| 00:07.34 | rbordeaux | hi all. just a question. I finished my asterisk installation and testing the moh. I do the usual example test but I cannot hear anything. The console shows the following -- Started music on hold, class 'default', on channel 'Zap/1-1' |
| 00:07.35 | rbordeaux | <PROTECTED> |
| 00:07.35 | rbordeaux | r any ideas? |
| 00:09.57 | dhanes | rbor: did you install mpg123? |
| 00:10.24 | Bullseye_Network | mpg321, dont need it anymore |
| 00:10.31 | rbordeaux | no, checking the web it says that it doesnt need it any mroe |
| 00:10.33 | *** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 00:10.34 | dhanes | mpg123, pretty sure you do |
| 00:10.41 | rbordeaux | http://www.voip-info.org/wiki/view/Asterisk+config+musiconhold.conf |
| 00:10.50 | Bullseye_Network | Definately dont need it |
| 00:11.03 | dhanes | moh wouldn't work for me with last night's svn |
| 00:11.08 | dhanes | w/o mpg123 |
| 00:11.12 | Bullseye_Network | how is you musiconhold.conf setup? |
| 00:11.47 | rbordeaux | [incoming] mode=quietmp3 directory=/var/lib/asterisk/mohmp3 |
| 00:11.55 | *** join/#asterisk yosyp (n=yosyp@pool-70-108-86-225.res.east.verizon.net) |
| 00:12.07 | rbordeaux | I also added this line.. default => mp3:/var/lib/asterisk/mohmp3 but not sure. |
| 00:12.14 | Bullseye_Network | here is mine.. http://pastebin.ca/149634 |
| 00:12.29 | rbordeaux | (got to get used to use pastebin) |
| 00:12.45 | Bullseye_Network | linein is for, well using the sound card to bring in audio. |
| 00:12.54 | Bullseye_Network | Default plays mp3's in that directory |
| 00:13.11 | Bullseye_Network | what asterisk version are you using? |
| 00:13.36 | rbordeaux | 1.2.11 |
| 00:13.40 | Bullseye_Network | wait |
| 00:13.46 | Bullseye_Network | that is for mpg123 that an old one |
| 00:13.55 | Bullseye_Network | I only use linein now holdon |
| 00:14.01 | Bullseye_Network | I will get the native one. sorry |
| 00:14.23 | Bullseye_Network | http://pastebin.ca/149636 |
| 00:14.29 | rbordeaux | thanks, checking |
| 00:14.40 | *** join/#asterisk hads (n=hads@mail.nice.net.nz) |
| 00:15.41 | Bullseye_Network | I really liked it when the linin option was made available. |
| 00:15.54 | Bullseye_Network | s/linin/linein/g |
| 00:16.18 | rbordeaux | didn't work.. |
| 00:16.50 | Bullseye_Network | what version are you using? |
| 00:17.19 | rbordeaux | of asterisk? |
| 00:17.23 | Bullseye_Network | yes |
| 00:17.36 | rbordeaux | I compiled asterisk-1.2-current.tar.gz |
| 00:17.52 | *** join/#asterisk caloi (n=caloi@nat-66-218-1-114.usadatanet.com) |
| 00:18.13 | rbordeaux | rbast:/etc/asterisk# asterisk -V |
| 00:18.13 | rbordeaux | Asterisk 1.2.11 |
| 00:18.13 | rbordeaux | rbast:/etc/asterisk# |
| 00:18.34 | caloi | Hey All - Quick questions.... Is http://www.voip-info.org/wiki/index.php?page=Asterisk%20RealTime the best resource to learn about running Asterisk realtime ?? |
| 00:18.38 | Bullseye_Network | did you install asterisk-addons? |
| 00:18.51 | rbordeaux | checked the musiconhold=default on zapata.conf |
| 00:18.57 | rbordeaux | eys |
| 00:18.57 | rbordeaux | yes |
| 00:19.20 | rbordeaux | yes... make and make install |
| 00:23.22 | caloi | anyone want to help with a compile problem of asterisk-addons? |
| 00:24.05 | rbordeaux | i will try with mpg123.. the default is not working. |
| 00:25.24 | Bullseye_Network | mpg123 will work. Just make sure you use the correct version |
| 00:27.12 | rbordeaux | ok.. is there any other log besides the terminal and the -v ? |
| 00:27.12 | Bullseye_Network | all logs are in /var/log/asterisk |
| 00:27.28 | Bullseye_Network | and you can setup what is logged in logger.conf |
| 00:28.10 | rbordeaux | thanks |
| 00:33.20 | *** join/#asterisk UForgotten (i=uforgott@laurel.dreamhost.com) |
| 00:34.46 | UForgotten | what's the deal with echo cancellation... I tried MG2 and the echo is much worse |
| 00:39.00 | Aurs | Aug 26 02:22:03 WARNING[21897] pbx.c: No application 'DB' for extension (macro-dinpris_dn, s, 1) |
| 00:39.04 | Aurs | any hints? |
| 00:39.16 | Aurs | just upgraded from 1.0.11 to 1.2.11 on a box here |
| 00:39.34 | Aurs | DB() isn't working |
| 00:39.40 | Aurs | DBput and DBdel is working |
| 00:39.50 | Aurs | Aug 26 02:38:39 WARNING[24599] app_db.c: This application has been deprecated, please use the ${DB(family/key)} function instead. |
| 00:39.55 | Aurs | aah |
| 00:40.18 | Aurs | ${DB it is then... not DB( |
| 00:40.49 | Aurs | nice when you can answer your own questions |
| 00:41.25 | *** join/#asterisk dieno (n=Dieno@124.29.194.150) |
| 00:41.48 | dieno | can n e tell me how to install libxml2 2.6.20 |
| 00:42.57 | Aurs | still fc4? tried yum? |
| 00:43.03 | dieno | :) |
| 00:43.07 | dieno | yup |
| 00:43.07 | Aurs | yum install libxml2 |
| 00:43.16 | dieno | ok |
| 00:43.25 | dieno | where can i get this file |
| 00:43.52 | Aurs | just type "yum install libxml2" in a root-shell |
| 00:43.59 | dieno | ok |
| 00:44.24 | *** join/#asterisk DaveCanoe (n=Dave@H5.C74.B0.tor.eicat.ca) |
| 00:44.24 | dieno | hmm |
| 00:44.28 | dieno | dats wrkn |
| 00:44.49 | Aurs | of course |
| 00:44.52 | Aurs | ;) |
| 00:45.01 | dieno | thnx |
| 00:45.06 | Aurs | don't know what version you will get, but |
| 00:45.50 | DaveCanoe | OK. IAX2 transfer question: I have a situation where an * box is just a remote with little dialplan smarts. So calls end up doing something like phone->asterisk1->asterisk2->asterisk1->phone --- and iax2 should transfer this, no? I'm having trouble convincing it to. |
| 00:50.15 | dieno | and wat abt kernel how can i get latest 1 |
| 00:56.07 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.18.235.Dial1.SanJose1.Level3.net) |
| 01:01.31 | *** part/#asterisk dieno (n=Dieno@124.29.194.150) |
| 01:09.36 | *** join/#asterisk knight_ (n=knight@209.85.11.98) |
| 01:09.55 | knight_ | Does anyone know why a Polycom IP 501 might sit at "Running.... App = sip.ld" ? |
| 01:10.09 | knight_ | I can't get into the menus, etc, and the phone doesnt respond to pings, except during provisioning |
| 01:11.31 | blitzrage | bad firmware update? |
| 01:12.10 | knight_ | seemed to update ok according to the log |
| 01:14.05 | mishehu | ugh. this crisco 7960 is starting to really annoy me. |
| 01:14.45 | knight_ | same with this polycrumb 501 |
| 01:15.15 | mishehu | I have better luck lately with the polly coms *sqwak* |
| 01:15.33 | mishehu | gotta becareful though, because that XML config can bite your finger off |
| 01:15.47 | knight_ | yeah |
| 01:15.58 | knight_ | too bad there's no gui tool to edit it |
| 01:16.12 | mishehu | I thought somebody scripted it though |
| 01:17.03 | mishehu | my crisco does not have port 5060 open, so whenever I try to call to the phone, it gets "circuit busy" and dumped straight into voicemail. |
| 01:18.14 | knight_ | yuck |
| 01:20.13 | knight_ | damn |
| 01:20.22 | knight_ | i even got rid of cfg files and downgraded the fimware |
| 01:20.25 | knight_ | firmware |
| 01:20.39 | knight_ | and it still says "Running... App = sip.ld" after provisioning |
| 01:21.19 | mishehu | knight_: what firmware did you try to use? |
| 01:21.31 | knight_ | 1.6.6 and 1.6.7 |
| 01:21.42 | mishehu | what bootrom do you have? |
| 01:22.18 | rbordeaux | Bulleyes_Network: mpg123 did the trick |
| 01:23.51 | knight_ | 3.2.1 |
| 01:23.59 | knight_ | i just upgraded from 3.1.2 |
| 01:24.20 | knight_ | this problem existed with 3.1.2 on it too (i've never gotten this phone to work yet) |
| 01:25.45 | knight_ | i'm wondering if there's local config on the phone that needs to be cleared |
| 01:25.52 | knight_ | i cant get into the menus to do it |
| 01:25.59 | mishehu | you might need to do that. |
| 01:26.01 | knight_ | i can clear the boot settings with 468* |
| 01:26.10 | mishehu | I think there's a keystroke that might do it |
| 01:26.12 | knight_ | but that just clears DHCP, FTP, etc settings |
| 01:26.21 | mishehu | nah, it clears other stuff too |
| 01:26.22 | knight_ | it doesnt clear the local config |
| 01:26.28 | knight_ | i've done it several times |
| 01:26.34 | knight_ | hmm |
| 01:26.48 | mishehu | I've gotten it to clear local config in the past with earlier firmwares. |
| 01:27.03 | knight_ | hmm |
| 01:27.12 | knight_ | how far back can i go with 3.2.1 boot rom? |
| 01:30.56 | UForgotten | mishehu are you having trouble with your 7960? It works if you get the right files in the right order |
| 01:31.28 | knight_ | mishehu, which bootrom and sip do you use on your 501's? |
| 01:31.44 | UForgotten | the trick is having a dhcp server to feed it all the config data :) |
| 01:33.06 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 01:33.57 | *** join/#asterisk BugKham (i=CKGLOB@221.128.113.28) |
| 01:41.13 | *** join/#asterisk Ayano (n=Ayano_@adsl-70-245-190-90.dsl.spfdmo.swbell.net) |
| 01:41.56 | *** part/#asterisk Ayano (n=Ayano_@adsl-70-245-190-90.dsl.spfdmo.swbell.net) |
| 01:46.19 | knight_ | the sid.ld application loads, and then it just says running |
| 01:47.33 | vlt | Hello. I'm using the ubuntu asterisk package. When I try to receive a fax I get "WARNING[2321]: pbx.c:1697 pbx_extension_helper: No application 'rxfax' for extension ..." I installed "libspandsp0" but it didn't help. How to enable rxfax and txfax? |
| 01:56.05 | knight_ | ( |
| 01:56.07 | knight_ | :( |
| 01:56.27 | hads | vlt: I don't think you'll find many people here using packaged asterisk but try `apt-cache search asterisk | grep fax` |
| 01:59.32 | knight_ | and polycom's site even says a reset won't change config overrides |
| 01:59.34 | knight_ | :( |
| 01:59.44 | *** join/#asterisk ahattar (n=aha@ool-43551487.dyn.optonline.net) |
| 01:59.44 | knight_ | so wtf do you do if the config overrides are breaking the app |
| 02:00.36 | knight_ | ah ha, i disabled DHCP and i have a Welcome! screen now |
| 02:00.37 | knight_ | wow |
| 02:01.43 | knight_ | yay my phone works |
| 02:01.47 | knight_ | weird, i guess i cant use dhcp |
| 02:02.39 | *** join/#asterisk OmGhY2u (n=jiggaman@CPE0002724fc55e-CM000f211fd29c.cpe.net.cable.rogers.com) |
| 02:02.46 | OmGhY2u | hy |
| 02:03.17 | OmGhY2u | anyone here? |
| 02:07.11 | lowlevel | yes. |
| 02:12.36 | OmGhY2u | what's the cheapest way to get a toll-free line setup that will end up putting calls to foreign mobile networks following a menu system? (i think using asterisk pbx) |
| 02:12.46 | OmGhY2u | i am fully aware i cant do it myself i just dont know where to start |
| 02:28.46 | *** join/#asterisk caloi (n=caloi@cpe-24-58-22-17.twcny.res.rr.com) |
| 02:29.50 | *** join/#asterisk Luhiwu (n=marsosa@200.63.89.210) |
| 02:31.36 | Luhiwu | hi all, i'm having polarity detection problems, anyone willing to help? it seems the polaritytv variable is not being filled |
| 02:32.30 | *** join/#asterisk ThaZZa (n=me@229.9.233.220.exetel.com.au) |
| 02:32.37 | ThaZZa | Hey All. Have a question. |
| 02:32.50 | caloi | ThaZZa - what's up? |
| 02:33.24 | ThaZZa | caloi: Do you know which file is the default busy and unavalible message for voicemail on asterisk. |
| 02:33.38 | caloi | sound file? |
| 02:33.55 | ThaZZa | We used to have different busy and unaval messages, they have now changed back to the default. |
| 02:33.58 | ThaZZa | caloi: yeah |
| 02:34.27 | ThaZZa | caloi: i have copies of the orignal gsm files, and want to copy them where they should go. |
| 02:34.32 | caloi | ThaZZa: if you watch your console you should see what Asterisk is playing |
| 02:36.20 | ThaZZa | caloi: yeah yet that is just playing the default.. vm-theperson, and then the digits for the mailbox number. |
| 02:36.34 | *** join/#asterisk vexorg (n=vexorg@CPE0003478eef7c-CM0016b531e87c.cpe.net.cable.rogers.com) |
| 02:36.39 | ThaZZa | caloi: i know that it can be set to just play 1 file if that file exists. |
| 02:36.51 | caloi | ahh |
| 02:37.10 | caloi | not sure on that one |
| 02:38.09 | knight_ | anyone know how to get these polycom 501's to register to a remote asterisk over NAT? |
| 02:41.11 | *** join/#asterisk aaugustf (n=aaugustf@copper-14.dynamic2.rpi.edu) |
| 02:41.27 | aaugustf | hi everyone |
| 02:42.37 | aaugustf | in working on my asterisk config, i can route a call to an extension using Dial(SIP/200), but how can I direct the call to an IVT called test_ivr? |
| 02:43.47 | caloi | aaugustf: That depends on how your IVR application works |
| 02:43.53 | caloi | aaugustf: what is the app ? |
| 02:43.53 | *** join/#asterisk annonimous (n=annonimo@dsl-201-129-251-31.prod-infinitum.com.mx) |
| 02:43.58 | annonimous | hiya! |
| 02:44.20 | aaugustf | caloi: i'm no too knowledgable about IVR's yet, so I used the freepbx GUI |
| 02:45.01 | *** join/#asterisk novafirst (n=kosta@wrt1.niclab.com) |
| 02:45.20 | caloi | im not firmilar with freepbx or it |
| 02:45.31 | caloi | it's routing, but checkout GoTo |
| 02:46.35 | aaugustf | well, i tried using GoTo(test_ivr) - but that didn't go too well |
| 02:47.33 | novafirst | hi, I need someone to help me. I keep seeing this error in the log file: pbx.c: Cannot find extension context 'sipusers'. I posted my config files and log here: http://pastebin.com/776171 |
| 02:48.11 | caloi | aaugustf: you need to state a context(IVR) extension and prority --- Goto([[context|]extension|]priority) |
| 02:49.05 | ThaZZa | caloi: I found it. |
| 02:49.17 | caloi | cool man, good lucj |
| 02:49.20 | caloi | *luck |
| 02:50.02 | ThaZZa | caloi: all fixed if you want to know, you need just under the INBOX folder, a file called busy and unavail |
| 02:50.18 | caloi | ahhh - nice! |
| 02:50.22 | novafirst | anyone... |
| 02:50.30 | *** join/#asterisk castellvector (n=Administ@adsl-70-240-226-83.dsl.hstntx.swbell.net) |
| 02:52.33 | annonimous | asterisk can link an "remote extension" to a local DID of the remote point? |
| 02:54.19 | *** join/#asterisk yosyp (n=yosyp@pool-70-108-89-137.res.east.verizon.net) |
| 02:54.34 | yosyp | Can I use a headset to do an echo test when locally setup? |
| 02:55.01 | *** join/#asterisk Druken (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 02:55.04 | Druken | hey everyone |
| 02:55.07 | Druken | anyone alive? |
| 02:55.10 | yosyp | nope |
| 02:55.22 | Druken | excelent, didn't want to bug :) |
| 02:55.37 | caloi | alive |
| 02:55.39 | caloi | :) |
| 02:55.47 | Druken | i'm having some unsual behaviour on a recent upgrade..... |
| 02:55.57 | UForgotten | this channel is full of bugs, it's a veritable windows millenium edition in here |
| 02:56.19 | annonimous | lol |
| 02:56.19 | yosyp | :S |
| 02:56.29 | yosyp | so yeah i'll use a headset to do an echo test :P |
| 02:56.36 | Druken | got a few clients who don't seem to want to register, and i got one, that seems to want to unregister like a million times |
| 02:56.38 | annonimous | asterisk can link an "remote extension" to a local DID of the remote point? |
| 02:57.02 | Druken | annonimous: yes |
| 02:57.12 | *** join/#asterisk benjk_ (n=benjamin@f8a01-0357.din.or.jp) |
| 02:57.16 | annonimous | Druken, where can i get some info? |
| 02:57.21 | novafirst | hi, I need someone to help me. I keep seeing this error in the log file: pbx.c: Cannot find extension context 'sipusers'. I posted my config files and log here: http://pastebin.com/776171 |
| 02:57.23 | Druken | www.voip-info.org |
| 02:57.24 | Druken | :) |
| 02:57.27 | UForgotten | heh |
| 02:57.37 | annonimous | Druken, lol |
| 02:57.47 | UForgotten | ? google |
| 02:58.01 | Druken | novafirst: READ that error message and then ask me again what is wrong |
| 02:58.14 | annonimous | yep but i want to know how can i test it with an asterisk and a siemens highpack 4000 |
| 02:58.33 | annonimous | (higgh pack or hi pro i dont know well siemens) |
| 02:58.38 | Druken | annonimous: ahh, you never said anything about a siemens highpak 4000 |
| 02:58.52 | novafirst | but sipusers extension context does exist in extension.conf |
| 02:59.02 | annonimous | Druken, hehe yeah i forgotten it =) (blushes) |
| 02:59.18 | Druken | novafirst: have you restarted asterisk since adding it in? |
| 02:59.34 | annonimous | the scenario is asterisk --------> siemens with e1/ and dids (needs authentification) |
| 02:59.36 | novafirst | Druken: yes couple of time |
| 03:00.20 | Druken | did they break something in 1.2.11 with the sip registry? |
| 03:00.30 | annonimous | Druken, jeje sorry my english maybe its not good but i can handle the language |
| 03:01.08 | *** join/#asterisk pbx1 (i=pbx1@netblock-66-245-195-153.dslextreme.com) |
| 03:01.10 | UForgotten | I've got cisco 7960's here which are notorious for not registering properly and they work OK as far as I know |
| 03:01.29 | UForgotten | the system isn't in production use yet tho, it's just me playing with it right now |
| 03:01.37 | UForgotten | in two weeks we'll have 6 lines running on it though |
| 03:02.30 | novafirst | Druken: maybe, I am not sure. I also see this error: db.c: Unable to find key 'myuser' in family 'SIP/Registry' |
| 03:03.08 | Druken | sounds like you may have the same problem i'm currently having... |
| 03:04.26 | Druken | UForgotten: i got a sip user that registered fine on 1.0.2 without problems... i just upgraded, and it takes a huge dump on my odbc and this one phone keep unregistering like every 3 seconds |
| 03:05.06 | novafirst | maybe is missing module, since I am using modules.conf to only load basic modules. |
| 03:05.51 | Druken | novafirst: if you do sip show registry you'll probably notice myuser isn't in there |
| 03:06.21 | novafirst | how to issue that command? |
| 03:07.42 | Druken | the phone is trying to unregister from asterisk, when asterisk doesn't know the registration yet |
| 03:07.42 | UForgotten | yeah druken for yours try sip show peers and see if its listed |
| 03:07.42 | Druken | UForgotten: of course it'll be listed, it's in the sip.conf |
| 03:07.42 | UForgotten | druken, maybe your db config is hoesed? you went straight from 1.0.2 to 1.2.11? |
| 03:07.43 | Druken | hehehe not db config, i'm not that stupid... |
| 03:07.43 | Druken | db only for dialplan use data |
| 03:08.00 | Druken | type it in and press enter? |
| 03:08.41 | UForgotten | heh. novafirst, this is in the asterisk command like interface (cli) that you get when you log into the asterisk console, not the linux shell |
| 03:09.04 | UForgotten | I'm still an asterisk noob myself, but I try to lurk and soak up tidbits here and there |
| 03:11.28 | Druken | apparently i've fallen back to being a newb... cause they have changed so much shit since i've been into it... |
| 03:11.50 | Druken | fuck at this rate, i may just go back to my old version that i know worked... hehehe |
| 03:12.08 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 03:12.08 | *** mode/#asterisk [+o mog] by ChanServ |
| 03:12.16 | novafirst | ok guys, basic question - how to log into asterisk console |
| 03:12.16 | Druken | MOG!!! you there man ? |
| 03:12.19 | UForgotten | or you could start clean and reconfigure your stuff lol |
| 03:12.25 | Druken | novafirst: asterisk -r |
| 03:12.35 | mog | > |
| 03:12.36 | mog | ? |
| 03:12.39 | Druken | UForgotten: ha ha ha, your hillarious |
| 03:12.54 | Druken | mog, did they fuck something up in the sip registry? |
| 03:13.03 | mog | ? |
| 03:13.09 | mog | Druken, you are? |
| 03:13.26 | Druken | uhmm... i is me... :) been a while since i've been around... |
| 03:13.35 | mog | i understand |
| 03:13.43 | mog | who would fuck up your sip registry? |
| 03:13.54 | Druken | for some unknown reason, i figured i'd upgrade... |
| 03:14.32 | Druken | i ment in the current rel. 1.2.11 |
| 03:15.34 | caloi | anyone firmilar with Asterisk Realtime w. MySQL ? |
| 03:21.24 | OmGhY2u | My occupation make reportings for Kazakhstan Television. Previous I gypsy catcher and icemaker. I also have work as computer maintenance - I the one who paint it and remove dead birds from its pipes. |
| 03:21.39 | *** join/#asterisk Druken (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 03:24.09 | Druken | rm / -rf |
| 03:25.01 | benjk_ | computer ... dead birds ... pipes ????? |
| 03:25.32 | UForgotten | sounds like fox news to me |
| 03:25.39 | file | ha |
| 03:26.18 | UForgotten | oh come on, that was worth more than a ha. :) |
| 03:26.34 | Druken | pfft.... |
| 03:26.50 | UForgotten | I'll take it, lol |
| 03:26.58 | benjk_ | I still don't understand how the dead birds get into the computers |
| 03:27.13 | file | benjk_: someone puts them in there obviously |
| 03:27.20 | benjk_ | heh |
| 03:27.21 | UForgotten | or they were alive when they got in |
| 03:27.31 | file | "Put birds into your computer" kit off eBay |
| 03:27.41 | UForgotten | and since there were no open windows, they couldn't get out |
| 03:27.46 | file | comes with a sticker, "Pigeon Inside!" |
| 03:27.53 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
