple add on top of an existing integer value without first reading it in? |
21:59.02 | Katty | and i do believe that is the fastest i have ever darted awake >.< |
21:59.02 | unspin | for example: update partner set balance=add(5) where iid=1; |
21:59.02 | unspin | i'm running mysql 5.1 |
21:59.09 | Qwell | unspin: #mysql ? |
21:59.25 | unspin | aw crap |
21:59.25 | unspin | wrong window |
21:59.25 | unspin | :) |
21:59.36 | Katty | another cup of tea |
21:59.38 | Katty | move down move down |
22:00.47 | Katty | eppigy: what's fur dinner tonight? |
22:00.58 | Deeewayne | hugs Katty |
22:01.04 | Katty | hugs Deeewayne |
22:01.25 | Katty | Deeewayne: you must some major cuteness. |
22:01.31 | eppigy | Katty: STEAK AND BAKED POTATO |
22:01.35 | eppigy | 8D |
22:01.36 | Katty | eppigy: again?! |
22:01.38 | eppigy | YESH |
22:01.44 | Katty | eppigy: leftovers? |
22:01.46 | eppigy | I am on a diet |
22:01.49 | eppigy | negative |
22:01.52 | Katty | oh |
22:01.59 | eppigy | the easiest way for me to do diets |
22:02.02 | Katty | well you're not gonna gain much weight on steak and potato, dear |
22:02.04 | eppigy | is just eat the same thing every day |
22:02.07 | DotComStu | <unspin> update abc set balance=balance+1 where x=y |
22:02.09 | *** join/#asterisk mmj_nix (~mmj069@c-76-27-116-95.hsd1.ut.comcast.net) |
22:02.14 | eppigy | I eat six meals a day |
22:02.20 | Katty | okay well that might work |
22:02.32 | eppigy | yesh |
22:02.34 | Katty | put some cheese on that potato |
22:02.38 | eppigy | chicken and baked tatoes |
22:02.43 | eppigy | and avocados |
22:02.48 | Katty | now i want a baked potato |
22:02.57 | Katty | it sounds DELISH |
22:03.02 | eppigy | yesh |
22:03.25 | Katty | why do i always get hungry at 4? :< |
22:03.36 | eppigy | you are a masochist? |
22:03.57 | Katty | no |
22:04.10 | Katty | what's that got do with getting hungry at 4 |
22:04.24 | lupine_85 | oooh, another step along |
22:04.25 | eppigy | I am not sure :< |
22:04.34 | Katty | mmmmmmmmmisee. k |
22:04.48 | niekvlessert | zafar_ you here? |
22:04.50 | *** join/#asterisk chilicuil (~sistemas@unaffiliated/chilicuil) |
22:05.04 | Katty | eppigy: how many calories are you shooting for per day |
22:05.43 | mmj_nix | whats the easiest way to do incoming CID pattern matching w/o freePBX or the like? |
22:05.50 | Katty | infobot: freepbx |
22:05.50 | 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 |
22:06.27 | niekvlessert | zafar_: plz!! |
22:06.28 | eppigy | Katty: like three thousand something |
22:06.28 | mmj_nix | not using a GUI is what I meant |
22:06.48 | Katty | eppigy: that's a lot. |
22:06.51 | eppigy | Its been 2 1/2 months |
22:06.57 | Katty | eppigy: how much have you gained? |
22:07.00 | eppigy | and I have put on like 27 lbs of muscle |
22:07.05 | Katty | hot. |
22:07.05 | eppigy | 5 pounds fat |
22:07.08 | eppigy | yesh |
22:07.11 | Katty | i wanna see. |
22:07.28 | eppigy | pic incoming |
22:07.41 | eppigy | with silly expression |
22:07.45 | Katty | cheers. |
22:08.09 | Katty | omg. that's hilarious |
22:08.20 | zafar_ | sorry budy i was getting a cup of tea :) |
22:08.28 | eppigy | yesh |
22:09.45 | p3nguin | mmj_nix: You can use two ways. Use the old callerid matching method of exten => 123/432,1,Stuff() where 432 is the caller id number to match against, or you can use the newer method if GotoIf([${CALLERID(num)} = 432]?placetogo) |
22:10.12 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
