| 00:11.54 | *** part/#asterisk pronai (n=fortheta@uu-217-199-146-246.uunet.co.ke) |
| 00:15.55 | *** join/#asterisk CrashHD (n=crashhd@c-76-20-22-240.hsd1.ca.comcast.net) |
| 00:16.20 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 00:16.51 | Carp1 | Is it just me, or is there not really any docs on call parking (even though it is supposed to be easy :-\)\ |
| 00:17.28 | *** join/#asterisk kannan (n=kannan@58.68.25.67) |
| 00:17.44 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 00:18.21 | kannan | hello. I am getting "Unknown sip command from <my-sip-prvider-ip> , but the calls are OK. What do i do for this error msg in the astersk cli? |
| 00:18.40 | kannan | actaully unknown sip command 'UPDATE' |
| 00:18.47 | *** join/#asterisk wnfaknd (i=wnfaknd@cpe-24-30-183-141.socal.res.rr.com) |
| 00:19.00 | [TK]D-Fender | Carp1 : ... |
| 00:19.01 | [TK]D-Fender | ~book |
| 00:19.05 | jbot | extra, extra, read all about it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 00:19.15 | wnfaknd | why does my shell window would get stuck when i telnet localhost 5038 |
| 00:19.16 | Nugget | telnet is eeeeeeevil! |
| 00:19.20 | wnfaknd | nothing happens after that |
| 00:19.33 | wnfaknd | i actually have to close the ssh window and open a new one, no way to get out of it |
| 00:20.11 | *** join/#asterisk JunK-Y (n=junky@modemcable140.185-70-69.mc.videotron.ca) |
| 00:20.18 | *** join/#asterisk jesselang|laptop (n=jesse@64.241.37.140) |
| 00:20.27 | JunK-Y | how can i reduce the echo from a meetme conf? ive wctdm loaded. |
| 00:20.42 | *** part/#asterisk russellb (i=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 00:20.49 | *** part/#asterisk jesselang|laptop (n=jesse@64.241.37.140) |
| 00:21.04 | [TK]D-Fender | JunK-Y : Nothing to do with MeetMe, its all Zaptel EC .... |
| 00:21.10 | [TK]D-Fender | ~wglwat |
| 00:21.12 | jbot | rumour has it, wglwat is well, good luck with all that |
| 00:21.15 | [TK]D-Fender | :O |
| 00:21.18 | Carp1 | i've done what the book says |
| 00:21.20 | Carp1 | nothing :0\ |
| 00:21.53 | [TK]D-Fender | Carp1 : PASTEBIN's of your setup would help... |
| 00:22.01 | Carp1 | ok |
| 00:22.04 | Carp1 | Do I need tT? |
| 00:22.10 | [TK]D-Fender | Carp1 : And a description of that actions you've taken to test.... |
| 00:22.37 | [TK]D-Fender | Carp1 : nope. |
| 00:22.46 | *** join/#asterisk anthony] (n=anthony@175.21.188.72.cfl.res.rr.com) |
| 00:24.09 | riddlebox | Carp1, what is so hard about it, in features.conf make sure that park is uncommented, then reload asterisk, then transfer a call to 700, it will tell you it is parked on 701 |
| 00:24.23 | riddlebox | then when you pick it up you just call 701 |
| 00:25.57 | [TK]D-Fender | Carp1 : And there's this thought about looking it up on the WIKI.... |
| 00:26.25 | Carp1 | http://pastebin.ca/341976 |
| 00:26.33 | Carp1 | I did look it up on the wiki |
| 00:29.50 | riddlebox | Carp1, you dont have any extensions that you can park, everything comes in plays a sound and hangs up |
| 00:30.22 | Carp1 | if I dial 100, it rings SIP 100 |
| 00:30.25 | Carp1 | I answer that call |
| 00:30.32 | Carp1 | we can talk back and forth |
| 00:30.34 | riddlebox | ohh missed that |
| 00:30.37 | Carp1 | now I want to park this call |
| 00:30.40 | Carp1 | doesnt work. |
| 00:31.02 | riddlebox | is sip 100 an analog phone or a sip phone |
| 00:31.08 | Carp1 | I just noticed everything under [featuremap] is commented, maybe this is a problem? |