| 03:27.55 | Druken | wtf is pbx_wilcalu ? |
| 03:28.06 | mog | old callback app |
| 03:28.07 | mog | its dead |
| 03:28.11 | file | a deprecated module that has been removed a long time ago |
| 03:28.12 | benjk_ | networking interface maybe |
| 03:28.36 | benjk_ | there is an RFC for TCP/IP over a pigeon transport |
| 03:28.37 | Druken | mog: anyidea where asterisk might be calling it from then? so i can remove it? |
| 03:28.48 | file | Druken: you didn't wipe your modules directory did you? |
| 03:28.50 | file | and you upgraded? |
| 03:28.53 | mog | pbx_wilcalu.so |
| 03:29.12 | file | ALSO |
| 03:29.14 | file | mog: hi! |
| 03:29.28 | mog | hey file |
| 03:29.38 | novafirst | for sip registry to function properly do I need to load any module? |
| 03:30.03 | file | chan_sip.so needs to be loaded if you want to use SIP... |
| 03:30.12 | wunderkin | o rly? |
| 03:30.15 | Druken | file, sounds about right... |
| 03:30.31 | file | Druken: did you build from source and ignore the warning about old modules during make install? :) |
| 03:30.46 | Druken | also sounds about right.... |
| 03:30.55 | file | then do: rm -rf /usr/lib/asterisk/modules |
| 03:30.57 | file | and type make install |
| 03:30.57 | Druken | i'm a true linux newb, and don't RTFM :) |
| 03:30.59 | file | again. |
| 03:31.04 | bobnormal | novafirst: show sip registry - if it shows a line at all its trying or succeeded. |
| 03:31.27 | novafirst | bobnormal: shows nothing |
| 03:31.46 | Druken | wholly shit.... it loaded.... |
| 03:31.54 | Druken | now lets see if sip acts up on me... |
| 03:32.01 | file | there's a reason make install warns you of these things |
| 03:32.06 | file | it's called version skew :P |
| 03:32.48 | Druken | can you explain why one of my clients is now trying to unregister it's self every second? |
| 03:33.01 | file | ...no |
| 03:33.39 | Druken | well, that's wut i got... unfortunatly |
| 03:35.15 | novafirst | where is the registry db and which permission should have? |
| 03:38.04 | novafirst | ? |
| 03:38.16 | Druken | file : http://pastebin.ca/149789 |
| 03:39.07 | file | sip_xmit of 0x81eb670 (len 550) to 69.197.96.235:5061 returned -1: Operation not permitted, mucho bad |
| 03:39.29 | novafirst | Druken: it looks like we have the same problem |
| 03:39.32 | Druken | what does that mean? aside from mucho bad :) |
| 03:39.36 | *** join/#asterisk surfdue (n=Surf@unaffiliated/surfdue) |
| 03:39.47 | surfdue | hi i kinda have an emergency is there any service i can use to call coasta rica? |
| 03:39.48 | surfdue | for free |
| 03:39.52 | surfdue | even a trial |
| 03:40.53 | file | Druken: are you running Asterisk as a user? is a second copy running? what's the sip.conf look like for bindaddr and externip/localnet? |
| 03:41.51 | Druken | asterisk is running as root, yes i know it's bad, but i do it anyways, how in hell do you get 2 asterisk's running? it's got a check for that.. and it's on a public ip.. no need for externip/localnet |
| 03:42.04 | *** join/#asterisk thx2000 (i=AgentFLY@adsl-66-51-192-221.dslextreme.com) |
| 03:42.22 | file | people have done it, had two running at once |
| 03:42.59 | Druken | hmm, well, nope.. only thing i have is the one |
| 03:43.04 | thx2000 | Is it possible to set up call forwarding on a PSTN line to route calls into a voip line? |
| 03:43.15 | surfdue | file, any help here man |
| 03:43.24 | thx2000 | If so, would i still be able to receive multiple connections on the voip line? |
| 03:43.34 | file | surfdue: no. |
| 03:43.36 | surfdue | file, its kinda important that i find a service, i knew a free one then i lost it |
| 03:43.45 | surfdue | free voip to phone |
| 03:44.43 | Druken | *** glibc detected *** corrupted double-linked list: 0x08135570 *** |
| 03:44.43 | Druken | Killed |
| 03:44.49 | Druken | that can't be good... ? |
| 03:44.56 | file | ...cool |
| 03:47.32 | file | because you just HAD to upgrade |
| 03:47.49 | Druken | yeah i know stupid move.... |
| 03:48.04 | Druken | i haven't upgraded in shit... a year and a half? |
| 03:48.09 | Druken | i was using 1.0.6 |
| 03:50.59 | file | did you at least back up your stuff? |
| 03:52.41 | Druken | i still have all the original sources... |
| 03:54.30 | anthonyu | does anyone use les.net for a DID? |
| 03:56.04 | thx2000 | Is it possible to get a pstn provider to forward calls to a voip line? |
| 03:56.29 | UForgotten | sure, probably only one call path though |
| 03:56.49 | UForgotten | standard call forwarding will take one call path to any number generaly |
| 03:56.56 | thx2000 | sheist |
| 03:56.56 | UForgotten | s/aly/ally/g |
| 03:57.14 | file | also depends on what you mean by a "pstn provider" |
| 03:57.20 | thx2000 | sbc/pacbell |
| 03:57.36 | file | doubt it |
| 03:57.44 | file | you could get a PRI, but that's not VoIP |
| 03:57.57 | file | or use a company that gets a PRI/DS3/whatever from them and delivers it over VoIP |
| 03:58.30 | UForgotten | maybe we should ask what you're trying to do? :) |
| 03:58.39 | thx2000 | hehe, maybe :P |
| 03:58.43 | file | taking over the world is also not an acceptable answer btw |
| 03:58.56 | UForgotten | that's reserved for pinky and the brain |
| 03:59.06 | thx2000 | basically trying to salvage my pstn line and when all is well forward all the calls to voip |
| 03:59.27 | thx2000 | yet still have the pstn line available so if the voip connection drops calls will still come in via pstn |
| 03:59.51 | UForgotten | ok, you only had one line to begin with? |
| 04:00.04 | Druken | get yourself a SPA-3000 and bring the damn PSTN in yourself :) |
| 04:00.10 | thx2000 | Well theres 4 lines now, but id like to drop 3 of em and move em to voip |
| 04:00.20 | UForgotten | you can just add call forwarding, forward the line to your voip number, and stop the forward if voip is down |
| 04:00.33 | thx2000 | but will i only be able to receive one call? |
| 04:00.53 | UForgotten | correct |
| 04:01.00 | UForgotten | if you keep one line, you get one call path |
| 04:01.05 | Druken | possibly 2... |
| 04:01.43 | thx2000 | sigh |
| 04:02.12 | Druken | i wish that was my problem... hehe |
| 04:02.19 | bkw__ | have I expressed how much asterisk sucks lately? |
| 04:02.23 | thx2000 | Can either of u recommend a rock solid voip provider? |
| 04:02.24 | Druken | file, any idea wtf my machine is bitching about? |
| 04:02.45 | file | not off the top of my head |
| 04:02.53 | Druken | bkw__: like a $0.30 hooker? |
| 04:03.11 | UForgotten | that wouldn't suck very much |
| 04:03.18 | mog | exactly |
| 04:03.24 | mog | just like asterisk |
| 04:03.32 | OmGhY2u | wtf mog |
| 04:03.36 | bkw__ | <PROTECTED> |
| 04:03.40 | bkw__ | suckage is good |
| 04:03.42 | bkw__ | mmkay |
| 04:03.44 | OmGhY2u | mog, gnaa? |
| 04:03.45 | thx2000 | no recommendations eh? |
| 04:03.49 | bkw__ | plus i'm drunk! |
| 04:03.49 | *** join/#asterisk AJaymn (n=boiwonde@156-77.dsl.scc.net) |
| 04:03.58 | Druken | news? |
| 04:04.07 | bkw__ | drunken i'm drunk.. is that news? |
| 04:04.25 | mog | you guy should trade nicks |
| 04:04.51 | Druken | am i missing something? i have upgraded, zaptel, libpri, asterisk and asterisk-addons |
| 04:05.01 | UForgotten | thx2000, I think the recommendation is buyer beware, lol |
| 04:05.28 | UForgotten | there are quite a few out there, each of them with pro's and con's. I'm going to re-research them again this weekend cos we have to pick one for my office next week |
| 04:06.01 | *** join/#asterisk vexorg (n=vexorg@CPE0003478eef7c-CM0016b531e87c.cpe.net.cable.rogers.com) |
| 04:08.31 | thx2000 | UForgotten, werd...seems like u can find conflicting info on just about any of em |
| 04:08.53 | UForgotten | exactly |
| 04:08.57 | Druken | <PROTECTED> |
| 04:08.57 | Druken | <PROTECTED> |
| 04:08.57 | Druken | <PROTECTED> |
| 04:09.00 | Druken | WTF? |
| 04:09.10 | thx2000 | been using teliax for a while and the uptime seems good, but getting ahold of support tends to be a bitch |
| 04:09.16 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 04:09.16 | *** mode/#asterisk [+o denon] by ChanServ |
| 04:09.37 | UForgotten | I liked teliax on the phone, guys seemed real knowledable |
| 04:10.10 | thx2000 | So...what kind of redundancy is available for someone trying to move 4 lines to voip and still be able to receive incoming calls should the sh** hit the fan? |
| 04:10.38 | UForgotten | well like someone said, you can keep the copper lines and put them into a tdm card |
| 04:11.13 | thx2000 | Just a more expensive solution...gotta buy the card, and keep the more expensive lines |
| 04:11.20 | Druken | thx2000: what are you looking todo exactly? cause your trying to have the best of both worlds, and well, you ain't gonna get it |
| 04:11.31 | thx2000 | lol, thats pretty much what its soundin like |
| 04:12.01 | Druken | move your DID (phone number) over to a voip carrier, and just go voip |
| 04:12.16 | Druken | or, buy the damn card, and do your own voip |
| 04:12.36 | UForgotten | I think he is looking for redundancy tho |
| 04:12.37 | thx2000 | I wanna go all voip im just worried about uptime |
| 04:13.01 | file | well if you go for VoIP, you're dependent on your provider plus transport (ie: internet in most cases) |
| 04:13.04 | UForgotten | we're running 3 copper lines into ours, and will run voip for the rest |
| 04:13.06 | Druken | as i'm sure the providers are worried about too... |
| 04:13.07 | Druken | hehe |
| 04:13.14 | file | so have to factor that in... |
| 04:13.16 | novafirst | I think I am getting there ... sip users are registering properly in DB |
| 04:13.24 | file | and on a bad day, your connection to your provider might go kaboom |
| 04:14.05 | thx2000 | Exactly...thats why its nice havin a pstn line in the mix...im just trying to figure out if theres a seemless way to integrate em |
| 04:14.27 | UForgotten | it makes sense to bring them in with a tdm card |
| 04:14.52 | novafirst | I am able to connect with xlite. now how to setup access to my voicemail? |
| 04:15.45 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 04:15.46 | Druken | file: if i wanted to erase asterisk (installed version) from my system, and do a complette reinstall, what directories would i remove? |
| 04:16.08 | thx2000 | So in a nutshell...if u've gotta have rock solid dependency stick w/ the lines and the tdm card. If u're trying to save a buck and can live w/ occasional downtime go w/ a voip trunk |
| 04:16.24 | Qwell | thx2000: basically, yes |
| 04:16.25 | UForgotten | we're doing both |
| 04:16.30 | file | or do a mix of both |
| 04:16.55 | UForgotten | we'll have 3 copper lines for incoming calls (cos we know they work) and voip for outgoing calls |
| 04:18.14 | thx2000 | are the tdm cards pretty good w/ echo cancelling? |
| 04:18.29 | file | the TDM400 has no hardware echo cancellation capability, the TDM2400 does |
| 04:18.39 | Qwell | but, you can do it in software, fairly reliably |
| 04:18.45 | UForgotten | really? our tdm400 sounds great |
| 04:18.49 | file | it all "depends" |
| 04:19.02 | file | some people it's horrible, some people it's fine |
| 04:19.11 | thx2000 | I've just got a spa3000 and i've gotten the echo down a bit but its still pretty bad |
| 04:19.26 | Druken | tdm can even be the card, or ports... |
| 04:19.30 | Druken | it's a shootmatch |
| 04:19.31 | Druken | hehe |
| 04:19.33 | file | this is no good, I'm not killing anybody! |
| 04:19.39 | novafirst | Druken: there is no uninstall script, which btw I think is stupid, but yes you can remove all files and dir manualy |
| 04:20.00 | Qwell | novafirst: there is, in trunk at least |
| 04:20.01 | thx2000 | but i think the zaptel features within asterisk have better echo cancelling options, correct? |
| 04:20.34 | file | the software one can't compensate for all echo, thus why hardware is available |
| 04:20.36 | novafirst | Qwell: no there is no make uninstall! |
| 04:20.41 | Qwell | novafirst: there is, in trunk at least |
| 04:20.41 | UForgotten | there is software echo cancelling, several breeds. I even got mine to somewhat process a data call over alaw |
| 04:21.13 | UForgotten | which doesn't make sense to me, there's no echo on pstn, full duplex should be clean, but I guess it's because of the delay |
| 04:21.22 | file | UForgotten: in a perfect world |
| 04:21.28 | thx2000 | but i mean, the echo canceling available to a tdm400 is gunna be much better than what i can do w/ a spa3000 |
| 04:23.05 | novafirst | Druken: instead go to your source dir of asterisk-1.2.10 and open Makefile. In there you can follow install script and reverse |
| 04:31.36 | *** join/#asterisk tabmeister1 (n=troy@CPE00907f17e478-CM014300013422.cpe.net.cable.rogers.com) |
| 04:33.04 | *** join/#asterisk tasat (n=tasat@c-67-180-181-221.hsd1.ca.comcast.net) |
| 04:34.02 | tasat | any nufone people here? |
| 04:34.24 | Qwell | tasat: #nufone if you have a support question for them |
| 04:36.17 | tasat | Qwell: ok... appears empty -- in general, trying to senddtmf through a proxy/gateway -- the DTMF is going into the gateway as RFC2833 but I don't see anything coming out of the proxy -- this normal? |
| 04:41.07 | tabmeister1 | any ideas people? -- codec_lpc10: using generic PLC |
| 04:41.07 | tabmeister1 | Ouch ... error while writing audio data: : Broken pipe |
| 04:41.07 | tabmeister1 | Segmentation fault |
| 04:44.57 | fafnir | asterisk takes sooo long to compile from the fbsd ports |
| 04:45.45 | tabmeister1 | haha |
| 04:46.07 | fafnir | 500mhz 512 megs ram |
| 04:46.11 | fafnir | been almost 12 hours |
| 04:46.26 | fafnir | all those crazy dependencies |
| 04:47.33 | *** join/#asterisk Givemelove2k (n=bob@adsl-75-16-45-183.dsl.irvnca.sbcglobal.net) |
| 04:56.57 | ThaZZa | Anyone good at troubleshooting zaptel compile problems? |
| 04:58.06 | tabmeister1 | yay i fixed my own asterisk error!!!!! |
| 04:58.26 | *** join/#asterisk Ayano (n=erik_lee@adsl-70-245-190-90.dsl.spfdmo.swbell.net) |
| 05:00.02 | shodan | does * support h.264 video phones ? (the videophones under 300$ don't seem to support anything else) |
| 05:01.08 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 05:11.13 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 05:11.13 | *** mode/#asterisk [+o denon] by ChanServ |
| 05:17.19 | *** join/#asterisk dieno (n=Dieno@124.29.194.150) |
| 05:17.32 | dieno | wmissing-prototype |
| 05:17.43 | dieno | i recieve this mesage when i compile asterisk |
| 05:30.23 | anthonyu | that's a compiler flag, not a message |
| 05:30.34 | *** join/#asterisk evilmnky (i=evilmnky@69-163-156-13.atlsfl.adelphia.net) |
| 05:37.42 | dieno | how can i solve this prb |
| 05:38.05 | anthonyu | what compiler are you using? |
| 05:38.22 | anthonyu | (try cc -v) |
| 05:38.30 | dieno | ok |
| 05:41.23 | dieno | its show s some thing GNU make 3.80 and just some agreement |
| 05:41.26 | dieno | nothin else |
| 05:41.36 | dieno | i tried make cc -v |
| 05:41.42 | dieno | i tried 'make cc -v' |
| 05:42.41 | *** join/#asterisk SantaRosaMark (n=mark@c-24-4-207-104.hsd1.ca.comcast.net) |
| 05:42.43 | anthonyu | no make |
| 05:42.49 | anthonyu | just cc -v |
| 05:42.54 | dieno | ok :) |
| 05:44.00 | dieno | target i386 -redhat-linux |
| 05:44.24 | anthonyu | ok, must be gcc |
| 05:44.28 | dieno | yup |
| 05:44.35 | dieno | wat should i do |
| 05:44.50 | dieno | do i hav to upgrade gcc or somkind of programming tool |
| 05:44.55 | anthonyu | now, run make by itself, then paste the entire error into http://pastebin.com/ and i will look at it |
| 05:45.31 | dieno | ok |
| 05:45.47 | dieno | 'make' or 'make install' |
| 05:46.21 | anthonyu | did you already do the whole 'make' without errors? |
| 05:46.28 | dieno | nop |
| 05:46.40 | anthonyu | just do the 'make' then |
| 05:46.48 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 05:46.49 | dieno | ok |
| 05:46.51 | dieno | its done |
| 05:46.58 | anthonyu | did you get an error? |
| 05:47.07 | *** join/#asterisk novafirst (n=kosta@wrt1.niclab.com) |
| 05:47.18 | novafirst | buya |
| 05:47.19 | dieno | yea alots of eror like wstrict |
| 05:47.30 | dieno | wmissing |
| 05:47.33 | novafirst | Unable to open vm-login (format gsm) ??? |
| 05:47.36 | anthonyu | those are not errors, they are compiler flags |
| 05:47.54 | dieno | but i m having prb runing asterisk |
| 05:48.04 | novafirst | where do you define path to sound folder ?? |
| 05:48.15 | anthonyu | novafirst, asterisk.conf |
| 05:48.30 | dieno | chan_sip.o |
| 05:48.33 | dieno | is not loaded |
| 05:48.42 | dieno | n tooo many modules like not loading |
| 05:48.48 | anthonyu | dieno, try the make install |
| 05:49.10 | novafirst | anthonyu: how ? |
| 05:49.43 | anthonyu | novafirst, vi /etc/asterisk/asterisk.conf |
| 05:49.54 | SantaRosaMark | i know i am on asterisk irc - looking for SER or OpenSER step-by-step manual...google is not helping |
| 05:50.28 | dieno | plz anthonyu |
| 05:50.28 | novafirst | I know that, what I mean is how do I define directive ?? |
| 05:50.34 | anthonyu | oh |
| 05:50.34 | dieno | y my modules is not wrkn |
| 05:50.50 | anthonyu | dieno, do the 'make install' |
| 05:50.55 | dieno | ok |
| 05:50.59 | BugKham | which command in Manager API is used to stop the user's Playback or Background action? anyone knows? |
| 05:51.20 | novafirst | default sound dir /var/lib/asterisk/sounds |
| 05:51.48 | anthonyu | novafirst, i am wrong, there appears to be no asterisk.conf directive for that |
| 05:52.21 | novafirst | but I think sounds should be copied to desired dir ie. /var/spool/asterisk |
| 05:53.18 | anthonyu | novafirst, you can put the full path to a sound in playback() and background(), but it appears to be otherwise unconfigurable |
| 05:55.05 | novafirst | ok, what is the default path that asterisk use to find sound file? |
| 05:55.49 | anthonyu | /var/lib/asterisk/sounds |
| 05:56.45 | novafirst | because asterisk keeps complaining that can not find vm-login. Ok all files are in there but I am still getting error |
| 05:57.16 | dieno | hmm anthonyu |
| 05:57.17 | anthonyu | what is the error? |
| 05:57.20 | dieno | install is done |
| 05:57.24 | dieno | do i hav to restart ma PC |
| 05:57.30 | anthonyu | dieno, nice |
| 05:57.40 | anthonyu | dieno, no |
| 05:57.47 | dieno | then :) |
| 05:58.01 | anthonyu | dieno, as root, run asterisk -vvvvv |
| 05:58.07 | dieno | ok |
| 05:58.10 | anthonyu | dieno, as root, run 'asterisk -vvvvv' |
| 05:58.35 | dieno | Loader.c:554 load_module: |
| 05:59.13 | BugKham | is the a way to stop Playback on a user's channel using Manager API? |
| 05:59.17 | dieno | how do i test my ASTERISK Server |
| 05:59.19 | anthonyu | dieno, is this the first time you are installing asterisk? |
| 05:59.23 | dieno | yup |
| 05:59.25 | dieno | :D |
| 05:59.35 | anthonyu | dieno, run 'make samples' |
| 05:59.38 | BugKham | sending DTMF? |
| 05:59.39 | dieno | ok |
| 05:59.51 | dieno | hmm dats wrkn |
| 06:00.06 | dieno | ok done |
| 06:00.34 | anthonyu | dieno, as root, run 'asterisk -vvvvv' |
| 06:00.39 | dieno | ok |
| 06:00.49 | dieno | done |
| 06:01.13 | anthonyu | dieno, ok, now you should have the demo PBX working |
| 06:01.20 | dieno | yup |
| 06:01.28 | dieno | i hav IAX2 accoutn |
| 06:01.51 | anthonyu | i don't know anything about that, when you figure it out, you can teach me |
| 06:01.57 | dieno | haha |
| 06:01.58 | dieno | lol |
| 06:02.09 | dieno | but how do i chek my server |
| 06:02.41 | anthonyu | do an 'asterisk -r' and run through the demo |
| 06:02.45 | *** join/#asterisk copantl (n=galel@190.4.23.242) |
| 06:03.00 | dieno | unable to connect remote |
| 06:03.27 | anthonyu | did you control-c out of the asterisk -vvvvv? |
| 06:03.30 | dieno | does /var/run/asterisk.ctl exist? |
| 06:03.37 | dieno | nop |
| 06:03.51 | anthonyu | do 'ps -ef |grep asterisk' |
| 06:04.44 | dieno | root 5040 |
| 06:05.09 | anthonyu | there's gotta be more than that |
| 06:05.16 | dieno | root 5040 30776 0 11:04 pts/1 00:00:00 grep asterisk |
| 06:05.22 | anthonyu | oh, ok |
| 06:05.47 | anthonyu | as root, run 'asterisk', then 'ps -ef |grep asterisk' |
| 06:06.13 | dieno | still same thing |
| 06:06.36 | anthonyu | it's not working then |
| 06:06.43 | dieno | ohke |
| 06:06.45 | dieno | :( |
| 06:06.49 | dieno | now |
| 06:06.54 | anthonyu | 'tail /var/log/asterisk/messages' |
| 06:08.21 | dieno | notice[5054] res_odbc.c:adding ENV var: inform |
| 06:08.48 | anthonyu | use the pastebin |
| 06:09.03 | dieno | :) |
| 06:09.05 | dieno | ok |
| 06:11.56 | dieno | i hav post my eror |
| 06:12.02 | dieno | can u take alook plz |
| 06:12.32 | *** join/#asterisk j0 (n=dan@CABLE-72-53-45-212.cia.com) |
| 06:12.42 | anthonyu | i think those errors are ok |
| 06:13.06 | anthonyu | why do you have informix in there? |