22:12.24 | niekvlessert | aaah zafar_ is back with tea :) |
22:13.01 | Katty | i want some tea. |
22:13.47 | Katty | and some steak. |
22:13.49 | mmj_nix | p3nguin: tried to use DID/CCID, stuff in trunk include, seems not to find it though |
22:13.54 | Katty | no, i don't want steak. |
22:13.58 | Katty | i want.... |
22:14.01 | Katty | Soup |
22:14.15 | niekvlessert | [Feb 4 23:13:36] WARNING[9437]: channel.c:4003 ast_request: No channel type registered for 'DAHDI' |
22:14.15 | niekvlessert | [Feb 4 23:13:36] WARNING[9437]: app_dial.c:1745 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 66 - Channel not implemented) |
22:14.19 | niekvlessert | what does this mean? |
22:14.20 | benngard | had onion soup for dinner |
22:14.26 | p3nguin | mmj_nix: I don't even know what "in trunk include" means. We don't support FreePBX here. I'm just telling you how it is done in extensions.conf. |
22:14.28 | Katty | benngard: do you have a recipe? |
22:14.43 | benngard | Katty: no, but my wife has |
22:15.03 | Warp4 | hmm |
22:15.43 | Deeewayne | mmmm.... onion soup ..... |
22:15.43 | Warp4 | ok having an issue with the phpAGI thingy |
22:15.43 | benngard | Katty: i ask her to write it down and then i pass it to u |
22:15.43 | Katty | benngard: :>>>>>>>>>> |
22:15.43 | Katty | benngard: tank you berry much. |
22:15.43 | Katty | Deeewayne: well i was thinking more like colby corn chowder, personally |
22:15.48 | Warp4 | not quite sure what this means: PHP Parse error: syntax error, unexpected T_DOUBLE_ARROW in /root/scripts/test_rw.php on line 15 |
22:16.17 | Deeewayne | I don't even know what I want for dinner :-( |
22:16.36 | Qwell | Deeewayne: the souls of the infidels |
22:16.41 | Qwell | with a little white wine sauce |
22:16.58 | Katty | Deeewayne: when you get hungry, it'll come to you |
22:16.59 | niekvlessert | niekvlessert: [Feb 4 23:13:36] WARNING[9437]: app_dial.c:1745 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 66 - Channel not implemented) |
22:17.03 | Deeewayne | Qwell, people would be walking into automatic doors everywhere |
22:17.04 | mmj_nix | p3nguin: not using freepbx, just editing extensions.conf with DID/CCID, stuff from an included context |
22:17.16 | Warp4 | http://pastebin.mozilla.org/701059 |
22:17.22 | Deeewayne | Qwell, I'm guessing you got the simpsons reference |
22:17.26 | Qwell | indeed |
22:17.34 | Katty | Deeewayne: http://farm4.static.flickr.com/3501/4035529177_2707198336_o.jpg |
22:17.43 | Qwell | would have come up with a witty response, if I wasn't responding to an sms |
22:17.52 | Deeewayne | Katty, is that bacon ? |
22:17.57 | Katty | Deeewayne: why yes, it is |
22:18.02 | Deeewayne | yay! |
22:18.07 | Katty | Deeewayne: is there a law that says soup can't have bacon? |
22:18.38 | Deeewayne | not sure, but I heard that there is a law in Washington state forbidding the sale of lollipops |
22:18.46 | Katty | most unfortunate |
22:19.18 | Katty | Deeewayne: what's your opinion of great northern beans? |
22:19.34 | Deeewayne | java beans ? |
22:20.06 | Deeewayne | honestly, I fear most non-green beans unless they are mixed in with something else |
22:20.26 | Katty | digs up an old recipe |
22:21.02 | Katty | Deeewayne: http://42ndrecipestreet.blogspot.com/2009/07/ryans-country-beans.html |
22:21.10 | Katty | Deeewayne: ^- uses 2lbs of bacon |
22:21.19 | Deeewayne | woot! |
22:21.31 | p3nguin | mmj_nix: Here's an example of how to use it: http://pastebin.com/d3abd8d07 |