| 00:31.14 | Carp1 | SIP Soft. |
| 00:31.27 | [TK]D-Fender | Carp1 : Thats a resounding "YES" |
| 00:32.11 | riddlebox | Carp1, under general parkext parkpos context, and findslot should be uncommented |
| 00:32.34 | Carp1 | THey are. |
| 00:33.01 | riddlebox | did you uncomment them? |
| 00:33.07 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 00:33.07 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 00:33.19 | Carp1 | Yes. |
| 00:33.29 | riddlebox | Carp1, now you have to restart asterisk |
| 00:35.06 | Carp1 | I havnt changed anything |
| 00:35.18 | Carp1 | I've had this uncommented and have restarted since then. |
| 00:37.27 | JunK-Y | [TK]D-Fender: and how to solve it? |
| 00:37.27 | riddlebox | you said 101 can call 100 and it works good |
| 00:37.42 | Carp1 | no |
| 00:37.46 | Carp1 | 101 isnt a phone |
| 00:37.49 | Carp1 | only 100 is a phone |
| 00:38.03 | Carp1 | when I call from outside, and then dial ext 100 |
| 00:38.08 | Carp1 | i can talk to the outside caller. |
| 00:38.11 | Carp1 | Now I want to park that call. |
| 00:38.32 | riddlebox | Carp1, you do transfer 700, and wait till it tells you something |
| 00:38.41 | Carp1 | I try |
| 00:38.45 | Carp1 | but doesnt tell me anything |
| 00:38.58 | Carp1 | and I can hear all the tones on the other end. |
| 00:38.58 | riddlebox | it is kinda slow it may take a second or two |
| 00:39.13 | Carp1 | Shows nothing in the CLI wither |
| 00:39.15 | Carp1 | either* |
| 00:39.29 | [TK]D-Fender | JunK-Y : Get a real card with on-board EC..... |
| 00:40.15 | JunK-Y | i had no isssue with 1.2.14 |
| 00:40.42 | [TK]D-Fender | JunK-Y : Well something changed in that zaptel revision so I guess you might be best to revert... |
| 00:42.45 | JunK-Y | i prefer fix that instead of reverting. |
| 00:42.59 | orlock | Hmmm |
| 00:43.00 | *** join/#asterisk strav (n=Skrit@dsl-131-189.aei.ca) |
| 00:43.18 | orlock | JT: sorted out my audio quality issues |
| 00:43.28 | hardwire | whois #asterisk |
| 00:43.33 | hardwire | it you of course! |
| 00:43.46 | orlock | JT: Migrated a site onto an Asterisk server i built last night, issues went away |
| 00:43.53 | *** join/#asterisk pounk (n=pounk@ip216-239-77-73.vif.net) |
| 00:44.53 | [TK]D-Fender | ok, I'm off for a bit..... back in a few hours... |
| 00:45.13 | orlock | Carp1: its finnicky, things are constantly changing and some things appear to be completely undocumented? |
| 00:45.30 | Carp1 | Yeah :-\ |
| 00:45.53 | JT | orlock: nice, more details? |
| 00:46.05 | JT | Carp1: not spending time to properly read the book can't help either :P |
| 00:46.09 | *** join/#asterisk yeoman (n=chatzill@ACA76065.ipt.aol.com) |
| 00:46.32 | orlock | JT: servers that were having problems were running SMEserver, which its Centos with a templating system for config files |
| 00:46.45 | Carp1 | I read the parts that I am working on, such as call parking, I followed every step in the book.....and nothing :( |
| 00:47.01 | Carp1 | I come here, I paatebin my conf.....no one knows |
| 00:47.11 | orlock | JT: i set up a bare bones Centos 4. system, no worries. But originally the phones were connecting directly to the voip provider, and they ad quality issues then to |
| 00:47.33 | orlock | Carp1: i cheat and use freepbx |
| 00:47.38 | Carp1 | Lol. |
| 00:47.51 | Carp1 | #freepbx |
| 00:49.01 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