| 06:13.49 | dieno | wats da |
| 06:13.52 | dieno | :( |
| 06:13.59 | anthonyu | its a database |
| 06:14.00 | *** join/#asterisk dineo2 (n=dineo2@124.29.194.150) |
| 06:14.11 | dieno | ok how do i remove it |
| 06:14.22 | anthonyu | i don't know, i don't have that |
| 06:14.49 | anthonyu | you have left my area of expertise |
| 06:14.55 | anthonyu | brb |
| 06:15.11 | dieno | ok |
| 06:20.18 | dieno | ok how do i reastart asterisk |
| 06:23.46 | dineo2 | <PROTECTED> |
| 06:33.22 | *** join/#asterisk Druken (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 06:35.28 | Druken | argh... ok... i'm having some major issues with upgrading asterisk... anyone feel like troubleshooting for me? |
| 06:40.15 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 06:40.47 | *** join/#asterisk adelas (n=booger@rrcs-24-199-21-138.west.biz.rr.com) |
| 06:41.45 | j0 | meh i just rebooted... and my voice quality is shot!.. damn upgrades |
| 06:41.55 | adelas | hey guys, when asterisk has sql interace with web, is there anyway to get the web interface login/pass? |
| 06:44.00 | *** join/#asterisk adorah (n=Administ@84.94.133.214.cable.012.net.il) |
| 06:56.14 | *** join/#asterisk af_ (n=af@ip-170-156.sn1.eutelia.it) |
| 06:56.23 | *** join/#asterisk profounded (n=profound@ool-44c4eae2.dyn.optonline.net) |
| 06:58.24 | *** join/#asterisk marta (n=a@host100-5.pool8711.interbusiness.it) |
| 06:59.23 | marta | hi |
| 06:59.29 | *** join/#asterisk novafirst (n=kosta@wrt1.niclab.com) |
| 07:00.04 | novafirst | what is this used for ? app_dictate.so |
| 07:16.13 | shodan | it's 3h14 |
| 07:16.34 | shodan | how abou some pi music ! |
| 07:16.34 | shodan | http://pi.ytmnd.com/ |
| 07:34.53 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43-static.72-81-b.business.telecomitalia.it) |
| 08:29.47 | *** join/#asterisk chapeaurouge (n=chapeaur@84.5.49.96) |
| 08:36.32 | *** join/#asterisk Tili (n=tili@202.133.67.223) |
| 08:51.31 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkit.com.au) |
| 09:07.48 | *** join/#asterisk MatsK (i=MatsK@83.233.97.229) |
| 09:16.15 | *** join/#asterisk bethaud (n=bethaud@host-84-9-82-198.bulldogdsl.com) |
| 09:17.08 | bethaud | Figures out my difficulty with my X101P card |
| 09:17.20 | *** join/#asterisk rephorm (n=rephorm@cpe-24-27-8-18.austin.res.rr.com) |
| 09:18.17 | bethaud | Turns out it has the Si3012 line chip which is impedance matched for 600 ohms, that's the US standard. I'm in the UK, so I need a card with the Si3014 line chip on it. |
| 09:19.22 | bethaud | Don't suppose anynody knows of a source of cards with that chip on them? |
| 09:20.34 | *** join/#asterisk svenna_ (n=svenna@p548D373F.dip0.t-ipconnect.de) |
| 09:29.15 | *** part/#asterisk bethaud (n=bethaud@host-84-9-82-198.bulldogdsl.com) |
| 09:32.45 | *** join/#asterisk daysmen3 (n=primus@host86-143-4-189.range86-143.btcentralplus.com) |
| 09:35.07 | *** join/#asterisk Sarum4n (n=some@saruman.demon.nl) |
| 09:44.03 | *** join/#asterisk L-info (n=Adam@62.69.102.99) |
| 10:12.13 | *** join/#asterisk FTexcom (n=strapse@169.Red-88-2-81.staticIP.rima-tde.net) |
| 10:12.17 | FTexcom | hello |
| 10:15.07 | Sarum4n | lo |
| 10:22.48 | tzafrir_home | Is it just me or the asterisk mailing lists are not delivering all the messages? I have quite a few messages ithout replies |
| 10:52.25 | BugKham | anyone knows if Set(DNID=${EXTEN}) works? |
| 10:54.24 | BugKham | it doesn't for me, by the way |
| 11:21.33 | FTexcom | how can a extension pick up a ringing extension that's not his own? |
| 11:44.51 | *** join/#asterisk af_ (n=af@ip-170-156.sn1.eutelia.it) |
| 11:46.52 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 11:48.48 | areq | hello |
| 11:49.13 | areq | I have strange problem with asterisk + Sangoma A101 + E1 + 100 DDI, when someone call me from ISDN phone with display when asterisk answer from their display disappear two last digits |
| 11:51.13 | RoyK | areq: what does the pri debug show? |
| 11:51.14 | areq | probably my asterisk should tell which DDI receive connection |
| 11:51.25 | RoyK | pri debug span 1 |
| 11:51.27 | RoyK | and make a call |
| 11:56.28 | *** join/#asterisk kamuix (n=kamuix@55.72.209.88.dslw0.libello.mc) |
| 11:59.03 | areq | RoyK: http://pastebin.com/776328 |
| 12:01.18 | RoyK | < Called Number (len=12) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '421111199' ] |
| 12:01.32 | *** join/#asterisk matmo (n=chatzill@doconnel.force9.co.uk) |
| 12:01.48 | RoyK | and on the users's display, it says what? |
| 12:02.04 | areq | without last 99 |
| 12:02.50 | RoyK | 421112629 is calling 421111199 here, so on the user's display it should show 421112629 |
| 12:02.53 | RoyK | right? |
| 12:03.57 | areq | no, I tall about display on source phone |
| 12:05.07 | areq | callerid on sip phones and outgoing connection works fine |
| 12:06.02 | matmo | sry if wrong place but I only recently discover * and have searched all over: I have a mac mini (jaguar) and need to know if * can use it's modem to log voice messages? |
| 12:09.01 | BugKham | anyone knows if Set(DNID=${EXTEN}) works? |
| 12:09.39 | areq | RoyK: somebody call me: put 421111199, and see it - until asterisk do exten => _4211111XX,n,Answer - in this moment hi see 4211111 on display |
| 12:09.44 | *** join/#asterisk asterisk88 (n=malik@124.29.198.146) |
| 12:09.48 | asterisk88 | hi |
| 12:10.20 | asterisk88 | ive installed 1.2.10 and having trouble with AGI |
| 12:10.32 | asterisk88 | with the CDR function specifically |
| 12:10.39 | asterisk88 | can someone help please |
| 12:14.26 | areq | any idea ? |
| 12:17.04 | *** join/#asterisk adorah (n=Administ@84.94.133.214.cable.012.net.il) |
| 12:19.54 | matmo | any mac users here? |
| 12:20.10 | asterisk88 | Undefined subroutine &main::CDR |
| 12:20.19 | asterisk88 | areq |
| 12:26.29 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust944.leed.cable.ntl.com) |
| 12:28.38 | *** join/#asterisk stephane_ (n=stephane@merlin.cabale.net) |
| 12:29.24 | *** join/#asterisk jhiver (n=jhiver@LReunion-151-2-164.w193-253.abo.wanadoo.fr) |
| 12:29.27 | jhiver | hi all |
| 12:29.35 | *** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk) |
| 12:29.44 | jhiver | i have a trick question :) |
| 12:30.12 | jhiver | i was wondering if there was a way to retrieve the IP address of the calling pary (assuming it's not coming from Zap) from within the dialplan |
| 12:30.32 | jhiver | any ideas? |
| 12:31.41 | RoyK | matmo: i use a mac |
| 12:32.03 | RoyK | matmo: but i seriously doubt you'll manage to make asterisk interface with a regular modem on a mac |
| 12:34.32 | matmo | RoyK: thanks. Any reason why not? It just seems a regular thing to do. |
| 12:35.48 | jhiver | so... nobody knows? the idea would be to have no auth. from within the sip.conf and call a fastagi sip to let me know if this IP is (or not) authorized to connect |
| 12:36.39 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 12:37.15 | benjk_ | matmo: we're working on a modem arbitrator for the Motorola SM56 based Apple modem which will feed into Unicall |
| 12:38.09 | matmo | benjk: thanks for the info. Any idea when this may be available? |
| 12:38.14 | benjk_ | important to notice that this sort of thing is only possible with SM56 based modems |
| 12:38.41 | benjk_ | because any other modems used in Macs did not support voice |
| 12:38.44 | matmo | hmmm, don't know the model in the mini but I'll check it out |
| 12:39.02 | benjk_ | the mini has the Motorola modem |
| 12:39.13 | matmo | no voice? |
| 12:39.22 | benjk_ | no the Mini is fine |
| 12:39.34 | matmo | great |
| 12:39.37 | benjk_ | The Motorola SM56 supports voice |
| 12:40.07 | benjk_ | also you can always buy the Apple USB modem dongle which has the same Motorola SM56 unit inside and uses the same driver |
| 12:40.26 | benjk_ | this will only work with Unicall though |
| 12:41.28 | benjk_ | we have a working port of Unicall that works with AstXcode |
| 12:42.14 | *** join/#asterisk Ciber311 (n=Ciber311@user-1087e94.cable.mindspring.com) |
| 12:43.12 | benjk_ | I can't make any promises when this will be released because its a part time effort but I would hope within the next 2 months or so |
| 12:44.30 | benjk_ | you may want to check this page from time to time for news: http://www.sunrise-tel.com/unicall-support.html |
| 12:44.47 | matmo | sry, I'm a complete newby to *. The mini sits around doing not very much (in the corner of a spare room (sad little b*stard)). Recently I discover it did Caller ID then got to wondering if it could log voice calls, then if it could route them. At which point I came across * |
| 12:45.20 | matmo | benjk: yep already found that and many other sites, just couldn't find an answer to my q |
| 12:46.03 | matmo | thanks for the info. Look forward to its release |
| 12:46.21 | benjk_ | I have played around with the now deprecated chan_modem at some point |
| 12:46.39 | matmo | and it does? |
| 12:46.42 | benjk_ | its possible to get that to work with the Motorola SM56 modem |
| 12:46.49 | benjk_ | that's half duplex only |
| 12:47.05 | benjk_ | but if all you want to do is voicemail, then that's ok |
| 12:47.58 | matmo | yeah, I don't use it to connect to the net (I have adsl for that) so just voice |
| 12:48.03 | benjk_ | but the chan_modem driver is very badly written, it mixes two different approaches how to read/write to the modem socket and this screws up on OSX |
| 12:48.42 | matmo | any recommendations for sw dedicated to voice logging? |
| 12:49.02 | benjk_ | I got some experiments working that showed it was possible to do, but it got out of hand timewise and so I didn't follow through with it |
| 12:49.36 | benjk_ | do you mean monitoring and recording an external call on the line? |
| 12:50.20 | benjk_ | for that sort of thing you'd need special hardware |
| 12:50.39 | benjk_ | what could be done with the chan_modem plugin is voicemail |
| 12:50.53 | benjk_ | that is, somebody calls in, leaves a message, hangs up |
| 12:50.57 | matmo | yep, an expensive answering machine! But seriously, think of adding Lazslo(?) so I can listen to message from pc elsewhere in house |
| 12:51.55 | benjk_ | what can be done with the Unicall conduit is true telephony, using the Mac Mini as a SOHO PBX switching calls through it |
| 12:52.00 | matmo | (sry, "I am thinking") |
| 12:52.34 | matmo | that would be good also. Will it be possible on a base mac, ie, 256M ram? |
| 12:52.41 | benjk_ | sure |
| 12:53.18 | benjk_ | unless you want to do video editing and other resource hungry stuff on that Mini at the same time |
| 12:54.15 | matmo | no, I use it mainly as a file server. Was going to get into mac-dev'ing at one point but it seemed just an academic exercise (job-wise) |
| 12:54.53 | benjk_ | you are welcome to join us working on AstXcode |
| 12:55.10 | benjk_ | http://www.sunrise-tel.com/screenshots/Asterisk-Xcodified.2.png |
| 12:56.57 | matmo | he he, would love to but I wouldn't know where to start now. These days I'm restricting my interests (primarily to learning Ruby) |
| 12:57.08 | RoyK | benjk_: are you doing this on linux or os x? |
| 12:57.24 | benjk_ | Xcode doesn't work on Linux |
| 12:58.03 | RoyK | :P |
| 12:58.03 | RoyK | no..... |
| 12:58.11 | benjk_ | matmo, you might want to take a look at Io |
| 12:58.22 | benjk_ | www.iolanguage.com |
| 12:58.57 | benjk_ | I am hoping to embed this into Asterisk as a replacement for the dialplan language |
| 12:59.05 | RoyK | benjk_: does xcode manage patches and so on any good? how about svn integration? |
| 12:59.20 | matmo | looks interesting, I'll check it out |
| 12:59.21 | benjk_ | its got both CVS and SVN integration |
| 12:59.58 | RoyK | how do you integrate such as project into xcodetools? |
| 13:00.14 | benjk_ | Io is interesting for telephony because of its concurrency model, it has coroutines built in |
| 13:01.11 | benjk_ | RoyK, I didn't, I recreated the project in Xcode, there is no makefile, no command line, just Xcode |
| 13:01.22 | RoyK | ok |
| 13:01.31 | matmo | how does it compare to Ruby? |
| 13:01.48 | benjk_ | there is a comparison to other languages section on the Io site |
| 13:01.59 | benjk_ | somewhere in the faq section I think |
| 13:02.03 | RoyK | Io looks neat |
| 13:02.37 | RoyK | i heard something some time back that someone was working on an erlang interface |
| 13:02.46 | benjk_ | the interesting thing as a dialplan engine is that it is very small, ideal for embedding and its concurrency model which is ideal for telephony |
| 13:03.07 | benjk_ | Io is like Erlang in respect of concurrency, but it is much leaner and easier to embed |
| 13:03.20 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
| 13:03.27 | benjk_ | Erlang is not embeddable, there is a special dialect of it for embedding and that is still rather heavy |
| 13:03.39 | RoyK | ok |
| 13:03.39 | matmo | About all I remember about Erlang is its size |
| 13:03.51 | RoyK | also, erlang is ugly :P |
| 13:03.51 | benjk_ | yeah, Io is tiny |
| 13:03.56 | benjk_ | and that too |
| 13:04.12 | *** join/#asterisk Ciber311 (n=Ciber311@user-1087e94.cable.mindspring.com) |
| 13:04.18 | coppice | I worry about performance when they name a language after a guy famous for queuing :-) |
| 13:04.22 | benjk_ | but you can expect to get the same kind of performance from it |
| 13:05.06 | florz | coppice: But it's just named after the company! ;-) |
| 13:05.21 | benjk_ | you should be more worried about performance with all that OS threading nonsense that is going on everywhere |
| 13:05.27 | matmo | hmmm, it claims Ruby is not pure OO, I thought it was! |
| 13:06.29 | coppice | well, anyone is free to define OO as they plase, so you can define any language to be/not to be pure OO |
| 13:06.49 | matmo | "everything is an object" |
| 13:07.01 | *** join/#asterisk dineo2 (n=dineo2@124.29.194.150) |
| 13:07.12 | coppice | except my wife. I would never treat her as an object :-) |
| 13:07.20 | dineo2 | Aug 26 11:38:21 NOTICE[5500] cdr.c: CDR simple logging enabled. |
| 13:07.20 | matmo | :-) |
| 13:07.20 | dineo2 | Aug 26 11:38:21 WARNING[5500] res_musiconhold.c: Unable to open pseudo channel for timing... Sound may be choppy. |
| 13:07.20 | dineo2 | Aug 26 11:38:22 ERROR[5500] res_config_mysql.c: MySQL RealTime: Failed to connect database server on (err 2002). Check debug for more info. |
| 13:07.20 | dineo2 | Aug 26 11:38:22 WARNING[5500] res_config_mysql.c: MySQL RealTime: Couldn't establish connection. Check debug. |
| 13:07.21 | dineo2 | Aug 26 11:38:22 NOTICE[5500] config.c: Registered Config Engine mysql |
| 13:07.23 | dineo2 | Aug 26 11:38:22 WARNING[5500] pbx_dundi.c: Unable to look up host 'dhcppc2' |
| 13:07.25 | dineo2 | Aug 26 11:38:23 WARNING[5500] chan_mgcp.c: Unable to get our IP address, MGCP disabled |
| 13:07.26 | RoyK | florz: from http://en.wikipedia.org/wiki/Erlang_programming_language, "Erlang is named after A. K. Erlang. It is sometimes thought that its name is an abbreviation of ERicsson LANGuage, owing to its heavy use inside Ericsson. According to Bjarne Däcker who headed the Computer Science Lab at the time, this duality is intentional." |
| 13:07.27 | dineo2 | Aug 26 11:38:23 ERROR[5500] chan_unicall.c: Unable to load config unicall.conf |
| 13:07.32 | dineo2 | Aug 26 11:38:23 WARNING[5500] loader.c: chan_unicall.so: load_module failed, returning -1 |
| 13:07.32 | RoyK | ~pb |
| 13:07.34 | jbot | somebody said pb was a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/, or http://www.paste-it.net/ |
| 13:07.34 | dineo2 | Aug 26 11:38:23 WARNING[5500] loader.c: Loading module chan_unicall.so failed! |
| 13:07.35 | benjk_ | dineo2 dont do that! |
| 13:07.36 | dineo2 | can ne 1 tell wats does da stands for |
| 13:07.41 | RoyK | ~pasterbin |
| 13:07.47 | dineo2 | ok sry |
| 13:07.50 | benjk_ | use pastebin |
| 13:07.53 | dineo2 | -pastebin |
| 13:07.56 | RoyK | ~pb |
| 13:07.57 | jbot | i guess pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/, or http://www.paste-it.net/ |
| 13:07.59 | dineo2 | ohh da website |
| 13:08.02 | dineo2 | sry |
| 13:08.11 | benjk_ | you are not supposed to flood the channel |
| 13:08.20 | dineo2 | ohh i am really sry for dat |
| 13:08.21 | florz | RoyK: Yeah, sure it is. At least it's claimed to be. So it probably isn't. Well, whatever ... ;-) |
| 13:08.40 | RoyK | :P |
| 13:08.43 | RoyK | bloody swedes |
| 13:09.30 | dineo2 | http://pastebin.com/776364 |
| 13:09.32 | florz | Anyway, this Io stuff looks like a nice thing to have in * ... |
| 13:09.34 | dineo2 | plz chek diz out |
| 13:09.36 | coppice | there used to be a database with a programming language called "English" They tried to make this a registered trade mark :-) |
| 13:10.06 | benjk_ | we just need res_io.so for starters |
| 13:10.16 | *** join/#asterisk nailbags (i=someone@c220-237-123-137.randw1.nsw.optusnet.com.au) |
| 13:10.40 | dineo2 | can n e plz consider on my error plz http://pastebin.com/776364 |
| 13:11.00 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:13.36 | dineo2 | haleo |
| 13:13.43 | dineo2 | is n e 1 there |
| 13:14.02 | blitzrage | nope |
| 13:14.11 | dineo2 | o k |
| 13:14.16 | dineo2 | so u can help me http://pastebin.com/776364 |
| 13:14.16 | dineo2 | :D |
| 13:14.59 | blitzrage | you have a bunch of errors there |
| 13:15.11 | dineo2 | i just wan to run it |
| 13:15.19 | dineo2 | whichi should i remove |
| 13:15.41 | blitzrage | ummm... did you configure any of those modules? Because none of them are configured right |
| 13:15.58 | dineo2 | nop |
| 13:16.02 | blitzrage | go to modules.conf and do a noload => <module>.so for all those modules since none of them are configured right |
| 13:16.13 | dineo2 | i never touched n e modules |
| 13:16.13 | dineo2 | ok |
| 13:16.31 | blitzrage | I don't think Asterisk tries to load realtime by default |
| 13:16.36 | blitzrage | at least it shouldn't... |
| 13:17.09 | dineo2 | can u tell me the default conf so i can let them on |
| 13:17.28 | blitzrage | the default config is what comes with 'make samples' |
| 13:17.44 | dineo2 | ok |
| 13:18.45 | dineo2 | ok now how do i restart my asterisk |
| 13:18.52 | dineo2 | plz telll me the command |
| 13:20.37 | blitzrage | wow... you really need to go read: |
| 13:20.39 | blitzrage | ~book |
| 13:20.41 | jbot | i heard book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 13:20.44 | dineo2 | haha |
| 13:20.49 | dineo2 | lol |
| 13:20.56 | blitzrage | seriously |
| 13:21.01 | blitzrage | 'restart now' |
| 13:21.05 | dineo2 | in asterisk i am new |
| 13:21.12 | blitzrage | right -- even more reason to READ |
| 13:21.15 | benjk_ | thats why you should read up |
| 13:21.20 | dineo2 | ok |
| 13:21.24 | dineo2 | thnx man |
| 13:21.55 | benjk_ | there's also www.voip-info.org |
| 13:22.04 | *** part/#asterisk kumamoto (n=eryco@24-178-2-212.dhcp.ftwo.tx.charter.com) |
| 13:22.04 | dineo2 | yea i read al the thing |
| 13:22.11 | dineo2 | except commands :D |
| 13:22.15 | RoyK | reading is for chickens. manuals is for idiots that haven't learned how to they can just get all the knowledge they want just by whining on IRC |
| 13:24.27 | blitzrage | then there's the lesson that you just have to ignore some people |
| 13:24.38 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 13:25.47 | RoyK | wtf is codec_zap? |
| 13:26.40 | blitzrage | is it new? |
| 13:26.46 | benjk_ | its when you're done with your codec and you no longer need it, you just zap it |
| 13:28.07 | RoyK | it's in trunk |
| 13:28.15 | blitzrage | I suppose you could always *read* the changlogs |
| 13:28.29 | blitzrage | I know, I know... BLASPHEMY |
| 13:29.02 | matmo | re IO: "operator precedence is a major headache" |
| 13:31.17 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 13:38.23 | *** join/#asterisk Ciber311 (n=Ciber311@user-1087e94.cable.mindspring.com) |
| 13:39.05 | RoyK | http://debri.ru/temp/pics/060824-194354.jpg |
| 13:39.40 | matmo | good pic! |
| 13:40.31 | Ciber311 | lol |
| 13:40.35 | Ciber311 | so i umm |
| 13:40.43 | Ciber311 | disabled usb legacy support |
| 13:40.48 | Ciber311 | hoping it would fix lcc |
| 13:40.54 | Ciber311 | now i can't get into the bios :P |
| 13:43.07 | *** join/#asterisk Druken (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 13:43.32 | Druken | morning everyone, does anyone know what the licensing costs are for g729 and how i obtain them ? |
| 13:44.54 | RoyK | Druken: $10 per concurrent translation. you get them from www.digium.com |
| 13:46.17 | FTexcom | is it true that in europe g729 is free? |
| 13:46.29 | coppice | nope |
| 13:47.02 | FTexcom | I heard that because of patent legislation in europe, g729 can be free |
| 13:47.21 | coppice | You hear lots of crap from the clueless |
| 13:47.37 | FTexcom | k... |