22:21.34 | Katty | Deeewayne: enjoy. |
22:21.41 | Qwell | eww |
22:21.41 | Deeewayne | rumor has it, it is Miller time at my office |
22:21.43 | p3nguin | mmj_nix: Pay close attention to lines 4, 5, and 6! |
22:22.21 | DotComStu | i would like to handle the calling of a sip extension thats not online, CHANUNAVAIL look like the DIALSTATUS to check - yet the response from dial is "exited non-zero" and it doesnt goto s-CHANUNAVAIL |
22:22.43 | p3nguin | sip extension, eh? |
22:22.48 | DotComStu | anyes |
22:23.04 | p3nguin | Extension discrimination! |
22:23.10 | DotComStu | lol |
22:23.21 | DotComStu | is that a known feature |
22:23.23 | Katty | Deeewayne: it was naptime in my office earlier |
22:23.24 | p3nguin | You know that devices are not extensions, yes? |
22:23.51 | DotComStu | yes but how do i tell if its registered |
22:23.59 | Katty | aka, I don't care. |
22:24.10 | DotComStu | the docs say "CHANUNAVAIL: Channel unavailable. On SIP, peer may not be registered. " |
22:24.20 | DotComStu | guess MAY is the operative word |
22:24.35 | p3nguin | If the device is not registered, it cannot receive a call. |
22:24.39 | Katty | p3nguin: we're supposed to get more weathers here :< |
22:24.46 | Katty | p3nguin: of the bad variety |
22:24.49 | p3nguin | I heard we should get snow tomorrow. |
22:26.11 | Katty | checks weather |
22:26.30 | Katty | well it was calling for ice |
22:26.56 | Katty | it's been changed to rain/snow now |
22:27.29 | Katty | ah, and snow all weekend. joy. |
22:27.49 | niekvlessert | <PROTECTED> |
22:27.50 | niekvlessert | argl!! |
22:28.13 | *** join/#asterisk rossand (~aross@dhcp-233-179.tb-classrooms.carleton.ca) |
22:31.40 | p3nguin | I was just told that we should expect rain to start around 6pm. |
22:32.37 | niekvlessert | http://www.voip-info.org/wiki/view/Bristuff |
22:32.47 | niekvlessert | sorry wrong window :) |
22:36.58 | *** join/#asterisk slima (slima@unaffiliated/slima) |
22:37.24 | *** join/#asterisk mnt_real (~sinan@bas1-montreal43-1177754708.dsl.bell.ca) |
22:37.36 | *** join/#asterisk mykhyggz (~col@evolone.org) |
22:41.27 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
22:43.37 | *** join/#asterisk rnp (~robertnpa@c-76-101-196-166.hsd1.fl.comcast.net) |
22:44.32 | lupine_85 | does anyone know offhand which bluetooth profile is required for sending SMS with chan_mobile? Looks like SPP? |
22:45.05 | Nugget | Get down with SPP (yeah you know me!) |
22:46.05 | Qwell | lupine_85: HFP supports SMS, I believe |
22:46.16 | mmj_nix | dialplan show 801.......@DID_trunk_1_default - shows a match from _NXXZXXXXXX, but still choosing 's' instead on the inbound call |
22:46.36 | Qwell | but yeah, HFP uses SPP,so.. |
22:47.08 | p3nguin | I decided _NXXZXXXXXX should really be _NXXNXXXXXX to meet NANP's specifications. |
22:47.34 | mmj_nix | trying |
22:47.45 | random_mike | greetings |
22:48.18 | p3nguin | mmj_nix: That change won't affect your result in this case, though. |
22:48.20 | dlynes_laptop | Is a TURN server a new alternative to STUN? |
22:48.39 | p3nguin | mmj_nix: Paste your debug of the failed call into pastebin.com |
22:49.03 | lupine_85 | Qwell: thanks. I'm trying to convince a windows mobile phone to provide the correct profile... |
22:49.11 | lupine_85 | as with any sort of windows thing, is Hard(tm) |
22:50.59 | Katty | eppigy: hmm. the hungries went away |