| 00:49.01 | *** mode/#asterisk [+o angler] by ChanServ |
| 00:50.04 | *** join/#asterisk drfreeze (n=Jim@www.freeze.org) |
| 00:50.13 | *** join/#asterisk CrashHD (n=crashhd@c-76-20-22-240.hsd1.ca.comcast.net) |
| 00:50.15 | strav | he there. *** noob question. I'm trying to configure asterisk for receiving faxes through sip. Rxfax, ulaw, libtiff and spandsp seems to be correctly installed. The context I defined for handling incoming faxes is quite basic: 1, answer, 2 wait 3 secs, set variable for fax file then receive fax (rx fax)... When the server receive the incoming fax, the rxfax process seems to be called but no tiff is written in the directory I assigned (which is read/wr |
| 00:50.15 | strav | ite accessible).. any clues? |
| 00:50.24 | drfreeze | Can anyone suggest some good tools to determine BW and jitter? |
| 00:50.37 | CrashHD | anyone recommend a good voip answering service? |
| 00:50.46 | drfreeze | Right now I am using mtr and scp, but that seems kindof crude way to do it |
| 00:50.53 | nibbler_de | drfreeze: mtr |
| 00:51.08 | nibbler_de | if you need long-term statistics -> smokeping |
| 00:51.13 | drfreeze | nibbler_de: using that now, but I don't know how to get a jitter or bw reading out of it |
| 00:51.27 | nibbler_de | it has a jitter mode |
| 00:51.36 | nibbler_de | type j :) |
| 00:51.44 | flenders | again, guys, is there a way to reduce the amount of time it takes to learn the echo (cancel)? |
| 00:51.45 | flenders | now it takes a few seconds, and people are finding it annoying |
| 00:51.56 | nibbler_de | scale should change to "Drop Rcv Avg Gmean Jttr Javg Jmax Jint" |
| 00:52.00 | *** join/#asterisk diclophis-work (n=jbardin@65.203.37.58) |
| 00:52.03 | diclophis-work | hello all |
| 00:52.05 | orlock | drfreeze: ethereal or mtr |
| 00:52.10 | strav | anyone? |
| 00:52.25 | orlock | drfreeze: Ethereal you need to dump the data to a file |
| 00:52.29 | diclophis-work | can anyone recomend a log-file analyzer that will show me the number of actives channels over time? |
| 00:53.14 | wnfaknd | im having a problem when i try to run AST_conf_update.pl |
| 00:53.27 | orlock | drfreeze: but then you can do a pretty decent analysis, it will tell ou about dropped/out of order packets, latency and jitter, etc |
| 00:53.32 | wnfaknd | i get this error: |
| 00:53.50 | anthony] | anyone here want a dedicated server for your pbx's? |
| 00:53.53 | anthony] | .ckear |
| 00:53.54 | wnfaknd | unknown remote host |
| 00:54.02 | wnfaknd | sure anthony |
| 00:54.03 | strav | just tell me if the question is plain too noob. I'll search elsewhere. |
| 00:54.11 | wnfaknd | if you are giving them away :) |
| 00:54.26 | wnfaknd | or priced right at the very least |
| 00:56.51 | wnfaknd | any guru out there that can help the noobs? |
| 00:57.28 | strav | yea, like, we need help, eh. |
| 00:58.45 | *** join/#asterisk lowlevel (n=Stuart@CPE000e0c057fad-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 01:01.44 | k-man__ | do i need NAT to be able to make a call using a voip provider? |
| 01:02.14 | orlock | k-man__: i dont know, do you need nat for the sip device to reach it? |
| 01:02.46 | k-man__ | orlock, good question, and no, i don't, my sip phone can reach it without nat |
| 01:02.51 | k-man__ | so i guess that is a no |
| 01:03.16 | k-man__ | damn it |
| 01:03.31 | orlock | :) |
| 01:03.39 | JT | have no idea what your question really was k-man__ |