| 13:48.08 | asterisk88 | hi guys |
| 13:48.28 | asterisk88 | im having problems with realtime CDR |
| 13:48.32 | coppice | Some people seem convinced that because purely software techniques cannot be patented in .eu, then nothing within a few metres of a computer can. :-) |
| 13:49.06 | asterisk88 | basically the CDR function in asterisk 1.2.10 |
| 13:49.16 | asterisk88 | <PROTECTED> |
| 13:49.16 | asterisk88 | <PROTECTED> |
| 13:49.42 | asterisk88 | these two variables are not returning the correct values.. can you guys help me with this |
| 13:49.48 | asterisk88 | <PROTECTED> |
| 13:50.29 | asterisk88 | im getting the correct values for other variables but the duration variables are being a pain |
| 13:52.04 | FTexcom | anyone knows how to change the date format on the email that voicemail sends you? |
| 13:52.47 | RoyK | asterisk88: wtf are you doing? There are apps and functions for CDR manipulation.... |
| 13:53.20 | RoyK | FTexcom: RTFS :P |
| 13:53.30 | asterisk88 | trying to execute them with ag |
| 13:53.32 | asterisk88 | agi* |
| 13:54.01 | asterisk88 | $var1 = $AGI->get_variable('CDR_ANSWER'); |
| 13:54.27 | asterisk88 | here's the exten (database entry) CDR_ANSWER=${CDR(answer)} |
| 13:54.30 | RoyK | is there such a variable? i've never heard of it... |
| 13:55.12 | RoyK | CDR(answer) might be better |
| 13:55.15 | asterisk88 | exten => h,n,Set(CDR_DST=${CDR(dst)}) |
| 13:55.24 | asterisk88 | that's the normal exten.conf definition |
| 13:55.57 | RoyK | but why stuff it in a variable in the first place, when it's already in CDR()? |
| 13:56.42 | *** join/#asterisk Druken (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 13:56.46 | FTexcom | RoyK: read the fucking scrip? |
| 13:56.51 | Druken | god damn, i tell ya, wireless sucks ass |
| 13:57.06 | asterisk88 | i tried using the function in the agi.. i couldnt find the correct syntax |
| 13:57.25 | asterisk88 | AGI->CDR(answer); |
| 13:57.37 | asterisk88 | is that how you use it in the AGI? |
| 13:57.59 | RoyK | ~rtfs |
| 13:58.02 | jbot | hmm... rtfs is probably read the f*cking source... |
| 13:58.03 | RoyK | :) |
| 13:58.13 | asterisk88 | :) thanks |
| 13:58.56 | asterisk88 | asterisk-1.2.10 README |
| 13:58.57 | asterisk88 | ? |
| 13:59.00 | RoyK | asterisk88: $AGI->get_variable('CDR(answer)') |
| 13:59.01 | RoyK | iirc |
| 13:59.44 | asterisk88 | thanks roy i'll just try this |
| 14:02.07 | Druken | ok, so once i order these damn codecs, how long does it take for them to process the damn order? |
| 14:03.01 | *** join/#asterisk marv (n=ilovekim@c-71-228-189-127.hsd1.al.comcast.net) |
| 14:04.32 | RoyK | Druken: pretty fast. or do you use realtime? |
| 14:05.53 | Druken | i used to use realtime.... |
| 14:05.56 | RoyK | s/Dru/Drun/ |
| 14:06.03 | Druken | but it was too much of a pain in the ass :) |
| 14:06.32 | asterisk88 | RoyK: $AGI->get_variable('CDR(answer)') not working :( |
| 14:06.33 | RoyK | Druken: not as much as the PITA with config files and 5k users |
| 14:06.48 | *** join/#asterisk negativecreep (n=xaeem@210.2.151.110) |
| 14:06.53 | RoyK | asterisk88: anyway, what are you trying to do+ |
| 14:06.53 | RoyK | ? |
| 14:07.00 | negativecreep | I see no debug messages on my asterisk console.. |
| 14:07.17 | RoyK | negativecreep: set verbose=9999999999999 |
| 14:07.23 | marta | cle |
| 14:07.24 | negativecreep | RoyK: killer. |
| 14:07.27 | marta | ops :) |
| 14:07.29 | RoyK | s/=/ / |
| 14:08.04 | bkw__ | that does nothing really |
| 14:08.11 | bkw__ | 100 would be just as effective :P |
| 14:08.22 | Druken | i use 13 :) |
| 14:08.25 | Druken | magic number |
| 14:08.27 | negativecreep | I had 37 |
| 14:08.27 | bkw__ | I use 666 |
| 14:08.30 | negativecreep | still see nothing.. |
| 14:08.34 | bkw__ | or 42 |
| 14:08.42 | bkw__ | did you start asterisk with -vvvgc? |
| 14:08.50 | *** join/#asterisk nettie (n=nettie@85-18-54-38.ip.fastwebnet.it) |
| 14:09.15 | Druken | so am i a dreamer in thinking i'll get these codecs today? hehe |
| 14:09.16 | asterisk88 | RoyK: need the duration & billsec values to bill for calls |
| 14:09.30 | RoyK | in realtime? |
| 14:09.36 | RoyK | callingcard stuff? |
| 14:09.39 | asterisk88 | RoyK: yes in realtime |
| 14:09.40 | bkw__ | Druken, what you need? |
| 14:09.51 | Druken | bkw__: g729.... |
| 14:09.52 | asterisk88 | RoyK: calling card kind of |
| 14:10.22 | anthonyu | isn't that what the cdr stuff is for? |
| 14:10.29 | matmo | thanks for all the help, bye |
| 14:10.30 | Druken | well,.... that and a million dollars.... |
| 14:10.33 | *** part/#asterisk matmo (n=chatzill@doconnel.force9.co.uk) |
| 14:10.44 | asterisk88 | it returns the call stats doesnt it? |
| 14:10.53 | asterisk88 | duration/length, seconds etc |
| 14:11.00 | RoyK | asterisk88: then pass ${CDR(whatever)} as an argument to the agi script :) |
| 14:11.01 | RoyK | i do that |
| 14:11.36 | asterisk88 | through realtime ext? how ? |
| 14:12.30 | RoyK | exten => h,1,DeadAGI(bill.agi,${DIALSTATUS},${CARDNUMBER},${CCSTARTCOST},${CCPRICEPERMINUTE},${CCCURRENCY},${CDR(billsec)}) |
| 14:12.35 | RoyK | not realtime ext |
| 14:12.38 | RoyK | that's crap |
| 14:12.39 | asterisk88 | script.pl|${EXTEN}|${CALLERIDNUM}|$ccing |
| 14:12.54 | asterisk88 | got it.. great. thanks RoyK |
| 14:13.32 | *** join/#asterisk brif8 (n=Administ@ns1.ttienterprises.org) |
| 14:14.01 | *** join/#asterisk _DAW (n=_DAW@adsl-222-12-239.msy.bellsouth.net) |
| 14:15.03 | _DAW | HI all.. Can someone tell me if * currently supports sip diversion header? |
| 14:15.26 | Druken | and that is? |
| 14:16.35 | _DAW | r u asking me? |
| 14:16.47 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 14:16.47 | *** part/#asterisk brif8 (n=Administ@ns1.ttienterprises.org) |
| 14:19.51 | asterisk88 | ($phoneno, $calleridnum, $billsec, $duration) = @ARGV; |
| 14:19.57 | asterisk88 | in the AGI |
| 14:20.48 | asterisk88 | exten => 12345,1,DeadAGI(tel.pl|${EXTEN}|${CALLERIDNUM)}|${CDR(billsec)}|${CDR(duration)} |
| 14:21.28 | blitzrage | I woke up way too early.. |
| 14:22.16 | negativecreep | I still dont see anything.. |
| 14:22.18 | negativecreep | this is weird. |
| 14:22.20 | negativecreep | just too weird |
| 14:22.39 | docelmo | _DAW to my knowledge no.. but go ask #asterisk-dev they wrote it |
| 14:27.15 | negativecreep | my iax registers fine with asterisk.. |
| 14:27.26 | negativecreep | but when I dial a number and make a call..i dont see any logs.. |
| 14:27.32 | negativecreep | the call says no route |
| 14:27.41 | negativecreep | still asterisk should show something on its console. |
| 14:31.09 | blitzrage | lol |
| 14:31.23 | *** join/#asterisk Nukemizer (n=Nuke@160.7.239.13) |
| 14:31.39 | *** join/#asterisk Druken (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 14:32.08 | negativecreep | blitzrage: ? |
| 14:35.27 | blitzrage | wrong window :) |
| 14:35.34 | *** join/#asterisk anthonyl (n=rmerz@c-24-12-179-217.hsd1.il.comcast.net) |
| 14:36.08 | *** join/#asterisk yogurt2ungue (n=yogurt2u@24-48-231-201.fibertel.com.ar) |
| 14:36.49 | negativecreep | k |
| 14:41.13 | nettie | guys, I'm debugging the SIP channel I use to terminate my calls to the PSTN (voip carrier) because during the day some of the calls are not going trough. I figured out that when this happens asterisk get a SIP/2.0 404 Not Found message instead of the usingal SIP/2.0 100 trying -- your call is important to us. I'm wondering what's actually causing it. |
| 14:41.55 | nettie | Anyone know if I should just send the error message to my carrier and try to figure out what's wrong togethere? |
| 14:42.41 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 14:42.41 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:50.27 | *** join/#asterisk bethaud (n=bethaud@host-84-9-82-198.bulldogdsl.com) |
| 14:51.23 | *** part/#asterisk gJon (n=jellis@dev.integrumtech.com) |
| 15:03.43 | *** join/#asterisk stephane_ (n=stephane@merlin.cabale.net) |
| 15:13.32 | *** join/#asterisk crlshn (i=kvirc@operaciones3.globalnet.hn) |
| 15:15.59 | *** join/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com) |
| 15:16.07 | *** join/#asterisk eKo1 (n=eKo1@190.4.7.90) |
| 15:16.39 | EmleyMoor | I'm trying, using the Debian testing packages, to get started with asterisk - I have no fxo or fxs so just pressed right on to SIP |
| 15:17.09 | EmleyMoor | I have set up pretty much what it shows - but cannot get linphone to authenticate |
| 15:17.11 | *** join/#asterisk Makenshi (n=makenshi@2001:630:1c0:2006:20c:29ff:fe7f:ba41) |
| 15:17.41 | EmleyMoor | What is the difference between "username" and "userid", and where are they defined? |
| 15:17.55 | EmleyMoor | (page 68 of book, 86 of PDF) |
| 15:18.42 | EmleyMoor | linphone constantly requests: "Authentication required from realm "asterisk" |
| 15:23.13 | *** part/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com) |
| 15:25.43 | *** part/#asterisk negativecreep (n=xaeem@210.2.151.110) |
| 15:36.47 | *** join/#asterisk L-info (n=Adam@62.69.102.99) |
| 15:38.58 | *** join/#asterisk DaveCanoe (n=Dave@H5.C74.B0.tor.eicat.ca) |
| 15:44.01 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 15:45.08 | *** join/#asterisk dlynes_laptop (n=dlynes@S01060016b6c052ee.vc.shawcable.net) |
| 15:48.18 | DaveCanoe | OK. Repeat question from last night. I have a setup where the call path goes phoneA->asteriskA->asteriskB->asteriskA->phoneB. All the inter-asterisk links are IAX2 and all the phone links are SIP. The reason for this circuitous route is that asteriskB does all the call routing and asteriskA's only job is IAX trunking. However, the IAX transfer to take asteriskB out of the loop after the call is setup fails. |
| 15:52.29 | *** join/#asterisk RoyK (n=roy@a217-118-45-74.bluecom.no) |
| 15:54.57 | *** join/#asterisk Damin_PDA (n=pocketir@89.sub-70-217-213.myvzw.com) |
| 15:55.21 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 15:55.54 | *** join/#asterisk FullService (n=jiggaman@CPE0002724fc55e-CM000f211fd29c.cpe.net.cable.rogers.com) |
| 15:57.01 | *** join/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com) |
| 15:57.30 | EmleyMoor | OK - giving up on SIP at least for the moment - how do I define iax phones in asterisk? |
| 15:57.46 | DaveCanoe | iax.conf |
| 15:57.48 | *** join/#asterisk adorah (n=Administ@84.94.133.214.cable.012.net.il) |
| 15:59.39 | RoyK | can someone give me a single good reason to use IAX2 instead of SIP except for trunking and NAT traversal? |
| 16:00.36 | EmleyMoor | DaveCanoe: Yes, you do them in iax.conf but how? Is there an example? |
| 16:01.22 | *** join/#asterisk toerkeium (i=oo@201.216.206.221) |
| 16:01.28 | JunK-Y | RoyK: if u only do voice, why use a monster? |
| 16:01.34 | EmleyMoor | RoyK: I would say "because SIP doesn't work" but I know it does work - just not for me - at least not as well as I had expected |
| 16:02.17 | EmleyMoor | DaveCanoe: Is it by and large the same as the definition for a SIP phone? |
| 16:02.31 | DaveCanoe | In this example, I modified iax.conf on each machine to have either one "friend" or a pair from friend/peer. |
| 16:02.37 | RoyK | EmleyMoor: well, we have thousands of customers on SIP, and it works. |
| 16:02.48 | DaveCanoe | The calls go through fine --- everything is good accept the "cannot transfer" message |
| 16:03.04 | RoyK | JunK-Y: there aren't too much TE supporting iax2, though. |
| 16:03.33 | RoyK | ~ping key2 |
| 16:03.43 | jbot | pong key2 |
| 16:04.13 | RoyK | ~lart himself |
| 16:04.30 | EmleyMoor | If I want to set up to use an IAX2 softphone, how do I set asterisk up for it? |
| 16:04.38 | RoyK | ~rtfm |
| 16:04.41 | jbot | somebody said rtfm was Read The F*cking Manual (TM). It is a suggestion to do your homework before posting a question. Sometimes used as RTFM $SPECIFIC_MANUAL to refer to a specific source of information. See also http://uncyclopedia.org/wiki/RTFM. |
| 16:04.41 | RoyK | ~book |
| 16:04.43 | jbot | from memory, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 16:04.43 | RoyK | ~docs |
| 16:04.46 | jbot | it has been said that docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 16:05.16 | EmleyMoor | OK - what chapter in the book covers IAX2 phones? |
| 16:05.26 | RoyK | I beleive there's an index there |
| 16:05.33 | JunK-Y | RoyK: i just think for ppl doing voice, IAX2 is more appropriate. |
| 16:05.56 | RoyK | are you positive? |
| 16:06.34 | JunK-Y | whatcha mean? |
| 16:07.09 | RoyK | well, how can iax2 be more appropriate, having SIP being well standardised and lots of equipment available for it? |
| 16:07.13 | *** join/#asterisk dimas (n=ds@81.18.135.125) |
| 16:07.52 | RoyK | for NAT or trunking, yeah, but networked clients, you don't want to stick to those few iax2 ATA vendors available |
| 16:08.09 | bethaud | Anybody know wgere to get an X101P with th4 Si3014 line chip? |
| 16:08.14 | JunK-Y | i just hope, we'll see large IAX2 gateways. |
| 16:08.37 | RoyK | are there any more news about sending custom arguments with iax2? |
| 16:08.40 | RoyK | variables and so on |
| 16:09.12 | RoyK | it's doable with SIP, but AFAIK the argument has been going on for a couple of years, at least, of how to implement that in iax2 |
| 16:09.16 | JunK-Y | i just know tony wrote something related to that a long time ago |
| 16:09.35 | RoyK | yes, i know |
| 16:09.47 | RoyK | still, i don't think it'll go into 1.4 |
| 16:09.52 | RoyK | it should have gone into 1.2 |
| 16:11.47 | EmleyMoor | Does IAX2 have friends or do you need both user and peer definitions for a softphone? |
| 16:12.38 | JunK-Y | RoyK: do u agree having IAX2 hardware will be a good thing? |
| 16:15.17 | dlynes_laptop | EmleyMoor: you can use friend for iax2 as well, yes |
| 16:15.39 | EmleyMoor | OK - so basically it's similar - OK |
| 16:16.28 | RoyK | JunK-Y: indeed. if someone was to make iax2 as admin-friendly as sip. I need those custom fields.... |
| 16:16.47 | RoyK | how does iax2 do it with regards to jitterbuffering/timestamps? |
| 16:16.58 | RoyK | are timestamps rewritten on bridging? |
| 16:17.02 | RoyK | iax/iax? |
| 16:17.31 | *** join/#asterisk Br0kenKeychain (n=thestor@CPE-24-94-241-227.wi.res.rr.com) |
| 16:17.47 | JunK-Y | i dont know |
| 16:17.50 | Br0kenKeychain | is anyone running asterisk on freebsd? |
| 16:19.14 | *** join/#asterisk luckasrg (n=lucasrg@200.124.168.244) |
| 16:19.33 | luckasrg | hi all |
| 16:20.01 | luckasrg | has any one have experience installing h323 on a trixbox system |
| 16:20.07 | RoyK | Br0kenKeychain: yes, lots |
| 16:20.24 | luckasrg | what will be the best way to get this done |
| 16:20.42 | luckasrg | or where can i get documentation |
| 16:21.06 | RoyK | luckasrg: quite possibly you would rather install asterisk from scratch on your favourite distro |
| 16:21.26 | Br0kenKeychain | RoyK, when I do /usr/local/etc/rc.d/zaptel.sh start it says zaptel.ko was already loaded, but i can't kldunload it because of an EBUSY, is there a workaround? |
| 16:21.29 | *** join/#asterisk [shodan] (n=shodan@ip103.96-113-216.pppoe1.joliette.intermonde.net) |
| 16:21.36 | luckasrg | because i have tryed the ooh323 but it has not worked for me |
| 16:22.05 | RoyK | Br0kenKeychain: no idea. sorry. i'm just using linux |
| 16:22.12 | RoyK | ~h323 |
| 16:22.18 | jbot | it has been said that h323 is An ITU-T standard for packet-based multimedia communications systems. This standard defines the different multimedia entities that make up a multimedia system - Endpoint, Gateway, Multipoint Conferencing Unit (MCU), and Gatekeeper - and their interaction. This standard is used for many voice-over-IP applications, and is heavily ... |
| 16:22.18 | Br0kenKeychain | ah, k |
| 16:22.19 | RoyK | ~h.323 |
| 16:22.20 | jbot | i guess h.323 is An ITU-T standard for packet-based multimedia communications systems. This standard defines the different multimedia entities that make up a multimedia system - Endpoint, Gateway, Multipoint Conferencing Unit (MCU), and Gatekeeper - and their interaction. This standard is used for many voice-over-IP applications, and is heavily dependent on other ... |
| 16:22.20 | luckasrg | and what will be a good administration gui |
| 16:22.59 | RoyK | luckasrg: admin gui? vim! |
| 16:23.10 | RoyK | s/guy/tui/ |
| 16:23.11 | RoyK | :P |
| 16:23.53 | EmleyMoor | Wow, got that working! |
| 16:24.07 | *** join/#asterisk doolph (n=doolph@200.46.148.58) |
| 16:24.45 | doolph | there's anything that can cause I get unauthorized while all my password is fine transmiting from nat? |
| 16:25.20 | doolph | i am having this issue |
| 16:25.37 | RoyK | doolph: pastebin full debug output and config files and ask again |
| 16:25.46 | luckasrg | and also if i am behind a nat how can i get h323 to send the outside address instead of the private one? |
| 16:26.05 | RoyK | h323 + nat == hell on earth |
| 16:26.08 | RoyK | don't think about it |
| 16:26.13 | RoyK | use something else |
| 16:26.14 | RoyK | not h.323 |
| 16:26.18 | RoyK | perhaps sip |
| 16:26.31 | RoyK | or even better, iax2, since that can be used with nat |
| 16:26.45 | RoyK | h.323 is NOT built for NATed networks |
| 16:26.54 | RoyK | trust me on that one |
| 16:28.50 | doolph | RoyK http://pastebin.ca/150176 |
| 16:32.06 | EmleyMoor | It seems I can't use my FWD account - it's like it doesn't want me to log in over iax2, despite the fact I've enabled it for that |
| 16:35.23 | EmleyMoor | Hmmm... - it started working Tommy Cooper fashion |
| 16:36.02 | luckasrg | yes i have faced the problem on other platforms |
| 16:37.14 | RoyK | doolph: the usual sequence is 'client sends REGISTER, server responds 401 Unauthorized with a nonce, client sends new REGISTER with password encrypted with the nonce, server answers 200 OK |
| 16:37.27 | RoyK | doolph: SIP does not support cleartext auth |
| 16:37.39 | doolph | i am using eyeBeam |
| 16:37.46 | doolph | what's wrong |
| 16:37.51 | RoyK | doolph: the usual sequence is 'client sends REGISTER, server responds 401 Unauthorized with a nonce, client sends new REGISTER with password encrypted with the nonce, server answers 200 OK |
| 16:37.52 | RoyK | doolph: SIP does not support cleartext auth |
| 16:38.02 | doolph | ok, there's any quick fix? |
| 16:38.14 | RoyK | quite possible you have not the whole debug |
| 16:38.21 | RoyK | i'm using eyebeam at work, and it works |
| 16:38.30 | RoyK | the problem lies in the config somewhere |
| 16:38.39 | doolph | can it be my router? |
| 16:38.44 | RoyK | doubt it |
| 16:38.50 | RoyK | seriously and utterly doubt it |
| 16:39.10 | doolph | mmm, I am trying with a Cisco ata 186 and I get the same problem |
| 16:40.14 | RoyK | bingo |
| 16:40.22 | RoyK | perhaps the asterisk setup is fscked? |
| 16:40.55 | doolph | yes maybe |
| 16:41.24 | doolph | but without nat it works perfectly |
| 16:41.31 | RoyK | pastebin all config, please |
| 16:41.34 | doolph | my asterisk is not natted, my extension does |
| 16:41.35 | RoyK | sip.conf for a start |
| 16:42.04 | doolph | if I try to connect without nat it works perfectly |
| 16:42.17 | eKo1 | nat is evil |
| 16:42.49 | doolph | but before was working |
| 16:43.00 | EmleyMoor | When I try to get a call from FWD's callme, asterisk reports: |
| 16:43.00 | benjk_ | yeah, I met Nat once, he's a nasty bugger |
| 16:43.20 | EmleyMoor | <PROTECTED> |
| 16:43.37 | EmleyMoor | What precisely does that mean and how do I fix it? |
| 16:49.50 | *** join/#asterisk kannan (n=trroot@125.22.66.211) |
| 16:50.38 | *** join/#asterisk thx2000 (i=AgentFLY@adsl-66-51-192-221.dslextreme.com) |
| 16:51.44 | RoyK | ~seen key2 |
| 16:52.07 | jbot | key2 is currently on #asterisk (2d 15h 27m 55s). Has said a total of 7 messages. Is idling for 2d 1h 41m 43s, last said: 'Ahrimanes: realplayer would play raw h263 ?'. |
| 16:52.07 | thx2000 | Can anyone recommend a provider that will let me manipulate my CIDName? |
| 16:52.07 | *** join/#asterisk hacked`` (n=lol@modemcable226.130-37-24.mc.videotron.ca) |
| 16:53.12 | doolph | omg |
| 16:53.13 | doolph | wtf |
| 16:53.19 | doolph | I have the right passwordddd |
| 16:53.41 | hacked`` | guys |
| 16:53.48 | hacked`` | lets say i want to accept 4 incoming calls at the same time |
| 16:53.53 | hacked`` | whats a good provider to go with |
| 16:53.59 | hacked`` | that isnt expensive |
| 16:54.02 | hacked`` | and reliable |
| 16:54.03 | RoyK | hacked``: lots |
| 16:54.10 | hacked`` | like whats the best one |
| 16:54.15 | hacked`` | i know lots are good, but which is best |
| 16:54.44 | RoyK | what is best? toyota or mazda? or volvo? |
| 16:54.51 | RoyK | depends what you're going to do |
| 16:55.24 | EmleyMoor | Why am I getting an error showing in asterisk about formats when I try to receive a call over FWD? |
| 16:55.40 | RoyK | hacked``: http://www.voip-info.org/wiki/view/DID+Service+Providers |