22:51.06 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
22:51.10 | p3nguin | Yeah, now I got 'em! |
22:51.11 | Katty | hi jaytee |
22:51.18 | Katty | p3nguin: oh? did you steal my hungries? |
22:51.29 | Katty | p3nguin: so kind of you ;> |
22:51.30 | p3nguin | I assumed you gave them to me. |
22:51.42 | Katty | could be, could be |
22:52.05 | jaytee | hi Katty |
22:54.08 | p3nguin | I have some fancy chips made of rice that I am considering devouring. |
22:54.58 | Katty | aren't you going home in 5 minutes? |
22:55.31 | p3nguin | riceworks sweet chili |
22:55.46 | *** join/#asterisk obnauticus (~obnauticu@about/windows/regular/obnauticus) |
22:55.49 | Katty | you're gonna ruin your dinner |
22:56.24 | p3nguin | Ah, good point. I'm making tacos for supper. |
22:56.34 | p3nguin | err, assembling tacos, really. |
22:56.57 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
22:56.59 | p3nguin | store-bought stuff, just have to cook the meat and put everything together. |
22:57.13 | Katty | ^_- |
22:57.19 | Katty | i don't get it. |
22:57.22 | Katty | that's all there is to tacos |
22:57.28 | Katty | meat, tortillas...and assorted chopped veggies |
22:57.32 | mmj_nix | p3nguin: http://pastebin.com/d5441307d |
22:57.43 | dlynes_laptop | Katty, yeah...no point to buying a 'mix', is there? |
22:57.49 | p3nguin | Well, I'm not making my own shells, not using my own meat seasoning, et cetera. |
22:58.06 | bmoraca_work | it's easy |
22:58.17 | p3nguin | I'll use Taco Bell brand seasoning, Old El Paso shells, Ortega sauce. |
22:58.34 | jaytee | the best of all worlds |
22:58.36 | p3nguin | mmj_nix: http://pastebin.com/d38bbdf00 |
22:58.59 | p3nguin | mmj_nix: Pay close attention to lines 4, 5, and 6. |
22:59.13 | bmoraca_work | windows XP is a peice of shit |
22:59.14 | Katty | p3nguin: ah, right |
22:59.15 | p3nguin | mmj_nix: CLOSE attention. |
22:59.33 | jaytee | can't remember if it's Old El Paso or Taco Bell but one of em makes a seasoning pack that is chipotle flavoring. it's pretty good |
22:59.41 | bmoraca_work | i can't run Dreamweaver, Photoshop and SQL Server Manager |
23:00.09 | p3nguin | I will shred my own cheese, though. Kraft block cheese is cheaper than buying it shredded alrady. |
23:00.16 | bmoraca_work | i have to shut one of them down before i can use another |
23:00.29 | p3nguin | jaytee: Taco Bell. That is the exact one I use. |
23:01.02 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
23:01.02 | bmoraca_work | p3nguin: i like the Lowrie's taco seasoning |
23:01.47 | p3nguin | jaytee: "chipotle flavor taco seasoning" is what's on the front. |
23:01.51 | Katty | hometime |
23:01.55 | Katty | later gaters |
23:01.56 | jaytee | yep |
23:01.58 | jaytee | later |
23:02.33 | p3nguin | I don't know why one brand can't have all the products I like, but I have to get all the different brands to be satisfied. |
23:03.27 | p3nguin | Oh, and I like store brand sour cream better than the national brand. :/ |
23:04.26 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
23:05.27 | p3nguin | mmj_nix: Are you understanding the significance of those lines that I say are important? |
23:05.27 | jaytee | I like the Stand and Stuff shells from Old El Paso |
23:05.38 | p3nguin | square bottoms? |
23:05.41 | jaytee | yep |
23:05.49 | p3nguin | Do they have those in super stuffer size? |
23:05.52 | jaytee | I make mine toasted |
23:06.26 | p3nguin | I prefer the super stuffer shells because I don't have to make and eat as many tacos. |