| 01:03.43 | k-man__ | which section in sip.conf is used for placing calls |
| 01:03.54 | JT | orlock: doesn't make much sense that it sitll had problems with a hardphone |
| 01:04.09 | k-man__ | jt, general undestanding questions |
| 01:04.21 | k-man__ | jt, i had no problems placing calls with my hardphone |
| 01:04.32 | k-man__ | i did nothing with NAT to be able to do that |
| 01:04.57 | *** join/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
| 01:05.52 | JT | k-man__: you mean setting up a nat gateway, or the nat= parameter in sip.conf? |
| 01:06.13 | *** join/#asterisk J4k3 (i=jsuter@dhcp-12-197-128-58.intrastar.net) |
| 01:06.24 | k-man__ | jt, oh, setting up a nat gateway... do i need to forward any ports from the firewall to the asterisk box, for the asterix box to be able to place calls? |
| 01:06.35 | JT | no |
| 01:06.43 | yeoman | second the guru for noobs request |
| 01:07.24 | JT | that made no sense, yeoman |
| 01:07.49 | yeoman | <wnfaknd>any guru out there that can help the noobs? |
| 01:08.03 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 01:08.09 | JT | ah ok |
| 01:08.46 | yeoman | we're testing ... from #wikiversity-en |
| 01:09.29 | yeoman | any help would be greatltly apreciated |
| 01:10.38 | yeoman | http://en.wikiversity.org/wiki/Topic:VoIP <- that's where we plan to document stuff :) |
| 01:10.44 | k-man__ | jt, so.. if i want to create a dial out only section in sip.conf, that is a type=peer? |
| 01:10.55 | Carp1 | I just reset my SIPURA 2100 and then I was told that the hardware is provisioned to the provider still and I still need to get the admin pass from them....I called them and they said they cant give it to me becuase they dont even know it :-\ |
| 01:12.16 | The_DoC^ | still fighting with that ATA Carp1 |
| 01:12.53 | Carp1 | I'm fighting with everything :( |
| 01:13.19 | Carp1 | The way it sounds with the ATA, its just a pile of hardware now.....cant use it without a password. |
| 01:14.13 | JT | Carp1: yeah probably unless you can reflash it |
| 01:14.25 | JT | that's whole idea of locked hardware |
| 01:14.33 | *** join/#asterisk trelane (i=trelane@unaffiliated/trelane) |
| 01:14.53 | Carp1 | I know that now :) |
| 01:15.03 | JT | yeoman: it's unlikely people can help without a specific request |
| 01:15.05 | The_DoC^ | did you try using admin/no password |
| 01:15.48 | Carp1 | When I bought the service, they said the setup fee was for the ATA that I would own once I recieved it.....thought it meant I would also have control over it. |
| 01:16.00 | Carp1 | Yeah, I tried. |
| 01:16.23 | The_DoC^ | also did you enable the web based configuration |
| 01:16.32 | *** join/#asterisk Shmattie (n=Shmattie@cpe-75-179-182-202.woh.res.rr.com) |
| 01:16.36 | Carp1 | Its already configured |
| 01:16.39 | yeoman | ah OK... we're just starting out... not far enough along yet. Just wanted to let you know that we're doing the tests... sorry to bother ya ;) |
| 01:18.52 | JT | what do the tests involve? |
| 01:19.22 | Carp1 | web based is on yes*** |
| 01:20.48 | The_DoC^ | http://www.dslreports.com/forum/remark,16475749?hilite=unlock+spa2100 |
| 01:21.41 | The_DoC^ | read that several times very closely |
| 01:22.23 | Carp1 | lol |
| 01:22.25 | Carp1 | Thanks. |
| 01:24.30 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 01:24.34 | *** join/#asterisk DaveCanoe (n=Dave@dhcp3022.nanog.merit.net) |