| 16:56.52 | hacked`` | well this is for business use |
| 16:56.56 | hacked`` | i want something cheap, with 4 incomings |
| 16:57.52 | RoyK | hacked``: google for it. or look through voip-info. there are lots there |
| 16:58.00 | hacked`` | i know |
| 16:58.04 | hacked`` | but im looking for user opinions |
| 16:58.08 | hacked`` | not 1 guy's opinion |
| 16:58.12 | hacked`` | see what im getting at |
| 16:58.15 | hacked`` | is teliax any good |
| 16:58.25 | RoyK | then email the asterisk-biz list and ask there |
| 16:58.51 | hacked`` | why cant i just ask here |
| 16:59.10 | hacked`` | its like you're doing everything you can to forward the question elsewher |
| 17:00.57 | RoyK | well. this is something like a technical channel. also, i would beleive you'd be getting FAR better answers if asking somewhere else. And again, you might want to specify where on the planet you're planning to use this, since that perhaps might help giving you a good answer |
| 17:01.29 | marta | someone can help me? I have this problem "make: *** No rule to make target `menuselect/menuselect.c', needed by `menuselect/menuselect'. Stop." what I can do? |
| 17:02.21 | thx2000 | Does anyone know of a provider that will let me manipulate my CIDName? |
| 17:02.35 | *** join/#asterisk SomethingISODD (n=Dan@h188.43.63.69.cable.ottr.cablerocket.net) |
| 17:03.15 | SomethingISODD | hello all question can * handle conference calls even if i don`t have a digium device, what i need to do is have a conference call but not take them in to a conference channel |
| 17:03.24 | Qwell | thx2000: cidname is looked up on the remote end. There are only a few circumstances where it will actually work - and it isn't up to your provider |
| 17:03.43 | thx2000 | thats what i was afraid of |
| 17:04.40 | thx2000 | is there any way to get your number registered w/ the local telco? |
| 17:05.31 | RoyK | thx2000: terminating on pstn or sip? |
| 17:05.39 | Qwell | SomethingISODD: yes, look into ztdummy |
| 17:05.51 | Qwell | marta: Is this svn trunk? |
| 17:05.52 | SomethingISODD | Qwell: thanks |
| 17:05.59 | marta | Qwell: yes |
| 17:06.14 | SomethingISODD | Qwell: i guess the right thing i would be wanting to do is three way calling |
| 17:06.15 | Qwell | marta: Did it get the menuselect/ directory like it was supposed to? |
| 17:06.25 | thx2000 | pstn...most of my calls will be placed to sbc pots lines...so it'd be nice if they knew who was callin |
| 17:06.42 | marta | Qwell: I don't have that directory |
| 17:06.55 | Qwell | marta: svn up |
| 17:07.00 | Qwell | tell me if you get any errors |
| 17:07.04 | doolph | RoyK I tried to connect to another Box with almost the same configuration and I can get connected :( |
| 17:07.13 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 17:07.29 | doolph | it seems that my server has problem with authentications over nat |
| 17:07.37 | RoyK | doolph: then the problem is quite possibly related to the weather |
| 17:07.52 | RoyK | s/grir/gri/ |
| 17:08.11 | doolph | ? |
| 17:09.15 | *** join/#asterisk jtodd (n=jtodd@ti.fox-den.com) |
| 17:09.20 | doolph | i am going to recompile the asterisk |
| 17:10.04 | thx2000 | RoyK: so i guess that is not possible? |
| 17:10.27 | EmleyMoor | Anybody able to take a call over FWD? |
| 17:10.58 | dhanes | hello |
| 17:11.01 | dhanes | Unable to add /var/lib/asterisk/sounds/custom/test.wav - Can not read file! |
| 17:11.06 | RoyK | thx2000: cidname is not transmitted over pstn |
| 17:11.12 | dhanes | and *77 is 'not found' |
| 17:11.42 | RoyK | ~rtfm |
| 17:11.45 | jbot | i heard rtfm is Read The F*cking Manual (TM). It is a suggestion to do your homework before posting a question. Sometimes used as RTFM $SPECIFIC_MANUAL to refer to a specific source of information. See also http://uncyclopedia.org/wiki/RTFM. |
| 17:11.45 | RoyK | ~docs |
| 17:11.46 | jbot | somebody said docs was Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 17:11.46 | RoyK | ~book |
| 17:11.48 | jbot | book is, like, a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 17:11.48 | thx2000 | RoyK: Yea, but i imagine SBC maintains a database of #'s and their corresponding names. Is there a way to get into that database? |
| 17:11.48 | dhanes | gsm is allowed |
| 17:11.54 | doolph | how do I uninstal zaptel |
| 17:12.00 | doolph | i am not going to use the digium card anymore |
| 17:12.07 | RoyK | thx2000: you'll need to contact the telco and ask. sorry. |
| 17:12.19 | thx2000 | thx |
| 17:12.35 | RoyK | doolph: forget it. it won't hurt if it's there. |
| 17:12.50 | RoyK | it's not a bloody windoze system where everything alian fucks up your box |
| 17:12.56 | dhanes | TFOT i have, both print and pdf |
| 17:14.16 | *** join/#asterisk The_LightSide (n=lightsid@wbs-196-2-125-235.wbs.co.za) |
| 17:14.22 | The_LightSide | isk-dev |
| 17:14.28 | The_LightSide | :/ |
| 17:14.32 | RoyK | isk isk |
| 17:14.33 | marta | Qwell: with a new checkout it's all ok, tnx. I think that something gone wrong with svk |
| 17:14.43 | Qwell | marta: yeah, menuselect is an svn:external |
| 17:14.50 | *** join/#asterisk zoa (n=d@pirus.securax.be) |
| 17:15.28 | marta | ok, thanks I will try to solve |
| 17:15.39 | The_LightSide | tzafrir, u around? |
| 17:15.52 | doolph | does asterisk-addons supports h323 or do i need to recompile something else |
| 17:17.15 | zoa | yoho |
| 17:17.25 | Qwell | zoa: hey |
| 17:17.33 | *** join/#asterisk negativecreep (n=xaeem@210.2.151.110) |
| 17:17.50 | *** join/#asterisk asterisk88 (n=malik@124.29.198.13) |
| 17:17.54 | asterisk88 | Hi guys |
| 17:17.59 | doolph | omg this is crap |
| 17:18.03 | doolph | lol |
| 17:18.09 | asterisk88 | im still having problems with CDR variables |
| 17:18.23 | doolph | what problems |
| 17:19.48 | asterisk88 | the CDR(answer) is not returning the values |
| 17:20.08 | luckasrg | roy have you tried h323 on trixbox |
| 17:20.35 | asterisk88 | i basically need the CDR(billsec) to work .. im using asterisk-1.2.10 |
| 17:20.53 | SomethingISODD | umm is there anyway to use ztdummy with out usb-uhci |
| 17:21.21 | asterisk88 | im using realtime sip & extensions |
| 17:21.56 | pyrom | Any easy way of doing a "WakeUp" Call via asterisk? e.g specify time and cancel it when the user picks up the extension? |
| 17:22.05 | SomethingISODD | anyone plz? |
| 17:22.21 | asterisk88 | im using AGI and CDR function is not returning the values |
| 17:22.28 | asterisk88 | i only need the billsec variable to work |
| 17:22.45 | asterisk88 | can anybody plese help me with this problem? if someone knows the correct syntax for AGI |
| 17:23.15 | areq | anyone use sangoma PRI ? |
| 17:23.31 | pyrom | Sorry for asking, google did the trick. |
| 17:23.35 | asterisk88 | i have tried AGI->get_variable(CDR(answer)); |
| 17:23.56 | asterisk88 | ive also tried AGI->get_variable(${CDR(answer)}); |
| 17:24.03 | asterisk88 | none is returning the values |
| 17:24.10 | SomethingISODD | anything please? |
| 17:25.21 | asterisk88 | help please :( |
| 17:30.44 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-71-241-191-122.buff.east.verizon.net) |
| 17:30.55 | SuPrSluG | hello |
| 17:30.59 | asterisk88 | hi |
| 17:31.42 | *** part/#asterisk negativecreep (n=xaeem@210.2.151.110) |
| 17:31.48 | SuPrSluG | <PROTECTED> |
| 17:33.37 | *** join/#asterisk ariel_ (n=Ariel@70-46-87-158.ftl.fdn.com) |
| 17:34.09 | *** join/#asterisk Aurs (n=Aurs@host-81-191-123-189.bluecom.no) |
| 17:34.27 | Aurs | hello. is there a max length of the IAX user/peer names? |
| 17:34.46 | Aurs | stopped working after a upgrade from 1.2.7.1 to 1.2.11 |
| 17:35.29 | asterisk88 | hey Aurs i'm also having a problem after the upgrade |
| 17:35.43 | asterisk88 | i upgraded to 1.2.10 |
| 17:35.48 | Aurs | ok |
| 17:35.55 | Aurs | the iax peers won't register anymore |
| 17:36.25 | asterisk88 | no idea about that.. i have no problems with sip registrations |
| 17:36.40 | Aurs | no, not me neither |
| 17:36.43 | Aurs | only the iax registrations |
| 17:36.53 | asterisk88 | i was having trouble with the CDR function.. the AGI->get_variable('CDR(answer)'); aint working for me |
| 17:37.20 | Aurs | isn't it ${CDR(answer)} |
| 17:37.24 | asterisk88 | may be there's an update about iax in some README file |
| 17:37.41 | asterisk88 | yeah that's correct if you're using it in extensions.conf.. |
| 17:37.49 | asterisk88 | im using realtime extensions |
| 17:38.03 | asterisk88 | db and agi combination |
| 17:38.09 | asterisk88 | do you know the syntax for that? |
| 17:38.53 | asterisk88 | AGI->get_variable(${CDR(answer)}); |
| 17:38.56 | asterisk88 | would this work? |
| 17:39.55 | *** join/#asterisk SpaceBass (n=sp@static-71-251-230-6.rcmdva.fios.verizon.net) |
| 17:40.43 | Aurs | that was what i meant yes |
| 17:40.47 | Aurs | but i dont know |
| 17:40.58 | *** join/#asterisk Avalone (n=Avalone_@dial-412.vl-cen-as2.avtlg.ru) |
| 17:41.26 | Aurs | i know that asterisk thinks it is an app and not a function when you dont use the ${} things |
| 17:41.36 | Aurs | (in extensions.conf) |
| 17:42.24 | Aurs | so CDR() would be considered an application, and ${CDR()} would be considered a function |
| 17:42.27 | Aurs | ... me thinks.. :P |
| 17:42.32 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 17:42.52 | asterisk88 | ive tried both |
| 17:43.06 | asterisk88 | and get errors on execution |
| 17:43.17 | asterisk88 | Undefined subroutine &main::CDR called |
| 17:44.32 | Aurs | don't know exactly what i am looking for in the changelog for this problem.. |
| 17:46.14 | asterisk88 | here's the result when i execute it as an app.. $AGI->exec("CDR billsec"); Could not find application (CDR(exec)) |
| 17:48.29 | *** join/#asterisk Nukemizer (n=Nuke@160.7.239.13) |
| 17:50.07 | *** join/#asterisk BugKham (i=CKGLOB@221.128.113.28) |
| 17:51.20 | BugKham | can we reset the value of DNID? |
| 17:54.31 | Aurs | hmm |
| 17:54.43 | Aurs | i found out my problem here asterisk88... kind of |
| 17:56.45 | *** join/#asterisk hmmhesays (n=hmmhesay@24-117-135-28.cpe.cableone.net) |
| 18:02.10 | *** join/#asterisk techie (n=gus@adsl-068-209-242-072.sip.mia.bellsouth.net) |
| 18:03.46 | Aurs | isn't it possible to get asterisk to accept iax registrations on both eth0 and eth0:1 ?! |
| 18:04.21 | Aurs | bindaddr=0.0.0.0 in iax.conf, but only the eth0:1 interface accepted iax registrations for some weird reason |
| 18:04.51 | doolph | anyone know why I keep getting unauthorized even my sip user/pass are correct ? (I am connecting from nat, server is not natted) |
| 18:05.28 | KDan | how do you play a background sound in AGI? |
| 18:05.54 | KDan | (ie the equivalent of Background() but in AGI) |
| 18:06.12 | asterisk88 | $AGI->stream_file('invalid'); |
| 18:06.43 | KDan | that doesn't seem to do it - i've got some wait_for_digit() commands afterwards but they don't record any digits |
| 18:07.16 | KDan | (if pressed while the file is still being streamed) |
| 18:08.29 | KDan | aha. escape_digits is the answer. |
| 18:08.39 | asterisk88 | cool |
| 18:09.00 | asterisk88 | Dan you know how to get the billsec (CDR) variable through AGI? |
| 18:09.16 | asterisk88 | ive tried several options.. various syntaxes :( |
| 18:09.21 | asterisk88 | nthing seems to be working |
| 18:09.34 | asterisk88 | $AGI->get_variable(CDR(billsec)); |
| 18:09.56 | asterisk88 | $AGI->get_variable(${CDR(bill)}); |
| 18:10.01 | asterisk88 | nothing's wroking |
| 18:10.19 | asterisk88 | its asterisk 1.2.10 |
| 18:10.29 | asterisk88 | sip users & exten both are realtime |
| 18:12.04 | *** part/#asterisk BugKham (i=CKGLOB@221.128.113.28) |
| 18:14.08 | The_LightSide | tzafrir ? |
| 18:18.47 | *** join/#asterisk vlt (n=daniel@dslb-088-073-199-006.pools.arcor-ip.net) |
| 18:19.23 | vlt | Hello. How can I Playback() a file when I dial a channel from CLI? |
| 18:20.50 | *** part/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com) |
| 18:22.32 | *** part/#asterisk techie (n=gus@adsl-068-209-242-072.sip.mia.bellsouth.net) |
| 18:27.01 | JunK-Y | vlt: show application dial , see options |
| 18:29.36 | JunK-Y | kdan: u want to play a file, while waiting for digitS? |
| 18:31.09 | *** join/#asterisk tmccrary (n=tmccrary@d14-69-160-83.try.wideopenwest.com) |
| 18:32.48 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 18:32.48 | *** mode/#asterisk [+o anthm] by ChanServ |
| 18:33.03 | *** join/#asterisk dieno (n=Dieno@124.29.194.150) |
| 18:33.06 | *** part/#asterisk SomethingISODD (n=Dan@h188.43.63.69.cable.ottr.cablerocket.net) |
| 18:33.24 | dieno | Asterisk Segmentation Fault wat does da stands for |
| 18:33.51 | JunK-Y | your're in trouble! |
| 18:33.56 | dieno | :D |
| 18:34.06 | JunK-Y | which version? |
| 18:34.06 | dieno | how can i get out of this trouble |
| 18:34.16 | dieno | 1.9 RPM |
| 18:34.30 | jbroome | ugh, rpm |
| 18:34.32 | JunK-Y | submit a report for that package. |
| 18:34.34 | Qwell | dieno: Don't use the rpm :) |
| 18:34.42 | docelmo | Get used to seg fault it happens fairly a good bit.. :) |
| 18:34.42 | JunK-Y | we dont support distro packages. |
| 18:34.43 | dieno | :'( |
| 18:34.46 | jbroome | use the src like |
| 18:34.47 | The_LightSide | theres a asterisk rpm?! |
| 18:34.47 | docelmo | EXCPECIALLY with rpm |
| 18:34.49 | jbroome | *luke |
| 18:34.54 | dieno | now how cani remove it |
| 18:34.54 | JunK-Y | hey docelmo ! |
| 18:34.59 | docelmo | whadup man |
| 18:35.08 | docelmo | rm -rf / |
| 18:35.20 | dieno | hmm |
| 18:35.21 | dieno | chekin |
| 18:35.23 | docelmo | dieno: rm -rd / |
| 18:35.29 | docelmo | err rm -rf / |
| 18:35.30 | docelmo | :) |
| 18:35.50 | JunK-Y | docelmo: nothing, same routine, gf is sick and school is starting soon, ya ? |
| 18:35.53 | *** join/#asterisk benjk_ (n=benjamin@f8a01-0357.din.or.jp) |
| 18:36.01 | docelmo | You should be able to do a rpm -qa | grep asterisk |
| 18:36.12 | docelmo | then rpm -e package_name |
| 18:36.37 | docelmo | Same.. Well my school hasnt stopped.. Im due to grad in feb or march next year.. |
| 18:36.41 | dieno | ok done |
| 18:36.54 | docelmo | ok done what? |
| 18:36.57 | docelmo | What did you type? |
| 18:37.00 | dieno | remove |
| 18:37.13 | dieno | rpm - e asterisk*..rpm |
| 18:37.36 | dieno | package asterisk.....rpm is not installed |
| 18:37.43 | Qwell | just rpm -e asterisk |
| 18:38.13 | dieno | ohh ok |
| 18:38.32 | vlt | JunK-Y: Thank you. I read the description of Dial() but still don't see how to Playback() a file to an established call from CLI. Any further hints? ;) |
| 18:38.54 | dieno | ok processed for few sec n return to root |
| 18:39.25 | dieno | now should i try 1.2.11 tar.gz |
| 18:39.44 | vlt | JunK-Y: I you meant option "A", how must I create the command? `dial 55513245@outgoing,,A(file)` doesn't work ... |
| 18:39.54 | docelmo | yes.. which fixed the fuck ups in 10 but I would watch for 12 it will be coming out to fix the fuckups in 11 |
| 18:40.07 | vlt | JunK-Y: *If you ... |
| 18:43.01 | JunK-Y | vlt: its for dialplan command dial |
| 18:43.34 | vlt | JunK-Y: And then how can I Playback from CLI? |
| 18:44.06 | JunK-Y | u dial an exten, that exten does the dial. |
| 18:44.38 | vlt | JunK-Y: Aah, that way ... |
| 18:46.01 | vlt | JunK-Y: Mmh ... I can't test it because my whole * server hangs after the last dial attempt from CLI ... :( |
| 18:47.11 | doolph | anyone know why I keep getting unauthorized even my sip user/pass are correct ? (I am connecting from nat, server is not natted) |
| 18:49.29 | vlt | doolph: Can you try "insecure=very" in sip.conf? |
| 18:50.10 | doolph | ill try |
| 18:51.17 | doolph | Transmitting (NAT) to 200.46.148.58:9204: |
| 18:51.17 | doolph | SIP/2.0 401 Unauthorized |
| 18:51.20 | doolph | same problem |
| 18:56.33 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 19:00.27 | hmmhesays | nat problems suck man |
| 19:01.02 | doolph | so is it a nat problem or autentication problem |
| 19:01.12 | doolph | I tried to connect to another server and it is ok |
| 19:01.29 | hmmhesays | do your credentials match? |
| 19:01.38 | doolph | absolutly |
| 19:01.52 | doolph | username=100 |
| 19:01.53 | hmmhesays | then its a nat problem isn't it |
| 19:01.54 | doolph | password=100a |
| 19:02.06 | doolph | its the asterisk server problem |
| 19:02.15 | doolph | because I tried to connect to another server and it was ok |
| 19:02.20 | doolph | so i don't know what it could be |
| 19:02.48 | doolph | its so weird |
| 19:03.26 | hmmhesays | try a different client |
| 19:03.43 | doolph | like which |
| 19:03.55 | hmmhesays | different phone |
| 19:04.17 | doolph | well I have a cisco ata 186 |
| 19:04.25 | doolph | if it is connected without nat works |
| 19:04.35 | doolph | behind nat cannot get fail |
| 19:05.18 | hmmhesays | do you have nat=yes ? |
| 19:05.27 | doolph | yes |
| 19:05.28 | *** join/#asterisk linlin (i=linlin@c-67-173-38-87.hsd1.il.comcast.net) |
| 19:07.09 | doolph | why i get authentication MD5 |
| 19:07.12 | doolph | in debug |
| 19:07.18 | *** join/#asterisk linlin (n=linlin@c-67-173-38-87.hsd1.il.comcast.net) |
| 19:09.43 | *** join/#asterisk SomethingISODD (n=Dan@h188.43.63.69.cable.ottr.cablerocket.net) |
| 19:09.54 | SomethingISODD | can asterisk handle three way calling? |
| 19:10.04 | doolph | WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0250d378" |
| 19:10.05 | doolph | Content-Length: 0 |
| 19:10.08 | SomethingISODD | through voip lines?? |
| 19:10.11 | JunK-Y | doolph: cause u use md5 for authentication? |
| 19:10.20 | JunK-Y | SomethingISODD: sure, * is powerful :) |
| 19:10.50 | doolph | JunK-Y my asterisk is using md5 for some reason |
| 19:11.09 | JunK-Y | in ur sip/iax.conf? |
| 19:11.54 | doolph | there's no md5 there |
| 19:12.52 | SomethingISODD | JunK-Y: do you know of any examples on doing that? |
| 19:13.43 | JunK-Y | SomethingISODD: take a look on www.voip-info.org |
| 19:13.47 | JunK-Y | theres so MUCH infos tehre |
| 19:14.04 | SomethingISODD | JunK-Y: ya i have been but all i find is doing it through your telco |
| 19:14.10 | SomethingISODD | nothing about doing it through asterisk it self |
| 19:15.14 | *** join/#asterisk rbordeaux (i=hidden-u@80.169.196.234) |
| 19:16.45 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 19:17.57 | JunK-Y | just do a meetme conf |
| 19:18.52 | hmmhesays | ugh I need some food |
| 19:18.55 | hmmhesays | fooooood |
| 19:19.46 | hmmhesays | cool, thanks |
| 19:20.07 | file | that'll be $3.95 |
| 19:20.15 | hmmhesays | bah |
| 19:20.28 | *** join/#asterisk Deeewayne (n=dwayne@ool-44c0d56e.dyn.optonline.net) |
| 19:20.34 | file | Deeewayne: IT'S YOU! |
| 19:20.54 | Deeewayne | hey file. how's the day treating you? |
| 19:21.06 | file | great, the house is empty and I am at one with the world |
| 19:21.07 | file | you? |
| 19:21.29 | Deeewayne | babies are asleep, so I am at one as well |
| 19:21.53 | Deeewayne | the sound of silence is deafening |
| 19:22.02 | *** join/#asterisk spr1te (i=spr1te@194.187.130.227) |
| 19:22.13 | spr1te | hi.. speak rus? |
| 19:22.29 | Deeewayne | kak dela? |
| 19:22.32 | spr1te | =) |
| 19:22.45 | spr1te | ó ìåíÿ âîïðîñ íà ñ÷åò àñòåðèñêà è êîäåêà g729.. |
| 19:23.18 | file | Deeewayne: it's... party time |
| 19:23.34 | Aurs | doolph: host=dynamic in sip.conf on the sip account? |
| 19:24.08 | Qwell | Deeewayne: babies sleep? simultaneously? lucky you |
| 19:24.13 | spr1te | åñòü àñòåðèñê.. íà íåì ñòîèò êîäåê g729.. êîãäà 2 ñèï òåëåôîíà âêëþ÷åíû â 1 ëèíêñèñ - ìåæäó íèìè ñâÿçü ëèøü îäíîñòîðîíÿÿ (1 ñèï íà âòîðîé çâîíèò, à âòîðîé íà ïåðâûé íåò).. + îòñóòñòâóåò âòîðîé êàíàë (òîåñòü êîãäà ñ êåì-òî ãîâîðèøü, íåëüçÿ íàæàòü ôëåø è íàáðàòü |
| 19:24.13 | spr1te | äðóãîé íîìåð).. ëå÷èòñÿ çàìåíîé â ñèï_àääèøèíàë g729 íà alaw |
| 19:24.18 | spr1te | åñòü ìûñëè ïî ýòîìó ïîâîäó? |
| 19:24.27 | Qwell | spr1te: english? |
| 19:24.33 | Qwell | even broken english is fine |
| 19:24.47 | Aurs | hehe |
| 19:24.48 | Deeewayne | well one is asleep, the other is lying in bed watching barney and eating raisinets - she'll be asleep in 10 minutes |
| 19:24.49 | spr1te | Qwell, i very bad speak english =) |
| 19:24.54 | Qwell | spr1te: That is fine |
| 19:25.06 | Qwell | Your English is better than my Russian |
| 19:25.40 | Qwell | If all else fails - we call in the moginator |
| 19:25.49 | Aurs | what's a moginator |
| 19:25.56 | Qwell | a russian speaking mog |
| 19:26.35 | Aurs | is it impossible to get * to listen for iax registrations on eth0 and eth0:1 ? |
| 19:26.53 | Qwell | just bind to 0.0.0.0 |
| 19:26.57 | Aurs | did that |
| 19:27.04 | spr1te | i have asterisk, codec g729,.. when 2 sip phones from 1 lynksys call between itself (first sip call to second sip - OK, but second sip call to first - broken), and when codec g729 isset - second line (at one sip) - dont work... when i set codec alaw - all is ok... do you have some ideas? |