23:06.47 | jaytee | super stuffer size? don't think there's much difference between the Stand and Stuff from them and the Super Stuffer ones because they're wider than the regular shells so you can really fill em up |
23:06.54 | Qwell | pfft, shells. those aren't real tacos |
23:07.07 | p3nguin | Mexican-American! |
23:07.21 | jaytee | Qwell, I take it you're a soft taco man? |
23:07.31 | Qwell | jaytee: I'm a real taco man |
23:07.47 | p3nguin | Real tacos don't use tortilla? |
23:07.52 | jaytee | Qwell, please elaborate on what you think makes a real taco |
23:08.02 | Qwell | http://soundbites.typepad.com/photos/uncategorized/tacos1.jpg |
23:08.03 | mmj_nix | p3nguin: those are the CID matching lines (4 & 5) the NoOp line is the key? |
23:08.04 | Qwell | that |
23:08.40 | p3nguin | mmj_nix: The NoOp line, which also has the SAME PRIORITY as the CID matches allows non-matching calls to continue down the dialplan. |
23:08.47 | Qwell | jaytee: *completely* different |
23:08.53 | Qwell | ground beef. pfft! |
23:09.00 | *** join/#asterisk unspin (~unspin@96.49.129.159) |
23:09.05 | Qwell | and...cheese?! |
23:09.09 | jaytee | those look like soft flour tortillas |
23:09.10 | p3nguin | qwell: Yummy! Is that what you're making for me tonight? |
23:09.16 | Qwell | jaytee: they are corn |
23:09.25 | Qwell | well, okay, those might be flour |
23:09.40 | Qwell | (they should be corn though) |
23:09.42 | jaytee | might be masa flour |
23:09.50 | *** join/#asterisk hhkahya (~hulusikah@88.247.127.66) |
23:09.55 | jaytee | looks tasty |
23:10.32 | jaytee | there's a place near me that makes authentic tamales |
23:10.38 | p3nguin | I should do real homemade tacos some day soon. |
23:10.53 | jaytee | and it's real shredded beef or chicken |
23:11.17 | Qwell | I used to live in the ghetto. We had a tamale guy that would walk around the neighborhood with his cart. |
23:11.25 | p3nguin | I have two restaurants here: Tequila and El Rancherito |
23:11.26 | Qwell | $1/each. pretty good |
23:11.43 | p3nguin | They both make authentic food. |
23:12.45 | Qwell | I miss the corn man and his van. :( |
23:13.21 | Qwell | corn on the cob, buttered, put on a bunch of like parmessan cheese, and chili powder |
23:13.26 | Qwell | mmm |
23:13.39 | *** join/#asterisk garymc (~chatzilla@host86-158-86-203.range86-158.btcentralplus.com) |
23:13.40 | jaytee | had a guy like that in Oklahoma City. 4 for a buck back in the 70's. really good homemade tamales |
23:14.05 | Qwell | http://www.saveur.com/article/Recipes/Mexican-Corn-on-the-Cob-1000075465 |
23:14.13 | Qwell | I guess it was mayo and not butter |
23:14.23 | dlynes_laptop | damn....why don't they have food that cheap here? :( |
23:14.24 | p3nguin | I don't know how to shred the beef correctly, so mine might not be authentic when I'm finished. :/ |
23:14.32 | ruben23 | hi anyone used eyebeam softphones for attended transfer on asterisk. |
23:14.37 | garymc | Hi peeps, anyone know if or how I can get into a polycoms software in the office to fix the phone for the boss? The Server IP is opent to the web and i know the ip for the phone on the subnet is 192.168.0.34 |
23:14.39 | Deeewayne | Qwell, my in-laws in georgia get their potatoes from a guy who fills his car with potatoes and drives from building to building trying to off load them |
23:14.46 | garymc | im at home |
23:14.51 | Qwell | potatoes? |
23:14.54 | Deeewayne | yes |
23:15.00 | Qwell | like, just regular old potatoes? |
23:15.05 | ruben23 | i have requirements on it but dont know how to setup the attended transfer.. |
23:15.13 | Deeewayne | yup. a tiny old russian car full of potatoes |
23:15.19 | Qwell | are they like not very common there? |
23:15.35 | Deeewayne | I don't know |
23:15.39 | Qwell | how odd |
23:15.40 | garymc | is it possible to do via net or putty etc? |
23:15.57 | p3nguin | How do you normally admin the phones? |
23:16.07 | garymc | at the office |
23:16.15 | p3nguin | That's not HOW. |
23:16.24 | p3nguin | That is WHERE. |
23:16.25 | garymc | i log into them at the office with their ip |
23:16.28 | p3nguin | HOW |
23:16.32 | garymc | on the internet browser |
23:16.36 | garymc | ;) |
23:16.42 | p3nguin | okay, now we're getting somewhere. |
23:16.43 | *** join/#asterisk fofware (~chatzilla@190.7.25.160) |
23:16.46 | garymc | :) |
23:16.51 | Qwell | oh, man, and the breading stuff with lemon juice and chili.. I forget what that was called :( |
23:17.03 | *** join/#asterisk hachi (hachi@shego.kuiki.net) |
23:17.04 | Qwell | wow. I miss living in the ghetto. wtf? |
23:17.10 | garymc | is it possible to do ? like quickly? |
23:17.14 | dlynes_laptop | garymc, install squid to bind to localhost on the server, then do ssh -L 3128:localhost:3128 serveraddress, and then fire up firefox, and tell it use a proxy of 127.0.0.1:3128 |
23:17.16 | p3nguin | You can use ssh (use PuTTY) to create a tunnel (socks proxy) from your home computer to your server in the office. It will be similar to a VPN. |
23:17.24 | p3nguin | garymc: ^^ |
23:17.40 | garymc | do you know what command i need to use? |
23:17.48 | dlynes_laptop | garymc, i just gave it to you up above |
23:18.16 | garymc | fuk it hell have to wait :S |
23:18.21 | garymc | he will* |
23:18.23 | dlynes_laptop | garymc, you need to have squid installed, however...I suggested binding to localhost though, so you're not running it as an open proxy server |
23:18.24 | p3nguin | That's a lot of unnecessary work. |
23:18.26 | p3nguin | garymc: Just wait. |
23:18.38 | p3nguin | garymc: You don't need squid. Just wait while I get the putty command. |
23:18.43 | garymc | thanks |
23:18.51 | dlynes_laptop | p3nguin, oh...there's another way? |
23:19.04 | hachi | I used to use voip-info (a while ago) for all my asterisk dialplan command documentation needs, is there a new source of this information? because the http://www.voip-info.org/wiki/view/Asterisk+cmd+Record page isn't even showing the comma syntax that is apparently what asterisk wants |
23:19.25 | jaytee | hachi, what version of *? |
23:19.38 | dlynes_laptop | hachi, replace all pipe (|) symbols with commas (,) |
23:19.46 | hachi | yes, but the docs don't even show it |
23:19.53 | hachi | 1.6.2.0, from debian... laziness |
23:20.05 | hachi | I'm looking for the docs on the 'options' field, and things like that |
23:20.22 | jaytee | best docs are on the CLI, type core show application dial (or another application) |
23:20.23 | dlynes_laptop | hachi, Record(filename:format,silence,maxduration,option) |
23:20.24 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
23:20.37 | jaytee | so core show application record |
23:20.40 | hachi | yes, thank you, but the docs also say .format |
23:20.49 | p3nguin | garymc: PuTTY.EXE -C -D 443 -P 22 -l gary -ssh -N server.domainname.com |