| 19:27.05 | Qwell | check your firewall |
| 19:27.06 | Aurs | but only the eth0:1 address was working |
| 19:27.12 | Aurs | tried with no firewall |
| 19:27.40 | Qwell | spr1te: What linksys? |
| 19:27.42 | Aurs | (centos 4.3 and asterisk 1.2.11) |
| 19:28.01 | Qwell | spr1te: Some ATAs can only do 1 g729 call, even though it has 2 ports. |
| 19:28.18 | spr1te | box... in this box i insert 2 sip phones |
| 19:28.25 | Qwell | spr1te: not ATA? |
| 19:28.29 | Qwell | What phones? |
| 19:28.35 | spr1te | sip phones |
| 19:28.40 | Qwell | which sip phones? :) |
| 19:28.42 | Qwell | brand |
| 19:28.52 | Aurs | linksys? |
| 19:29.01 | Qwell | brand and model |
| 19:29.12 | *** join/#asterisk tsurk0 (n=tsurko@85.187.160.157) |
| 19:29.14 | spr1te | linksys=cisco |
| 19:29.17 | spr1te | 1 min |
| 19:29.21 | spr1te | i found at google=) |
| 19:29.36 | Aurs | Qwell: can it be that eth0 and eth0:1 has to be on the same subnet or something? |
| 19:29.45 | Aurs | because they weren't |
| 19:29.52 | Qwell | Aurs: Do you have a deny? |
| 19:30.02 | jets | aurs:you can have two different subnets on subinterfaces. |
| 19:30.18 | Qwell | permit/deny, rather |
| 19:30.44 | Aurs | jets: so you could have 192.168.0.1, 255.255.255.0 on eth0, and 10.0.3.1, 255.0.0.0 on eth0:1 ? |
| 19:30.50 | Qwell | absolutely |
| 19:31.00 | Aurs | ok |
| 19:31.06 | spr1te | http://www.linksys.com/servlet/Satellite?c=L_Product_C2&childpagename=US%2FLayout&cid=1117044308483&packedargs=site%3DUS&pagename=Linksys%2FCommon%2FVisitorWrapper |
| 19:31.07 | Qwell | Just make sure not to deny 0.0.0.0 and only permit one of those, or something |
| 19:31.11 | spr1te | sorry for long link |
| 19:31.15 | spr1te | this is my linksys |
| 19:31.17 | jets | aurs: sure can |
| 19:31.22 | Aurs | Qwell: you mean in iax.conf? |
| 19:31.24 | Qwell | okay, so they are analog phones |
| 19:31.25 | Qwell | Aurs: yes |
| 19:31.27 | jets | ifconfig eth0 192.168.0.1 netmask 255.255.255.0 up |
| 19:31.35 | jets | ifconfig eth0:1 10.0.3.1 netmask 255.0.0.0 up |
| 19:31.40 | Aurs | i could ping both ip's |
| 19:31.41 | Qwell | spr1te: Yes, what I said above, about only 1 g729 channel is true, with that ATA, I believe |
| 19:31.49 | Aurs | så I guess the network part was ok |
| 19:31.50 | Qwell | Aurs: check for deny/permit in iax.conf :p |
| 19:31.50 | spr1te | hmm |
| 19:31.52 | spr1te | thanks |
| 19:31.56 | Aurs | Qwell: will do |
| 19:31.59 | jets | Aurs: your upstream router may confuse this if they are public IP's but that's a whole different scenario |
| 19:32.02 | Qwell | spr1te: Do you understand what I mean? |
| 19:32.09 | spr1te | yes =) ... |
| 19:32.14 | Aurs | jets: they are public ips |
| 19:32.14 | Qwell | good :) |
| 19:32.35 | Aurs | but as i said, I could ping both ip's |
| 19:32.40 | spr1te | i should or change this davice for another, or set codec alaw =) |
| 19:32.48 | jets | Right, could the 'world' ping both? |
| 19:32.49 | spr1te | what device you recomended for me? =) |
| 19:32.56 | Aurs | i was in the world |
| 19:33.00 | jets | heh |
| 19:33.00 | jets | k |
| 19:33.01 | Aurs | the server is on a remote site |
| 19:33.02 | Aurs | hehe |
| 19:33.26 | jets | oh so whats the matter? i think i jumped in to this conversation in the middle of it. |
| 19:33.42 | Aurs | nothing fancy in my iax.conf, Qwell |
| 19:33.57 | Aurs | jets: i couldn't get iax clients to register on the ip of eth0 |
| 19:34.03 | Aurs | but they worked on the ip of eth0:1 |
| 19:34.13 | Aurs | when I removed eth0:1, I could register on eth0 |
| 19:34.22 | *** join/#asterisk _deg_ (n=deg@201-40-223-25.ctame704.dsl.brasiltelecom.net.br) |
| 19:34.37 | jets | can you telnet to a remote port on eth0? like port 22 or something (i'm seeing if a full tcp syn/ack connection can happen) |
| 19:34.47 | jets | awww |
| 19:34.51 | jets | nevermind you answered that question |
| 19:34.54 | jets | its something in software then |
| 19:34.55 | Aurs | ssh to eth0 worked |
| 19:35.08 | Aurs | and also sip registrations to eth0 |
| 19:35.17 | Aurs | but in sip.conf there is a bindaddr=ip-of-eth0 |
| 19:35.19 | spr1te | Qwell, do you now alike device.. which supports g729 (a lot of channels) |
| 19:35.19 | spr1te | <PROTECTED> |
| 19:35.30 | Qwell | spr1te: I do not know. |
| 19:35.45 | spr1te | ok.. thanks for help.. |
| 19:35.48 | Aurs | could it be that I have found a bug here? |
| 19:35.54 | Aurs | that would be cool |
| 19:36.18 | *** join/#asterisk jhiver (n=jhiver@LReunion-151-2-164.w193-253.abo.wanadoo.fr) |
| 19:36.21 | jhiver | hey guys |
| 19:36.29 | jhiver | there is something i can't figure out in the docs |
| 19:36.46 | *** join/#asterisk wulfy814 (n=wulfy814@c-67-165-37-20.hsd1.pa.comcast.net) |
| 19:37.04 | jhiver | when you set priorityjumping=no, does it jump to + 1 or does it stop processing alltogether if a channel is, say, busy? |
| 19:40.20 | Aurs | jhiver: it jumps +101 if it is set to yes |
| 19:40.22 | Aurs | me thinks |
| 19:40.26 | jhiver | and also, wouldn't it be possible to do a failover using 'n' rather than priority numbers? |
| 19:40.37 | jhiver | something like |
| 19:40.52 | jhiver | exten => 1,_X.,Dial(SIP/foo) |
| 19:41.07 | jhiver | exten => _X.,n,Dial(SIP/bar) |
| 19:41.40 | jhiver | because you know, all this +101 malarki is handy but it isn't feel very clean |
| 19:42.11 | Qwell | yeah, priority jumping is a bit silly |
| 19:42.36 | jhiver | Qwell, so do you know if it's possible to fail over without it? |
| 19:42.41 | Qwell | sure |
| 19:42.57 | jhiver | how do you do it? |
| 19:43.09 | Qwell | check ${DIALSTATUS} |
| 19:43.20 | jhiver | ok so I can do |
| 19:43.29 | Qwell | see macro-stdexten for examples |
| 19:43.30 | jhiver | exten => _X.,1,Dial(SIP/foo) |
| 19:44.10 | jhiver | exten => _X,n,GotoIf(${DIALSTATUS}=ANSWERED?10000:n) ? |
| 19:44.27 | jhiver | and then in 10000 i could just NoOp() or something |
| 19:44.29 | Qwell | jhiver: look at the stdexten macro... |
| 19:44.36 | Qwell | No need for numbered priorities |
| 19:44.37 | jhiver | ok |
| 19:45.32 | jhiver | http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+macro this one? |
| 19:45.41 | Qwell | jhiver: it's in the extensions.conf.sample |
| 19:45.48 | jhiver | ah ok |
| 19:47.12 | jhiver | thx |
| 19:47.14 | *** join/#asterisk roving_prole (n=Harper@c-71-199-16-110.hsd1.co.comcast.net) |
| 19:47.31 | jhiver | i have the bad habit of nuking all conf files before configuring asterisk, i find them too overwhelming |
| 19:47.55 | Qwell | jhiver: not really a bad idea |
| 19:49.35 | jhiver | ok i see |
| 19:49.49 | jhiver | can Macros recursively call themselves? |
| 19:49.57 | Qwell | yes, a certain number of levels |
| 19:50.04 | Qwell | well...maybe not themselves |
| 19:50.16 | jhiver | i might pass 9 or so arguments max, then if it fails shift the arguments and get the macro to recall itself |
| 19:50.57 | jhiver | the thing is I have written a LCR fastagi which defines ${LCR_DIAL_0}, ${LCR_DIAL_1}, etc |
| 19:51.09 | jhiver | and i'm looking for the cleanest way of implementing this in the dialplan |
| 19:51.23 | Qwell | jhiver: you probably want to look at the ARRAY function |
| 19:51.31 | jhiver | aah |
| 19:51.42 | jhiver | there's something I haven't heard of yet :) |
| 19:51.47 | Qwell | it's in trunk, but... |
| 19:51.57 | Qwell | you can get it for 1.2, from svncommunity.digium.com/ |
| 19:52.06 | Qwell | I think anyhow |
| 19:52.14 | jhiver | what does it do? |
| 19:52.22 | jhiver | Synopsis |
| 19:52.22 | jhiver | <PROTECTED> |
| 19:52.24 | jhiver | ok :) |
| 19:53.03 | Qwell | http://svncommunity.digium.com/view/func_odbc/1.2/ |
| 19:53.06 | Qwell | That comes with func_array |
| 19:56.16 | *** join/#asterisk AJaymn (n=boiwonde@71-36-133-35.mpls.qwest.net) |
| 19:57.36 | *** join/#asterisk Doctor (n=mage@S01060007950ade7b.vc.shawcable.net) |
| 20:00.20 | Doctor | how tricky would it be to use asterisk + an FXO card as a replacement for voicemail that dumps out mp3s? |
| 20:05.18 | *** join/#asterisk novafirst (n=kosta@wrt1.niclab.com) |
| 20:05.18 | Qwell | Doctor: not at all |
| 20:05.58 | novafirst | where can I get one mp3 for MusicOnHold ? |
| 20:06.07 | *** join/#asterisk anthonyl (n=rmerz@c-24-12-179-217.hsd1.il.comcast.net) |
| 20:06.14 | jhiver | check out 'freeplay' |
| 20:06.36 | novafirst | ? website? |
| 20:06.38 | jhiver | or use fp-calmriver :) |
| 20:06.41 | jhiver | yeah |
| 20:06.48 | jhiver | ~google freeplay |
| 20:07.10 | jhiver | jbot is dead tonight? |
| 20:07.36 | Aurs | but google isn't |
| 20:07.50 | jhiver | which is fortunate :) |
| 20:07.52 | Aurs | hehe |
| 20:08.05 | Doctor | I think half the world would stop without google |
| 20:08.07 | benjk_ | do you play the music only to yourself or to callers which are not part of your household? |
| 20:08.08 | jhiver | it's getting seriously scary as well |
| 20:08.21 | jhiver | I couldn't do a thing in a day without this stupid website |
| 20:08.23 | jhiver | it's mad |
| 20:08.27 | jhiver | pretty mad |
| 20:08.43 | anthonyl | well, you can start using yahoo ;) |
| 20:08.49 | jhiver | how did we manage to live before? :) |
| 20:08.54 | Aurs | and funny things happen because of google |
| 20:09.06 | jhiver | like? |
| 20:09.10 | Aurs | I got a call from someone asking for the CEO |
| 20:09.27 | Aurs | and he said a name that was known to me, but he is not the CEO of my company |
| 20:09.28 | hmmhesays | heh nice |
| 20:09.34 | Aurs | he is the CEO of one of our customers |
| 20:09.35 | jhiver | ah |
| 20:09.37 | hmmhesays | your average moron doesn't know how to use google |
| 20:09.42 | jhiver | funny |
| 20:09.43 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) [NETSPLIT VICTIM] |
| 20:09.47 | Aurs | and this customer has a "testemonial" statement on our website |
| 20:10.01 | Aurs | so i searched google for "mycompany +CEO" |
| 20:10.23 | Aurs | then i understood why he asked for that guy |
| 20:10.25 | Aurs | hehe |
| 20:11.05 | Aurs | and I also read a blog about a guy that has created some kind of order system. which of course was indexed by google |
| 20:11.14 | Aurs | where is that story.. that was kinda cool |
| 20:11.40 | jhiver | yeah, where is it? |
| 20:11.43 | jhiver | go google it :) |
| 20:11.53 | Aurs | trying .P |
| 20:12.47 | benjk_ | Google officially commented on the fact that "to google" has become a verb in the Oxford dictionary |
| 20:13.10 | jhiver | yeah, google is becoming like a word now |
| 20:13.18 | jhiver | i mean, you can google msn right ? :) |
| 20:13.33 | benjk_ | they said they disagreed, that people should use the phrase "to conduct a Google search" |
| 20:13.34 | jhiver | (not that it would seem a very useful thing to do...) |
| 20:13.43 | jhiver | lol |
| 20:13.46 | bkw__ | toobad for google |
| 20:14.08 | bkw__ | ho ho ho |
| 20:14.10 | Qwell | I verbified your noun, y0 |
| 20:14.28 | Qwell | (yes, I *DID* just use "to verb") |
| 20:14.36 | benjk_ | they are unhappy about the trademark dilution of the fact that a verb "to google" brings with it |
| 20:14.39 | jhiver | lol |
| 20:14.55 | jhiver | yeah well that's what happens when you're victim of your success |
| 20:15.16 | jhiver | i don't hear coca cola complaining that a "coke" can be used for all kinds of sodas |
| 20:15.33 | benjk_ | its silly really |
| 20:15.49 | benjk_ | but that's software for you |
| 20:15.52 | novafirst | what specs for mp3 MusicOnHold ?? and which tool I can use to change mp3's sample rate |
| 20:16.15 | benjk_ | software companies and their executives are mentally underage |
| 20:16.16 | Qwell | novafirst: sox |
| 20:16.28 | benjk_ | mostly |
| 20:16.32 | jhiver | hey maybe we should use the verb 'asterisk' to designate interconnecting voip systems... |
| 20:16.51 | jhiver | "I asterisked these guys yesterday... they were doing H323 only that was a pain in the ass" |
| 20:16.59 | jhiver | :) |
| 20:17.00 | Qwell | jhiver: I don't see that happening |
| 20:17.04 | benjk_ | unlikely to happen |
| 20:17.12 | Qwell | "call" is already a proper verb for that |
| 20:17.28 | Qwell | with google, there was no "proper" single verb for "search the web" |
| 20:17.36 | jhiver | nono, i say "interconnecting voip systems", not calling somebody ;) |
| 20:17.43 | Qwell | it's still a "call" :p |
| 20:17.49 | benjk_ | yeah, interconnect is the word |
| 20:17.53 | jhiver | and you think there is a proper verb for "interconnecting voip systems" ? :) |
| 20:18.02 | Qwell | I'm gonna have to agree with benjk_ |
| 20:18.06 | Qwell | ...strangely |
| 20:18.10 | benjk_ | and there's slang too, like "hookup" |
| 20:18.14 | jhiver | aaah guys this is just so un-fun :) |
| 20:18.53 | benjk_ | besides asterisk is too long a word for that sort of thing |
| 20:19.19 | benjk_ | people use skype as a verb |
| 20:19.20 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 20:19.23 | benjk_ | its short |
| 20:19.32 | Qwell | ugh |
| 20:19.42 | Qwell | I propose a new verb |
| 20:19.42 | novafirst | Qwell: do you know of any website with examples? |
| 20:19.44 | Qwell | Nortel |
| 20:19.53 | benjk_ | heh |
| 20:19.58 | Qwell | benjk_: I'm gonna nortel you, pick up your phone |
| 20:20.09 | bkw__ | Digium really will have a hard time with copyright on the word "Asterisk" because they don't enforce it |
| 20:20.21 | benjk_ | imagine Skype was called Luxemburg Internet Communications |
| 20:20.22 | jhiver | lol |
| 20:20.36 | jhiver | well |
| 20:20.43 | bkw__ | http://www.asterisknerds.com/ |
| 20:20.45 | jhiver | you could use an acronym |
| 20:20.46 | jhiver | like |
| 20:20.50 | bkw__ | http://asteriskmall.com/ |
| 20:20.52 | jhiver | I'm gonna LIC you |
| 20:20.53 | benjk_ | and they software was called LuxIntercom |
| 20:21.04 | benjk_ | nobody would say "I Luxintercom you" |
| 20:21.20 | jhiver | no but I LIC you would be kind of fun |
| 20:21.30 | bkw__ | broadvoice advertises on the keyword asterisk |
| 20:21.40 | jhiver | lol - lots of sillyness :) |
| 20:21.59 | jhiver | yeah well they're not competitors are they |
| 20:23.07 | Qwell | bkw__: trademark, not copyright. :) You knew that though |
| 20:23.16 | bkw__ | ya ya ment that |
| 20:24.12 | jhiver | okay, then maybe the verb "to asterisk" could mean "to fuck about with ip telephones" ? :) |
| 20:24.32 | jhiver | "i'm tired, i asterisked all night" :) |
| 20:24.33 | benjk_ | or to write extremely messy C |
| 20:24.39 | jhiver | kinda hard to say |
| 20:24.40 | jhiver | hahaha |
| 20:24.59 | jhiver | there's no way they can fireme, i've asterisked up my code pretty good |
| 20:25.29 | bkw__ | "to asterisk" - What does it mean? To totally do the wrong thing at the wrong time with out any reguard for the proper way to do something. |
| 20:25.37 | jhiver | that being said, seen from the outside, it looks pretty clean with all these nifty separate modules |
| 20:25.41 | bkw__ | "to asterisk" suck at everything? |
| 20:25.52 | jhiver | now that's a bit harsh |
| 20:26.04 | jhiver | if it was that bad not that many people would be using it :) |
| 20:26.13 | jhiver | what it really needs is IMHO |
| 20:26.18 | benjk_ | I think I heard that one before |
| 20:26.19 | jhiver | a) improvements on the H323 stack |
| 20:26.25 | jhiver | b) scalability improvements |
| 20:26.28 | jhiver | that's it :) |
| 20:26.35 | benjk_ | I think there was this company in er Seattle? |
| 20:26.36 | bkw__ | jhiver, ok why do people use windows? It sucks too. |
| 20:26.45 | bkw__ | benjk_, :P |
| 20:27.02 | jhiver | bkw__, maybe it doesn't suck that much then, considering people are PAYING to use it :) |
| 20:27.05 | UForgotten | question: how do I get it so it doesn't play a ring while waiting for a trunk dial to connect? |
| 20:27.05 | stephane_ | re |
| 20:27.13 | jhiver | not that i use windows of course :) |
| 20:27.22 | UForgotten | cos it rings 6 times before it's an actual trunk ringing, pissing me off |
| 20:27.31 | bkw__ | jhiver, well asterisk has some fatal flaws. But nobody seems to want to fix those |
| 20:27.38 | benjk_ | dont put the "r" in your Dial() statement |
| 20:27.39 | jhiver | bkw__, like what? |
| 20:27.40 | bkw__ | asterisk does great at small tasks |
| 20:27.54 | jhiver | UForgotten, avoid using Ringing() or the 'r' flag |
| 20:28.16 | UForgotten | I'll try that - thanks |
| 20:28.26 | jhiver | speaking of which, i've heard that yate scaled a bit better than * |
| 20:28.37 | jhiver | i've taken a look at it, the conf files are a mess and there is almost no docs :( |
| 20:28.41 | bkw__ | jhiver, the whole "fear of threads", The locking model its crazy. Some base things like the channel list need to be hash tables or something that doesn't lock the whole PBX up when you do a "show channels" on a loaded system. |
| 20:28.54 | benjk_ | andi if all else fails, maybe you wanna see a doctor to examine your ears, just in case its your ears which are ringing ;) |
| 20:29.10 | bkw__ | the biggest issue with asterisk is they have to reinvent the wheel everywhere because they have the need to own the code outright so they can lisense it outside the GPL for money. |
| 20:29.44 | bkw__ | their are excellent thread, hasing and memory abstraction layers but they can't use any of them. |
| 20:29.45 | benjk_ | that's still no reason not to use well established techniques |
| 20:30.25 | jhiver | bkw__, yeah but we don't know how big the code licensing is in terms of revenue |
| 20:30.37 | benjk_ | who cares |
| 20:30.40 | bkw__ | and the fact the console out put can affect audio quality on a very loaded system... marks answer to that is "Just don't do that" insteead of lets fix it. |
| 20:30.44 | jhiver | it could be that if they couldn't do that, they might have less people working on the software |
| 20:30.46 | *** join/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 20:30.51 | jhiver | which means less improvements for the rest of us |
| 20:31.06 | benjk_ | the fact that asterisk even has a console is already totally silly |
| 20:31.08 | bkw__ | well we accept the fact that people are going to rip the code off .. which is why we choose MPL |
| 20:31.26 | jhiver | btw how is that openpbx thing going? |
| 20:31.34 | bkw__ | I wouldn't know.. i'm not involved |
| 20:31.44 | jhiver | oh |
| 20:31.51 | benjk_ | a server daemon should never ever have an interactive CLI in the first place |
| 20:31.53 | jhiver | what's you involved with then? :) |
| 20:32.39 | jhiver | bkw__, how can it affect audio quality? what can happen? |
| 20:32.53 | jhiver | i've never head that one before |
| 20:33.36 | jhiver | might be why i'm having some issues lately... my system is pretty loaded... maybe 40-50 concurrent calls on an AMD Sempron :) |
| 20:33.49 | benjk_ | because the CLI eats up resources that the audio processing subsystem then starves for |
| 20:34.00 | benjk_ | CLI should never be in the daemon |
| 20:34.04 | jhiver | it's not doing any kind of transcoding or anything like that though |
| 20:34.09 | benjk_ | ought to be separate |
| 20:34.26 | jhiver | and peak load average is around 0.70, i think it's still sort of reasonable |
| 20:34.36 | jhiver | oh yeah, asterisk needs T.38 as well |
| 20:34.40 | jhiver | that would be nice to have |
| 20:34.45 | Qwell | It has T.38 passthrough |
| 20:34.47 | benjk_ | not going to happen |
| 20:34.53 | jhiver | improved H323 stack and T.38 |
| 20:34.58 | jhiver | please pretty please :) |
| 20:35.03 | Qwell | jhiver: somebody is working on h323 |
| 20:35.17 | jhiver | i use chan_h323 at the moment |
| 20:35.25 | bkw__ | jhiver, say you're connected to your asterisk box over say GPRS doing a reload |
| 20:35.26 | jhiver | it works sort of good but has some spooky bugs |
| 20:35.50 | bkw__ | all that printing to the screen will start to cause the audio quality of calls that are up to suffer. |
| 20:36.11 | benjk_ | if you have an average asterisk setup your system will have anywhere between 100 and 150 apps |
| 20:36.32 | jhiver | mine has far less |
| 20:36.47 | jhiver | <PROTECTED> |
| 20:36.50 | benjk_ | if you have 50 concurrent calls each of which run a macro that does on average about 10 app calls |
| 20:37.10 | benjk_ | then you are getting something in the order of 252.000 function calls to strcmp and strcasecmp |
| 20:37.28 | benjk_ | where 500 integer compares would suffice |
| 20:37.37 | jhiver | i understand |
| 20:37.48 | jhiver | so it could be optimized |
| 20:37.49 | benjk_ | everytime you run Dial() |
| 20:38.00 | jhiver | i mean, there is still lots of room for improvement at that level right |
| 20:38.11 | benjk_ | it will compare each of the characters D i a l to each character of all applications registered |
| 20:38.18 | jhiver | have you guys given a try to yate / freepbx / freeswitch? what are your thoughts? |