23:20.53 | dlynes_laptop | hachi, eg Record(myrecording:WAV,10,120,${OPTIONS}) |
23:21.02 | p3nguin | garymc: That will create a socks proxy to your server. |
23:21.18 | garymc | ok then what?@ |
23:21.27 | p3nguin | garymc: Then you need to open your browser's proxy settings and set SOCKS proxy to localhost port 443. |
23:21.52 | p3nguin | garymc: Now you have a proxy that takes you into the server and LAN in the office. |
23:21.53 | garymc | command not found |
23:22.07 | dlynes_laptop | garymc, are you on linux, or windows? |
23:22.09 | p3nguin | You need to get putty, obviously. |
23:22.17 | garymc | im on a windows laptop |
23:22.25 | garymc | the server is linux |
23:22.43 | garymc | bash -c not found |
23:22.50 | garymc | bash -C not found |
23:22.58 | p3nguin | What are you doing? |
23:23.07 | p3nguin | On your Windows client.... |
23:23.13 | garymc | im in putty |
23:23.13 | dlynes_laptop | p3nguin, he's trying ot run your command on the linux server |
23:23.14 | p3nguin | open CMD.EXE |
23:23.19 | hachi | what can I use to record and keep the data if the user hangs up? |
23:23.19 | garymc | ahh ok |
23:23.43 | hachi | Record() actually says in the internal docs that all data will be lost, so this is what I experience, but I would like to record till the hangup |
23:23.47 | p3nguin | Use the path\to\putty.exe with the options I gave you. |
23:25.24 | p3nguin | I use a bat file on my USB flash drive so I don't ever have to type the commands. |
23:25.43 | p3nguin | I also use pageant so I don't have to enter my password. |
23:26.58 | p3nguin | "Program Files\PuTTY\PAGEANT.EXE" "Program Files\PuTTY\myserver.com.ppk" -c "Program Files\PuTTY\PuTTY.EXE" -C -D 443 -P 22 -l rob -ssh ns1.myserver.com |
23:27.24 | p3nguin | The more things I need to tunnel to the server, the more -D <port>s I put in. |
23:27.33 | garymc | ok that opened putty |
23:27.33 | p3nguin | Want to tunnel xmpp, add in -D 5223. |
23:27.55 | p3nguin | garymc: Good. Leave it alone. That is your client side of the tunnel. |
23:28.04 | p3nguin | garymc: Now adjust the browser's proxy settings. |
23:28.24 | p3nguin | garymc: SOCKS type, localhost, port 443 |
23:28.46 | p3nguin | garymc: Then you can put the phone's IP address into the browser just like if you were at work. |
23:28.52 | mmj_nix | <PROTECTED> |
23:29.01 | garymc | ok just need to find proxy settings |
23:29.08 | p3nguin | garymc: Which browser? |
23:29.15 | garymc | firefox |
23:29.35 | p3nguin | What's the menu next to Help? |
23:29.38 | p3nguin | Tools? |
23:30.02 | garymc | yep |
23:30.11 | p3nguin | It's the bottom selection on that menu. |
23:30.43 | p3nguin | I can't remember which item in the settings, but it should be like the second from the right side. |
23:30.50 | p3nguin | There is a tab for Networking. |
23:31.26 | garymc | ok in options |
23:31.28 | *** join/#asterisk Akiraa (~Akiraaaa@79.112.21.86) |
23:31.40 | *** join/#asterisk mattwj2002 (~Matt@wikisource/pdpc.active.mattwj2002) |
23:31.46 | mattwj2002 | well guys I did it |
23:31.57 | garymc | manual prox settings or auto? |
23:32.02 | p3nguin | manual |
23:32.07 | mattwj2002 | I order a Cisco 7960 |
23:32.14 | mattwj2002 | *ordered |
23:32.24 | p3nguin | garymc: Leave all the boxes empty except for the socks one. |
23:32.32 | niekvlessert | cu guys |
23:33.06 | garymc | BINGO LAAAA!!!! |
23:34.17 | p3nguin | openssh's sshd is a pretty good socks proxy. It's great for browsing securely through your server at home or work when you are in the opposite place. |