| 20:38.26 | benjk_ | there is no tokenisation and no hashing |
| 20:39.12 | benjk_ | input is not validated either |
| 20:39.22 | benjk_ | when you have something like [foobar] |
| 20:39.42 | benjk_ | it will just to a strchr for "[" and a strrchr for "]" |
| 20:39.51 | benjk_ | and take anything in between without any further checking |
| 20:39.58 | benjk_ | [[[[[[[[[[[[[[[[[[[[[[[[[[[[[[[] |
| 20:40.03 | benjk_ | is a valid context name |
| 20:40.27 | benjk_ | so is [$[foobar}] |
| 20:40.35 | benjk_ | try to do include => ${foobar} |
| 20:40.42 | benjk_ | will give you something else |
| 20:40.53 | jhiver | hehe |
| 20:40.58 | jhiver | funky bugs :) |
| 20:41.06 | benjk_ | and some of those techniques can be used to hijack contexts |
| 20:41.14 | dhanes | Hello...I've been all over asteriskguru/voip-info et al, TFOT, and googled 'til the cows came home. Would someone be so kind as to help me with system recordings not working? e.g. *77 gets a 'not found' on any phone, and Unable to add /var/lib/asterisk/sounds/custom/s.wav - Can not read file!. I've uninstalled and deleted Recordings module, rebooted, reinstalled...checked perms all to noavail. :( |
| 20:41.31 | benjk_ | which is particularly interesting when you look at hosted pbxes with multiple clients on the same box |
| 20:41.41 | *** join/#asterisk tessier__ (n=treed@69.43.173.46) |
| 20:41.43 | tessier__ | Hello all! |
| 20:42.09 | tessier__ | Anyone here heard of Telepacket Inc.? Or LoudPacket? |
| 20:42.27 | tessier__ | Scumbags. Major scumbags. They burned me out on VOIP pretty good. But I guess it is time to get back into it. |
| 20:43.27 | *** join/#asterisk SpaceBass (n=sp@static-71-251-230-6.rcmdva.fios.verizon.net) |
| 20:43.30 | SpaceBass | hey folks |
| 20:43.34 | SpaceBass | anyone know if broadvoice is having issues?> |
| 20:43.45 | benjk_ | jhiver, the trouble is that those things are not just ordinary bugs, they are the result of design flaws |
| 20:44.03 | dhanes | spacebass: I just dialed out on BV just fine |
| 20:44.10 | SpaceBass | dhanes, I can dial out |
| 20:44.16 | SpaceBass | but dialing in doesnt seem to be working... |
| 20:44.17 | dhanes | lemme check in |
| 20:44.48 | jhiver | benjk_, yes but you also need to avoid the "version 2" syndrome |
| 20:44.59 | jhiver | of projects that never get finished because it's "version 2" |
| 20:45.09 | jhiver | and everybody wants it to be perfect |
| 20:45.13 | benjk_ | that's a different issue though |
| 20:45.31 | jhiver | the fact is that asterisk remains a solid, industrial code base, and it can be made to be stable |
| 20:45.35 | jhiver | i mean |
| 20:45.37 | benjk_ | what you are talking about is adding features/extending functionality |
| 20:45.38 | Qwell | benjk_: When will we see a patch that fixes these? Or a bug report? |
| 20:45.45 | benjk_ | what I am talking about is getting the house in order |
| 20:45.50 | dhanes | spacebass, i'm getting 'your number cannot be completed as dialed...' |
| 20:45.53 | jhiver | i have these quescom gateways i need to reboot every week or so because they run win nt embedded 4 |
| 20:45.55 | benjk_ | there is no patches that can fix those |
| 20:46.02 | SpaceBass | dhanes, i was worried it was just on my end |
| 20:46.02 | jhiver | and compared with that, asterisk is incredibly solid |
| 20:46.04 | benjk_ | you have to rewrite a bunch of stuff |
| 20:46.06 | benjk_ | one by one |
| 20:46.25 | SpaceBass | been trying to register with e164.org....was afaride that did it |
| 20:46.29 | dhanes | Space: was working for me as of yesterday close of business...and i haven't monkeyed with trunks |
| 20:46.37 | dhanes | so it must be them |
| 20:46.43 | SpaceBass | dhanes, thanks for checking |
| 20:46.46 | dhanes | np |
| 20:47.19 | benjk_ | Qwell, http://trac.openpbx.org/cgi-bin/trac.cgi/browser/openpbx/branches/benjk |
| 20:47.41 | Qwell | okay, great, is it for asterisk? |
| 20:47.46 | Qwell | is it disclaimed? |
| 20:47.53 | benjk_ | its MIT licensed |
| 20:48.26 | benjk_ | you just cant copy "written by Mark Spencer into the files" anything else goes |
| 20:51.46 | docelmo | benjk_ HAHAAH! |
| 20:51.56 | benjk_ | ? |
| 20:52.10 | docelmo | <benjk_> you just cant copy "written by Mark Spencer into the files" anything else goes |
| 20:52.17 | benjk_ | ah |
| 20:52.35 | docelmo | As in Qwell's words.. Its not written properly then |
| 20:52.53 | tessier__ | I think it would have been nice if asterisk were written to be more modular. |
| 20:53.05 | tessier__ | And if it used ser instead of doing its own thing. |
| 20:53.10 | UForgotten | it is pretty modular I thought |
| 20:53.15 | Qwell | ser is a proxy... |
| 20:53.23 | tessier__ | Qwell: Right. |
| 20:53.29 | tessier__ | Qwell: And asterisk isn't. |
| 20:53.31 | tessier__ | Proxies are nice. |
| 20:53.34 | Qwell | so use ser |
| 20:53.40 | Qwell | asterisk isn't supposed to be a proxy |
| 20:53.40 | benjk_ | "Under no circumstances is it permitted for any licensee to take credit for the creation of this software, to claim authorship or ownership in this software or in any other way give the impression that they have created this software. Credit to the true authors ..." |
| 20:54.04 | tessier__ | Is there any way to do third party call control yet? ie. switching the audio between different sources progammatically? |
| 20:54.04 | jhiver | SER is a complete mess to work with, i find it a lot harder than Asterisk, and god knows i've tried |
| 20:54.12 | tessier__ | jhiver: I agree. SER is no jewel either. |
| 20:54.20 | tessier__ | It is way complicated. |
| 20:54.26 | tessier__ | You really need to grok sip. |
| 20:54.59 | benjk_ | if you want SER, use SER |
| 20:55.12 | tessier__ | yeah, yeah |
| 20:55.19 | benjk_ | if you want a monolithic system, use Bayonne |
| 20:55.35 | tessier__ | How is asterisk less monolithic than Bayonne? |
| 20:55.59 | jhiver | tessier__, take a look at ~google voip-info asterisk slimming |
| 20:56.05 | benjk_ | the module loading concept of * has its downs but at least it allows you to disable stuff without rebuild |
| 20:56.15 | *** join/#asterisk mmealling (n=michael@c-24-98-171-50.hsd1.ga.comcast.net) |
| 20:56.36 | tessier__ | I see, so that is what you are counting as modular. Fair enough. |
| 20:56.50 | jhiver | mhhh |
| 20:57.02 | benjk_ | there are different flavours of modular |
| 20:57.03 | jhiver | i wonder if bayonne could be used to do decent SIP / H323 translation |
| 20:57.33 | benjk_ | Bayonne doesn't have a hub and spoke concept |
| 20:57.54 | tessier__ | jhiver: I tried that once with asterisk...oh my god. Lost my mind. |
| 20:57.54 | jhiver | what do you mean? i don't understand |
| 20:58.04 | benjk_ | so any Foo-to-Bar translation requires explicit foo2bar code |
| 20:58.19 | jhiver | benjk_, understood |
| 20:58.32 | jhiver | kind of sucks when you have a lot of foos and a lot of bars |
| 20:58.34 | benjk_ | which has its advantages and disadvantages |
| 20:58.43 | jhiver | increases the number of translators exponentially |
| 20:59.09 | benjk_ | but it simplifies the call and bridging model substantially |
| 20:59.29 | benjk_ | everything has upsides and downsides |
| 20:59.35 | Qwell | that is kinda...gross |
| 20:59.54 | Qwell | would suck to write chan_iax2tosip |
| 21:00.05 | *** join/#asterisk X-Rob (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
| 21:00.20 | tessier__ | How's the call queuing stuff in asterisk these days? Been ages since I looked at it. There used to be a couple different implementations. I am building this system for a 20 person call center. |
| 21:00.21 | benjk_ | well, its not like that with bayonne, it hasn't got chan_foo modules |
| 21:00.28 | Qwell | benjk_: You know what I mean |
| 21:00.50 | fafnir | Quick question... I was told that most voip communication occurs in UDP. I have an application for voip that only uses TCP, is asterisk configurable to use solely TCP over UDP, or is that just not possible? |
| 21:01.03 | Qwell | fafnir: What does the application do? |
| 21:01.10 | benjk_ | I was discussing this with David Sugar some time ago and he had some pretty good arguments in favour of his architecture |
| 21:01.16 | Qwell | and yes, audio and control is often done via udp |
| 21:01.23 | Qwell | s/audio/media/ |
| 21:01.34 | benjk_ | he knows his stuff |
| 21:01.50 | *** join/#asterisk caloi (n=caloi@cpe-24-58-22-17.twcny.res.rr.com) |
| 21:01.50 | benjk_ | like I said, there are advantages and disadvantages |
| 21:02.01 | fafnir | Qwell: application as in use |
| 21:02.06 | fafnir | not as in a program |
| 21:02.08 | benjk_ | hub-and-spoke is certainly more convenient |
| 21:02.17 | Qwell | fafnir: So why not change it to udp? |
| 21:02.23 | Qwell | define "use" |
| 21:02.28 | fafnir | Are you saying "no" |
| 21:02.36 | caloi | hey all - anyone using ser or open ser as a proxy with Asterisk? |
| 21:02.39 | Qwell | fafnir: No, I'm saying your question isn't yet valid. :) |
| 21:02.58 | fafnir | There is no possible way for it to be changed to UDP |
| 21:03.09 | Qwell | okay, what "use"? |
| 21:03.30 | fafnir | nevermind, its a yes or no question |
| 21:04.04 | Qwell | You haven't mentioned any protocols or anything |
| 21:04.10 | Qwell | Your question is invalid. |
| 21:04.45 | JunK-Y | caloi: a lot of ppl do. |
| 21:04.49 | fafnir | Your answer is just plain arrogance |
| 21:04.58 | Qwell | ... |
| 21:05.00 | Qwell | I'm trying to help you. |
| 21:05.17 | benjk_ | you'll need to tell us which protocol you are talking about |
| 21:05.28 | caloi | Junk-Y: That's what i've read, I was hoping to discuss the pros and cons with someone who is using it in a production environment |
| 21:06.03 | caloi | brb... |
| 21:06.34 | benjk_ | for realtime payload like telephone audio, tcp isn't necessarily such a good idea |
| 21:06.56 | benjk_ | it will increase jitter |
| 21:07.04 | *** join/#asterisk caloi (n=Chris@cpe-24-58-22-17.twcny.res.rr.com) |
| 21:07.10 | benjk_ | you don't care about lost packets, up to a point |
| 21:07.18 | benjk_ | its better to skip them |
| 21:07.25 | *** join/#asterisk salmandr (n=salmandr@mdsnwigjbas01-pool0-a158.mdsnwigj.tds.net) |
| 21:07.27 | benjk_ | you don't want them to be resent |
| 21:07.36 | Qwell | fafnir: I prefer not to help people privately. |
| 21:07.51 | fafnir | I wasnt asking for help privatly. |
| 21:07.55 | benjk_ | cause if you resent, the problem of stuff queuing up will make the whole thing worse |
| 21:08.02 | Qwell | So then ask the question in here. |
| 21:08.11 | benjk_ | so better just drop the packets and move on |
| 21:08.24 | benjk_ | that's why telephony uses udp, not tcp |
| 21:08.48 | benjk_ | in other words, timely audio is better than complete audio |
| 21:08.55 | fafnir | i did, its a yes or no question, can asterisk be configured to comepletly forgo udp or what, thank you benjk_ |
| 21:09.13 | Juggie | no |
| 21:09.16 | Qwell | fafnir: I imagine TOR takes udp packets, and converts them to tcp |
| 21:09.17 | florz | fafnir: Yes, it can. |
| 21:09.19 | benjk_ | asterisk handles many protocols |
| 21:09.23 | benjk_ | including telnet |
| 21:09.28 | Qwell | So, it would also be able to handle the "conversion" back to udp |
| 21:09.28 | florz | fafnir: Just edit the source. |
| 21:09.36 | benjk_ | some of those you could possibly to tcp |
| 21:09.37 | fafnir | Qwell: I imagine you need to read up on it then |
| 21:10.21 | *** join/#asterisk tsurk0 (n=tsurko@85.187.160.157) |
| 21:10.23 | Qwell | is tor, or is tor not like a vpn? |
| 21:10.36 | fafnir | I dont know, this isnt the tor support channel |
| 21:10.37 | Qwell | If the answer is yes, then yes, you can use it with asterisk |
| 21:10.39 | Qwell | so ask them |
| 21:10.47 | fafnir | I dont need to, you do apparently |
| 21:10.50 | Qwell | ... |
| 21:11.01 | fafnir | I already have my answers for tor. |
| 21:11.10 | Qwell | Do you, or do you not want help? |
| 21:12.01 | fafnir | As I said, I wanted an answer to a quick question, not in depth questioning to completly solve my problem for me, I am capable of researching the rest of it, and if I run into other problems asking question that are more specific. |
| 21:12.05 | Aurs | Qwell: you should stick to helping me :) |
| 21:12.10 | benjk_ | fafnir, the answer is yes |
| 21:12.16 | fafnir | benjk_, thank you. |
| 21:12.17 | benjk_ | voicemail is sent using smtp |
| 21:12.21 | benjk_ | that uses tcp |
| 21:12.27 | benjk_ | the answer is also no |
| 21:12.35 | Qwell | benjk_: _unless_ tor is like a vpn :) |
| 21:12.36 | benjk_ | cause iax and sip do udp |
| 21:12.39 | florz | benjk_: Nah, the answer is only yes. |
| 21:12.41 | Qwell | or like, say...stunnel |
| 21:13.06 | florz | benjk_: One could, after all, "configure" the source code to only use tcp sockets =:-) |
| 21:13.13 | benjk_ | florz, by configuring asterisk, not anything else, not changing source? |
| 21:13.30 | benjk_ | yeah, well, I didn't want to risk being called names |
| 21:13.47 | florz | benjk_: Well, pointless answers for pointless questions, no? =:-) |
| 21:14.03 | fafnir | i still have to do testing to see if one of the other layers will work in conjunction with tor or not |
| 21:14.19 | benjk_ | florz, seems to yes |
| 21:14.37 | Qwell | fafnir: I have a telephony device. Will it work with asterisk? |
| 21:15.01 | fafnir | Fucked if I know. |
| 21:15.26 | benjk_ | how many lightbulbs does it take to screw an asterisk guru? |
| 21:15.35 | Qwell | benjk_: depends on the asterisk guru |
| 21:15.40 | benjk_ | heh |
| 21:15.46 | Qwell | I personally don't exchange sex for lightbulbs |
| 21:15.54 | fafnir | Qwell: I have a two by four and a box of 1 and 3/4 inch nails, will they poke out the other side if i nail them all the way in so the heads are flush with the wood? |
| 21:16.17 | benjk_ | and how long is a piece of string |
| 21:16.32 | fafnir | arbitraurily long |
| 21:16.47 | Qwell | benjk_: it depends |
| 21:16.52 | Qwell | How thick is the string? |
| 21:16.56 | benjk_ | Qwell, do you know the one with the guy who comes to the shop and asks for lightbulbs? |
| 21:17.05 | Qwell | can't say that I do |
| 21:17.32 | benjk_ | he asks "do you have two four-watt bulbs?" |
| 21:17.41 | benjk_ | no got it wrong |
| 21:17.45 | Qwell | You lose |
| 21:17.47 | benjk_ | four two-watt bulbs? |
| 21:17.57 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 21:18.00 | benjk_ | the shopkeeper asks back, two what? |
| 21:18.23 | benjk_ | nah, too late today |
| 21:18.37 | benjk_ | it was right first time |
| 21:18.38 | Qwell | ... |
| 21:18.42 | benjk_ | :) |
| 21:18.47 | Qwell | does it make a difference? heh |
| 21:19.04 | benjk_ | two what? asks the clerk |
| 21:19.16 | benjk_ | no, four watt, says the guy |
| 21:19.18 | Qwell | "no, 4 watt" |
| 21:19.22 | Qwell | yeah... |
| 21:19.38 | Qwell | I think I can figure out the rest :P |
| 21:19.40 | benjk_ | shop keeper is puzzled and gives in, two what? then |
| 21:19.49 | benjk_ | the guy says "yes" |
| 21:20.02 | benjk_ | the shopkeeper even more consterned says "no" |
| 21:20.07 | *** join/#asterisk Un1x (n=x@CPE001731208485-CM0011ae8a7b0a.cpe.net.cable.rogers.com) |
| 21:21.09 | Qwell | okay, and? |
| 21:21.15 | fafnir | yeah, an? |
| 21:21.17 | Un1x | hey is 1.2.11 the one that was Trunk the one thats got that volume update for Sip Calls? |
| 21:21.31 | Qwell | Un1x: no, 1.2.11 was svn branch 1.2 |
| 21:21.56 | caloi | hey guys, can anyone explain the the difference between branch tags and trunk is? |
| 21:22.05 | Qwell | caloi: sure |
| 21:22.05 | Un1x | Qwell when is trunk going to be released because svn branch hasn't got the Volume thing for sip calls only for the calls made over traditional PSTN. |
| 21:22.18 | Qwell | tags are actually releases, like 1.2.10, 1.2.11, etc |
| 21:22.38 | caloi | so tags are formal releases? |
| 21:22.38 | Qwell | branch 1.2 is where bug fixes are put in - this will eventually become the next release |
| 21:22.42 | *** join/#asterisk aaugustf (n=aaugustf@copper-14.dynamic2.rpi.edu) |
| 21:22.58 | Qwell | trunk is where active development happens, and will become the next major release. pretty much anything goes in trunk |
| 21:23.09 | Qwell | 1.2 branch is generally only bug fixes - no new features |
| 21:23.53 | caloi | Qwell: thanks man, so the best bet, for a "stable" release is to run the latest tags ? |
| 21:24.02 | Qwell | correct |
| 21:24.17 | *** join/#asterisk stephane_ (n=stephane@merlin.cabale.net) |
| 21:24.21 | Qwell | some people like using 1.2 branch, because they get bug fixes quicker |
| 21:24.52 | Un1x | oh so i gotta wait for 1.3 :) before i can use that feature :D |
| 21:25.02 | aaugustf | i've noticed on my asterisk install that any user with a phone can user *72 to forward another user's calls anywhere they want - is there anything to prevent this capability being used maliciously and why is it allowed in the first place? |
| 21:25.12 | Qwell | Un1x: 1.4 |
| 21:25.30 | caloi | Qwell: thanks man :) |
| 21:25.40 | Un1x | :O?. are we skipping one :P? |
| 21:25.48 | Qwell | Un1x: sorta |
| 21:26.00 | Qwell | trunk is often considered to be "1.3", but... |
| 21:26.07 | Un1x | Qwell know of when 1.4 will be available, Ahh i see |
| 21:26.11 | Qwell | soon |
| 21:26.31 | Un1x | Cool |
| 21:26.40 | Un1x | i'm looking forward to the new feature :D. |
| 21:26.46 | Qwell | benjk_: so, what happens next? |
| 21:27.11 | benjk_ | its an example of a "protocol mismatch" |
| 21:27.40 | Qwell | Does he get the right bulbs? |
| 21:27.46 | Qwell | does he even GET bulbs? |
| 21:27.48 | benjk_ | so when the two confuse for what and four watt and no sale can take place, that's the end of it |
| 21:27.54 | Qwell | perhaps he ends up with a hammer |
| 21:27.59 | benjk_ | heh |
| 21:28.09 | benjk_ | smashing all the bulbs in the shop with it |
| 21:28.25 | Qwell | Why doesn't the guy just go look in the aisle for the right bulbs? |
| 21:28.42 | Qwell | Does this store keep $.50 bulbs behind the counter or something? Did they have a theft problem? |
| 21:28.52 | Un1x | haha |
| 21:29.51 | benjk_ | heh |
| 21:30.09 | benjk_ | maybe some old town grocery store |
| 21:30.23 | *** join/#asterisk linlin (i=linlin@c-67-173-38-87.hsd1.il.comcast.net) |
| 21:30.25 | benjk_ | somewhere in the mid west of the US |
| 21:30.37 | benjk_ | like say Alabama |
| 21:30.59 | tessier__ | I'm thinking Snom is my favorite business class SIP phone to use with Asterisk... |
| 21:31.22 | tessier__ | And now that I have a pile of around 30 brand new Snom 320's in my office I can see the hardware has improved a bit since the 200/220 days. |
| 21:31.33 | tessier__ | Even the handset has more weight and fits bettre. |
| 21:31.41 | benjk_ | you have them on a pile? |
| 21:31.49 | Aurs | i like polycom |
| 21:32.58 | *** join/#asterisk pnlarsson (n=niklas@c83-248-0-248.bredband.comhem.se) |
| 21:33.32 | tessier__ | Aurs: I haven't tried polycom yet. When I last set up an asterisk system they did not work with asterisk but I hear they have done a 180 |
| 21:33.38 | tessier__ | I am definitely never touching Cisco again, that's for sure. |
| 21:34.12 | Un1x | lol |
| 21:34.18 | Qwell | What's wrong with Cisco? |
| 21:34.20 | Un1x | i like tho Polycom phone that tripod type one :D |
| 21:34.21 | Aurs | I've seen alot of complaints about polycom and NAT, but have never had any problems with them |
| 21:34.33 | Aurs | +myself |
| 21:35.09 | tessier__ | Qwell: I deployed quite a few with asterisk and they wasted a lot of my time. For one thing Cisco doesn't officially support them with asterisk. For another to be legal you gotta buy the SIP firmware. There are several models of Cisco phones and at least 3 different ways of configuration. |
| 21:35.29 | tessier__ | They come with skinny by default so you gotta flash every one of them. |
| 21:35.37 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) [NETSPLIT VICTIM] |
| 21:35.42 | tessier__ | And of course they are expensive. |
| 21:36.15 | Aurs | tessier__: know of any good wlan phones? |
| 21:36.46 | tessier__ | Aurs: Yeah. Sprint, Cingular, etc sell many millions every year. |
| 21:36.50 | Qwell | "gotta flash every one" says who? |
| 21:36.53 | tessier__ | I like my Treo 650 |
| 21:37.05 | Qwell | You can keep skinny on them |
| 21:37.08 | tessier__ | Qwell: Unless you want to run skinny. |
| 21:37.10 | Qwell | and, upgrading the firmware is dead simple |
| 21:37.43 | tessier__ | Some of them have one config file syntax, some have another. The 7912's (I think) need cisco's special binary to compile the configs... |
| 21:37.50 | Aurs | configuring polycom phones from the web gui is a pain |
| 21:38.02 | Qwell | tessier__: I've never heard of such a thing |
| 21:38.28 | Qwell | and, afaik, the configs are all the same now for sip |