23:35.21 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
23:35.36 | garymc | cool |
23:36.05 | garymc | after all that though I cant do what i need to through that. Ill have to do it in the phone lol but hey what a tool for future usage :P |
23:36.17 | p3nguin | heh |
23:36.28 | p3nguin | But you did reach the phone's web interface, right? |
23:37.07 | mattwj2002 | anyone want to configure my asterisk server for my new Cisco 7960 phone? |
23:37.23 | black | why wouldyou buy a cisco 7960? |
23:37.24 | p3nguin | How much does the job pay? |
23:37.24 | black | ~phones |
23:37.25 | infobot | While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Linksys SPA-9XX, Snom, everything else, and finally Grandstream phones. Do not consider Cisco phones. Ever. |
23:37.26 | black | ~phone |
23:37.26 | infobot | phone is, like, warbling while I'm updating the flash from blob...it's amusing now, it's like it knows I'm erasing its brain. Mwa ha ha ha. |
23:37.34 | black | "Do not consider Cisco phones. Ever." |
23:37.42 | black | ^ |
23:37.47 | Qwell | Who changed that? |
23:37.51 | p3nguin | Why would someone write that? |
23:37.58 | mattwj2002 | because I like Cisco |
23:38.02 | p3nguin | qwell: That's what I was wondering. |
23:38.04 | black | Then use a cisco pbx |
23:38.05 | black | =p |
23:38.10 | mattwj2002 | lol |
23:38.12 | mattwj2002 | :P |
23:38.16 | jaytee | ~gs |
23:38.17 | infobot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
23:38.28 | jaytee | ~grandstream |
23:38.29 | infobot | from memory, grandstream is the Yugo of VoIP hardware. Run. Run away now.. Though therealcircut says that they're not that bad |
23:38.31 | p3nguin | That's fairly recent. |
23:38.47 | p3nguin | mattwj2002: So how much was the job paying? |
23:38.51 | jaytee | the Cisco one is new to me but the other two have been around for awhile |
23:39.00 | Kobaz | what's crazy about grandstream... is that the speakerphone is higher quality than the handset |
23:39.05 | mattwj2002 | why do you ask? |
23:39.07 | Kobaz | the handsets are all staticy and crackly |
23:39.16 | Kobaz | doesnt happen with the speakerphone |
23:39.29 | p3nguin | jaytee: Someone recently added that statement about Cisco. |
23:39.42 | Qwell | p3nguin: it's gone |
23:39.42 | garymc | Thanks P3nguin what a star :P |
23:39.47 | garymc | good night |
23:40.30 | mattwj2002 | p3nguin: what do you ask? |
23:40.45 | jaytee | given Cisco's attitude about licensing I kind of agree with that statement |
23:43.00 | dlynes_laptop | p3nguin, your solution..how do you get it working with firefox? |
23:43.25 | dlynes_laptop | p3nguin, firefox doesn't give me the option to tell it to use socks, specifically |
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23:45.40 | hachi | thanks folks, got this all sorted now |
23:45.48 | *** part/#asterisk hachi (hachi@shego.kuiki.net) |
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23:48.51 | mattwj2002 | hey guys |
23:48.54 | mattwj2002 | I have a question |
23:50.02 | dlynes_laptop | p3nguin, nvm...figured it out...thanks for the tip |
23:50.33 | mattwj2002 | never mind I got it |
23:50.47 | mattwj2002 | well supper time |
23:50.49 | mattwj2002 | bye all! |
23:50.55 | *** part/#asterisk mattwj2002 (~Matt@wikisource/pdpc.active.mattwj2002) |
23:58.00 | bmoraca_work | OM NOM NOM! |
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