| 21:38.32 | Aurs | tessier__: don't they take config in binary OR xml? |
| 21:39.02 | tessier__ | Qwell: They must have changed over the last year. |
| 21:39.09 | *** join/#asterisk gugaiz (n=gugaiz@host200.200.61.156.ifxnw.com.ar) |
| 21:39.12 | tessier__ | Qwell: Do you still have to pay an extra $80 or whatever for a sip license? |
| 21:39.41 | Qwell | You can get a smartnet contract for about $10. |
| 21:39.44 | tessier__ | prserv.linux I think was the binary that compiled the cisco 7912 configs |
| 21:39.48 | *** join/#asterisk delmar (n=delmar@ip-58-28-155-76.ubs-dsl.xnet.co.nz) |
| 21:39.54 | tessier__ | Qwell: Will that get you unlimited sip licenses or something? |
| 21:39.58 | delmar | im getting lots of " rtc: lost some interrupts at 1024Hz " in my logs. anyone seen this? |
| 21:40.12 | Qwell | tessier__: I don't think there is a "sip license" |
| 21:40.14 | tessier__ | Getting SIP images back in the day was a real pain. |
| 21:40.31 | tessier__ | Qwell: You never heard of buying a "spare" cisco phone? |
| 21:40.45 | tessier__ | Came without license? Supposedly designed to replace one that had a license which broke? |
| 21:40.47 | Qwell | afaik, the license is for CCM |
| 21:41.04 | Qwell | Call your Cisco reseller. |
| 21:41.24 | Qwell | I will, however, ask them about it at VON. |
| 21:41.48 | Qwell | see if I can't get a straight answer from the sales manager that'll surely be there |
| 21:41.54 | tessier__ | http://www.sparco.com/cgi-bin/wfind2?spn=A366339 |
| 21:42.16 | tessier__ | I suspect if you just flash them with a SIP image without paying for the SIP license your phones are running pirated software. |
| 21:42.31 | tessier__ | Not that I care much. But it's just another annoyance. |
| 21:42.33 | Qwell | They don't list a cisco part number for those... |
| 21:42.56 | Qwell | oh, they do, nm |
| 21:43.29 | tessier__ | http://www.z-buy.com/product.asp?item=ET-SWCCMUL7960 |
| 21:43.35 | tessier__ | There's another one. A better deal it looks like too. |
| 21:43.49 | tessier__ | But the Snom's are also cool because they run Linux. That scores points with me. |
| 21:43.57 | X-Rob | Snoms are cool |
| 21:43.59 | Qwell | That second one IS for ccm |
| 21:44.04 | Qwell | It's got ccm in the part number :p |
| 21:44.06 | benjk_ | if you run your Cisco phones with pirated software, then all your phone calls will be charged at a premium and the phone companies will pay Cisco from that |
| 21:44.18 | tessier__ | benjk_: heh |
| 21:44.23 | *** join/#asterisk clive- (n=pirch@dsl-145-13-144.telkomadsl.co.za) |
| 21:44.32 | tessier__ | X-Rob: The only difference between 320 and 360 is bigger screen right? |
| 21:44.37 | X-Rob | yeah |
| 21:45.02 | benjk_ | I think the only difference between 320 and 360 is 40 |
| 21:45.03 | tessier__ | I am setting my whole company up with 320's. We'll have a working phone system on Monday. :) |
| 21:45.20 | tessier__ | And monday afternoon the PRI gets turned up. |
| 21:45.32 | X-Rob | Don't forget the snom call pickup patch |
| 21:45.50 | tessier__ | X-Rob: Yeah, I read something about that. Is that not in the latest asterisk stable? |
| 21:45.58 | X-Rob | Correct, it is not. |
| 21:46.24 | tessier__ | That is one thing that annoys me about this whole voip business. Every phone is different and slightly incompatible. |
| 21:46.40 | X-Rob | http://bugs.digium.com/view.php?id=5014 |
| 21:46.55 | tessier__ | The telco side of things is strictly defined standards but the IP side of things is very fast and loose. |
| 21:47.11 | X-Rob | No, SIP is quite strictly defined |
| 21:47.30 | benjk_ | SIP is white elephantism |
| 21:47.30 | X-Rob | all the extra bits (like, eg, call pickup) aren't. |
| 21:47.31 | tessier__ | X-Rob: SIP, yes. |
| 21:47.34 | tessier__ | X-Rob: Right. |
| 21:47.41 | tessier__ | There are a zillion related RFC's. |
| 21:47.46 | benjk_ | like OSI networking |
| 21:47.52 | tessier__ | And then there is stuff that isn't even RFC yet. |
| 21:47.53 | benjk_ | that was a standard too |
| 21:47.53 | X-Rob | Hah. |
| 21:48.00 | tessier__ | X-Rob: Say, how does the call record button work? |
| 21:48.09 | X-Rob | tessier__, quite poorly at the moment. |
| 21:48.27 | X-Rob | I think it's disabled in the 2006-07-28 patch |
| 21:48.31 | X-Rob | In the earlier ones it worked |
| 21:48.36 | benjk_ | it presume it works by applying pressure to it |
| 21:48.36 | X-Rob | but it was causing grief |
| 21:48.49 | tessier__ | Bummer. Our call center manager would blow their wad if I got that working. I guess that will be a future task. |
| 21:54.28 | *** join/#asterisk adorah (n=Administ@87.68.146.112.cable.012.net.il) |
| 21:54.44 | tessier__ | 1.2 is current? For some reason I thought 1.4 was... |
| 21:54.59 | clive- | tessier, I dont think 1.4 is released yet |
| 21:55.54 | adorah | they promised a release in August..now we r closing August..and nothing.. |
| 21:56.05 | X-Rob | tessier__, easy enough to make work. Or just record everythign by default. |
| 21:56.17 | tessier__ | X-Rob: Yeah, that is the other option. |
| 21:58.32 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkit.com.au) |
| 22:00.06 | gugaiz | hi, Is there any way to know the call progress (cdr status) with zap card? |
| 22:00.32 | gugaiz | I say it because, when I make a call from sip phone to pstn line, in the cdr logs, it ever appears "anwsered" flag. |
| 22:02.45 | marta | bye |
| 22:03.43 | *** join/#asterisk fnordus (n=dnall@s142-179-111-243.bc.hsia.telus.net) |
| 22:04.23 | tessier__ | X-Rob: There are a number of files attached to that snom call pickup bug. Which one do I want for asterisk 1.2.11? |
| 22:04.26 | gugaiz | even though I hook-off the call before the other side answer the call, it always appears as "answered" |
| 22:04.42 | X-Rob | tessier__, the latest one. |
| 22:04.48 | tessier__ | pickup-mgernoth-2006-07-28.patch.txt [^] (12,912 bytes) 07-28-06 05:24 ? |
| 22:05.03 | tessier__ | Ok. They all seem to have different names so it wasn't clear that they were supposed to do the same thing. |
| 22:05.40 | hmmhesays | Aug 26 17:04:25 WARNING[13870]: channel.c:2706 ast_channel_make_compatible: No path to translate from SIP/msp_trunk_1-0079e8a0(256) to IAX2/tech04-2(4) |
| 22:05.43 | hmmhesays | <PROTECTED> |
| 22:08.10 | X-Rob | yes |
| 22:09.07 | hmmhesays | which is odd because I have ulaw for both the iax2 and sip endpoints |
| 22:09.19 | *** join/#asterisk jonnysupersonic (n=jonathan@dsl-165-187-133.telkomadsl.co.za) |
| 22:12.52 | tessier__ | Neat, libpri checks for selinux being enabled and restores file contexts if it is |
| 22:13.02 | tessier__ | I am glad to see selinux being integrated into so much new software. |
| 22:13.09 | tessier__ | It really does make Linux a lot more secure. |
| 22:16.17 | tzafrir_laptop | tessier__, I, OTOH, would rather have a security system that didn't need adjusting every library |
| 22:16.31 | tessier__ | tzafrir: Adjusting every library? What do you mean? |
| 22:17.14 | tzafrir_laptop | hmmm... in this case actually: any new package installer |
| 22:17.37 | tessier__ | Right. Libraries in general don't need to be tweaked unless they somehow involve authentication and then only sometimes. |
| 22:17.45 | tessier__ | But installing new packages is definitely a security related event. |
| 22:17.55 | tessier__ | I think communication with the security system will be inevitable in that csae. |
| 22:17.56 | tessier__ | case |
| 22:18.02 | tessier__ | Installing a virus is just "installing a new package" |
| 22:18.07 | tessier__ | And that is the sort of thing we want to prevent. |
| 22:18.20 | tessier__ | So I think it is logical that installing software has to be changed to work with the security system. |
| 22:20.55 | hmmhesays | so damn asterisk doesn't want to do any transcoding between my sip and iax2 channels |
| 22:22.38 | *** join/#asterisk SomethingISODD (n=Dan@h188.43.63.69.cable.ottr.cablerocket.net) |
| 22:22.45 | gugaiz | any body can help me? :( |
| 22:23.00 | tessier__ | gugaiz: No. You have not asked a question yet. |
| 22:23.03 | SomethingISODD | hello guys |
| 22:23.13 | SomethingISODD | is there anyway to do three way calling all through sip? |
| 22:23.15 | gugaiz | tessier__ yes.. |
| 22:24.03 | hads|home | gugaiz: If you are using an analog card then no. |
| 22:24.09 | SomethingISODD | anyone plz |
| 22:24.20 | hads|home | Analog lines don't do progress. |
| 22:24.37 | hads|home | SomethingISODD: meetme |
| 22:24.41 | gugaiz | oops... |
| 22:24.49 | SomethingISODD | hads|home: is that the only way |
| 22:25.55 | hads|home | Some phones do it too. |
| 22:26.26 | tessier__ | Wow, I just ran into a typo bug in the RHEL kernel which prevented the zaptel driver from compiling |
| 22:26.32 | tessier__ | Google is great. I never would have tracked that down. |
| 22:26.42 | hads|home | ~centosbug |
| 22:26.43 | jbot | rumour has it, centosbug is a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h" |
| 22:26.53 | tessier__ | Yep, that's it |
| 22:26.57 | gugaiz | hads|home, will I always have dispositon=answered with analog lines? |
| 22:27.02 | SomethingISODD | hads|home: ok so asterisk will not handle it other then a conference room |
| 22:27.04 | *** join/#asterisk geoffl (n=geoff@gjctech.plus.com) |
| 22:27.10 | tessier__ | I guess I would have eventually asked in here and you would have pointed that out to me though. :) |
| 22:27.57 | hads|home | gugaiz: Yes. I believe there is experimental callprogress in Asterisk for analog lines, but I don't know if it works or not. |
| 22:28.11 | hads|home | gugaiz: Basically if you want progress you need ISDN |
| 22:28.36 | hads|home | SomethingISODD: Maybe, not that I can think of though. |
| 22:28.54 | SomethingISODD | hads|home: ok thanks |
| 22:29.13 | SomethingISODD | Anyone here that knows how to get three to work over sip with out putting them in to a conference |
| 22:29.38 | gugaiz | hads|home, more than the progress I need to know when the other side pick-up the phone, is it the same? |
| 22:29.50 | hads|home | Yes, same thing. |
| 22:30.20 | hads|home | There is the answer confirmation option to the dial command but it's not something I would use for external parties. |
| 22:30.23 | gugaiz | ok, thanks |
| 22:31.02 | tessier__ | Man, this sangoma driver install is slick... |
| 22:31.53 | SomethingISODD | hads|home: have you ever used ztdummy with out usb-uhci? |
| 22:32.53 | hads|home | SomethingISODD: No sorry, all my installs have cards. |
| 22:32.53 | gugaiz | hads|home, do you know the name of proyect to the callprogress of analog lines? |
| 22:33.04 | hads|home | gugaiz: Huh? |
| 22:33.09 | SomethingISODD | ya my local systems do as well |
| 22:33.09 | *** join/#asterisk tmccrary (n=tmccrary@d14-69-160-83.try.wideopenwest.com) |
| 22:33.17 | SomethingISODD | but its this remote system thats killing me |
| 22:34.10 | geoffl | Hi, I have a problem with voicemail. When the calling party on an incoming zap channel hangs up, asterisk doesn't seem to recognise the event - so I end up with a tied-up zap channel and a very long recording of an engaged tone. |
| 22:34.31 | hads|home | geoffl: Analog line? |
| 22:34.34 | pyrom | What is best for voip, nokia n80 or e60? |
| 22:34.43 | geoffl | pyrom: Yes |
| 22:35.00 | pyrom | n80? |
| 22:35.05 | hads|home | pyrom: The N80 doesn't have a SIP client does it? |
| 22:35.17 | pyrom | Guess you need third party software on n80 |
| 22:35.18 | geoffl | sorry - my last should have gone to hads|home |
| 22:35.33 | hads|home | geoffl: You need disconnect supervision from your telco. |
| 22:35.34 | pyrom | geoffl, we all gotta love mr autocomplete :) |
| 22:36.59 | tessier__ | Yes! It sees both PRI ports on my card! |
| 22:38.02 | tmccrary | I'm having an issue with Asterisk 1.0.7. Timout values are not working with dtmf at all. I want to check for 42000, but it parses it as soon as I hit 4. I've tried using DigitTimout and ResponseTimeout |
| 22:38.30 | hads|home | Why such an old version? |
| 22:38.45 | tmccrary | I think that's what is it debian stable |
| 22:39.13 | nextime | if i understand correctly, fastagi is like agi on network, no way to use it like a eagi, right? |
| 22:39.27 | hads|home | tmccrary: I'd install 1.2.11 from source if I was you. |
| 22:40.57 | tmccrary | great, but that's not really an option at the moment |
| 22:41.28 | geoffl | hads|home: The situation is a little more complex than that. I have an ISDN BRI service terminating on a PBX and I'm feeding Asterisk with two analog channels in a hunt group. When I've finished testing, I'll get my telco convert my ISDN service to analog PSTN. In the meantime, is there a way of setting a maximum length for voicemail messages - that would at least close the line on timeout. |
| 22:42.34 | hads|home | Why not stick with the BRI? |
| 22:43.11 | geoffl | Cost of BRI service is over a grand a year, cost of PSTN-based service is less than a quarter of that. |
| 22:43.20 | *** join/#asterisk znoG (n=gs@162-148-235-201.fibertel.com.ar) |
| 22:43.29 | hads|home | geoffl: Ah, good reason :) |
| 22:44.02 | hads|home | That's cheap. BRI is $110/month here. |
| 22:44.04 | *** join/#asterisk zotz (n=zotz@24.244.133.115) |
| 22:44.51 | tmccrary | my PRI is 150/m ;) |
| 22:45.00 | geoffl | hads|home: GB pounds - i.e. well over $1,500 a year |
| 22:45.27 | tzafrir_laptop | tmccrary, do you need more recent asterisk debs to test with? |
| 22:46.18 | tmccrary | I really don't want to upgrade unless there is some kind of known bug causing this. |
| 22:46.35 | tmccrary | I have 99% of the features I want working, I'm not going to jack the entire thing up by upgrading right now |
| 22:46.46 | hads|home | geoffl: Pricey |
| 22:46.48 | tmccrary | thanks though |
| 22:47.33 | geoffl | hads|home: Yep - it's one reason why I'm trying to get Asterisk working for me (although the main reason is flexibility). |
| 22:48.06 | hads|home | geoffl: You will have a tough time without disconnect supervision, which your PBX probably won't give you. You could try the busydetect option in zapata.conf |
| 22:49.08 | geoffl | hads|home: Thanks - I'll dive into the docs for details. FWIW, I've tried to set absolute call timeout immediately before calling the voicemail application - but the timeout is completely ignored :( |
| 22:49.28 | tessier__ | ZT_SPANCONFIG failed on span 1: No such device or address (6) |
| 22:49.29 | tessier__ | hmmm |
| 22:50.04 | tmccrary | geoffl welcome to my world. DTMF in asterisk kind of sucks |
| 22:50.14 | tzafrir_laptop | you used the wrong number? |
| 22:50.28 | tessier__ | tzafrir: You talking to me? |
| 22:50.48 | tzafrir_laptop | tmccrary, hmmm, any chance relaxdtmf may help? Which card is it, anyway? |
| 22:51.09 | tzafrir_laptop | tessier__, yes |
| 22:51.25 | tmccrary | I'm not using a card, SIP with axvoice |
| 22:51.34 | tessier__ | tzafrir: Ok...Not sure what you mean by "wrong number". Where might I have used a wrong number? |
| 22:51.37 | *** join/#asterisk Grizzy (i=Generic@adsl-68-124-190-164.dsl.pltn13.pacbell.net) |
| 22:51.38 | tmccrary | I have another box doing PRI stuff |
| 22:51.41 | tzafrir_laptop | tessier__, compare zaptel.conf to cat /proc/zaptel/* |
| 22:52.01 | tessier__ | tzafrir: /proc/zaptel contains nothing |
| 22:52.04 | tessier__ | No fiels |
| 22:52.05 | tessier__ | files |
| 22:52.35 | *** part/#asterisk geoffl (n=geoff@gjctech.plus.com) |
| 22:52.40 | hads|home | Got the modules loaded? |
| 22:53.00 | tzafrir_laptop | tmccrary, the DTMF in PRI is part of the signalling |
| 22:53.09 | tessier__ | I am using sangoma A102 and just used their wancfg util to configure it up. Then I edited the /etc/zaptel.conf file and ran ztcfg. zaptel and wanpipe modules are loaded |
| 22:53.22 | tmccrary | okay, well this box is not doing PRI or anything besides SIP |
| 22:53.28 | tmccrary | So, I doubt there is an issue there |
| 22:53.43 | tzafrir_laptop | tessier__, lsmod | grep ^zaptel |
| 22:53.59 | tessier__ | zaptel 196868 1 wanpipe |
| 22:55.39 | tzafrir_laptop | tessier__, so the driver is loaded, but failed to find the card. I believe (I don't know wanpipe) |
| 22:56.01 | tessier__ | tzafrir: That would seem to be the case. The wancfg util sees the card no problem |
| 22:56.17 | tessier__ | I wonder if I don't have some sort of mismatch between what I told wancfg and what I told zaptel.conf |
| 22:56.41 | tessier__ | Interface Name-----> w1g1 |
| 22:56.45 | tessier__ | Operation Mode-----> TDM_VOICE |
| 22:56.52 | tessier__ | TDM Voice Span-----------------> 1 |
| 22:57.04 | tessier__ | I wonder how "Interface Name" relates to zaptel. |
| 22:57.05 | tzafrir_laptop | tessier__, the issue is not what you have on zaptel.conf . It'sthe fact that the wanpipe driver registered no span |
| 23:00.42 | tessier__ | http://www.ip-phone-forum.de/archive/index.php/t-97354.html |
| 23:00.53 | tessier__ | This guy had the same problem and I think maybe he solved it but his answer is in german |
| 23:00.56 | *** join/#asterisk fnordus (n=dnall@s142-179-111-243.bc.hsia.telus.net) |
| 23:01.30 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.186) |
| 23:02.00 | *** join/#asterisk razu_ (n=razu@80-235-91-54-dsl.prn.estpak.ee) |
| 23:02.06 | tessier__ | hmm...he says the linux isdn driver was still loaded. |
| 23:02.24 | tessier__ | I am not using any isdn driver stuff though. |
| 23:03.51 | tzafrir_laptop | tessier__, that card is a different card with a totally different set of drivers |
| 23:04.09 | tzafrir_laptop | you have Sangoma, not ZapBRI, right? |
| 23:04.31 | tessier__ | tzafrir: Correct. I have a Sangoma a102 |
| 23:05.24 | tessier__ | wanpipe: AFT-A102c T1/E1 card found (HDLC rev.25), cpu(s) 2, bus #2, slot #1, irq #209 |
| 23:05.24 | tessier__ | wanpipe: Allocating maximum 2 devices: wanpipe1 - wanpipe2. |
| 23:05.47 | *** join/#asterisk fnordus (n=dnall@s142-179-111-243.bc.hsia.telus.net) |
| 23:13.20 | tessier__ | hmm...I haven't touched anything in /etc/asterisk/zapata.conf yet |
| 23:13.54 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 23:14.41 | tessier__ | I don't see anything in /etc/asterisk/zapata.conf that might be related. |
| 23:16.45 | *** join/#asterisk geoffl (n=geoff@gjctech.plus.com) |
| 23:18.17 | *** part/#asterisk geoffl (n=geoff@gjctech.plus.com) |
| 23:21.28 | rbordeaux | hi znog... |
| 23:22.00 | znoG | rbordeaux: hi. |
| 23:25.43 | *** join/#asterisk r0d3nt|m (n=RatMan@foster.stonedcoder.org) |
| 23:32.42 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 23:43.26 | *** part/#asterisk tmccrary (n=tmccrary@d14-69-160-83.try.wideopenwest.com) |
| 23:45.19 | pyrom | Does nokia e70 has nat troubles connecting from a localIP to your asterisk server with public ip? |
| 23:45.35 | pyrom | ehm, sorry for the bold. |
| 23:45.39 | pyrom | Dont know how that happened. |
| 23:48.26 | caloi | hey guys - anyone want to talk about load balancing an asterisk solution? |
| 23:49.10 | tessier__ | caloi: Wouldn't it be more honest to say "Someone tell me how to load balance asterisk" ? :) |
| 23:49.20 | tessier__ | caloi: And do you really have enough traffic to need to worry about such things? |
| 23:50.20 | caloi | tessier: I'm not asking someone to tell me how to do anything, and yes I do have enough traffic to worry about load balancing and redundancy.... Why else would I ask? For fun? |
| 23:50.52 | tessier__ | caloi: A lot of people get ahead of themselves when considering scalability |
| 23:51.06 | caloi | tessier: agreed |
| 23:51.14 | tessier__ | caloi: If you have that much traffic you probably need to look at SER for call routing etc and use asterisk only as a media server. |
| 23:51.42 | caloi | tessier: that's what I am doing; I just haven't been able to make the decsion if SER will benefit me |
| 23:52.23 | caloi | most of my traffic is going to be application specific (call queue and voicemail) which would need to be passed to Asterisk, so I don't know is SER will benefit me |
| 23:52.56 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
| 23:53.23 | caloi | tessier: are you using SER ? |
| 23:53.33 | tessier__ | caloi: No. I don't have that much traffic. |
| 23:54.11 | caloi | tessier: how do you determine what "that" much traffic is? before you need it? |
| 23:58.26 | tessier__ | caloi: "that" much traffic is so much traffic that you need to start considering balancing traffic across multiple boxes to be able to handle it. |
| 23:59.23 | caloi | tessier: wouldn't it be smarter to solve the problem before it occurs?? |
| 23:59.36 | tessier__ | caloi: Depends. Will it ever occur? For me, at my current shop, never in a million years. |
| 23:59.48 | tessier__ | caloi: For you? Investing zillions of dollars to drum up customers or something? Then perhaps. |