| 01:24.46 | *** part/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
| 01:28.31 | The_DoC^ | read, read, read |
| 01:30.55 | [hC] | <PROTECTED> |
| 01:30.58 | [hC] | Any ideas? |
| 01:33.07 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 01:33.07 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 01:34.43 | *** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 01:35.15 | FuriousGeorge | my two fxs modules dont make a dialtone... everything appears fine, and the cli even says zap-1-1 is ringing but its totally not |
| 01:35.44 | FuriousGeorge | if the molex that i used wasnt working, (which i checked, and think it is) i wouldnt even be able to start *, right? |
| 01:37.23 | *** join/#asterisk thazza (n=me@229.9.233.220.exetel.com.au) |
| 01:37.26 | thazza | Hey all. |
| 01:38.21 | trelane | FuriousGeorge, not necessarily... you might want to pull out a multimeter and make sure that you're both getting the correct voltage, and consistent voltage. |
| 01:38.46 | thazza | Why does SIP have to be such a pain in the arse? |
| 01:38.50 | Carp1 | http://www.dslreports.com/forum/remark,16475749?hilite=unlock+spa2100 |
| 01:39.09 | Carp1 | About half wat down the page, TGPOPI's post |
| 01:39.26 | Carp1 | This only works with Zingotel service right? |
| 01:47.10 | The_DoC^ | you have to change those settings to match your provider |
| 01:51.45 | Carp1 | This is way over my head loll. |
| 01:53.11 | booray | so in 1.4, is there an easier way to set subscribecontext rather than an exten entry for every sip phone? |
| 01:53.14 | hal2385612 | hi all - after installing and loading the zaptel drivers, how can I load them automatically after startup? |
| 01:53.24 | JT | asterisk requires knowledge of linux system administration, networking, telecommunications, as well as asterisk itself to be able to use it with relative ease |
| 01:53.28 | booray | hal2385612: put them in /etc/modules |
| 01:54.18 | The_DoC^ | night all |
| 01:54.45 | hal2385612 | thank you booray - do I just create a new file, called modules? |
| 01:55.29 | Qwell | hal2385612: no |
| 01:55.29 | The_DoC^ | here comes the spoon feeding |
| 01:55.41 | booray | heh |
| 01:55.58 | booray | hal2385612: it's already there, just add to it... |
| 01:56.20 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 01:56.31 | hal2385612 | there is a modules.conf |
| 01:56.34 | hal2385612 | do you mean that? |
| 01:56.43 | booray | hal2385612: no offence of course, but stuff like that is easily answered by typing in your same question to google, usually without scrolling down |
| 01:56.55 | booray | no, not modules.conf. modules by itself |
| 01:57.17 | Qwell | booray: you don't know what distro he's using, so you can't possibly answer his question so vaguely |
| 01:57.19 | hal2385612 | OK, I will try |
| 01:57.33 | hal2385612 | thank you for your help |
| 01:57.33 | Carp1 | Are there any Linux System Administration Video's? |
| 01:57.48 | Qwell | Carp1: search youtube for "lart" |
| 01:58.02 | Qwell | that's how one should administer systems for users :P |
| 01:58.36 | Qwell | hal2385612: he's wrong :) |
| 01:58.47 | Qwell | hal2385612: There are init scripts in zaptel - type `make config` |
| 02:00.05 | hal2385612 | thank you qwell - what do I do next? |
| 02:00.12 | Qwell | reboot |
| 02:00.54 | JT | evil command ;) |
| 02:03.41 | *** join/#asterisk sumasuma (n=kurukko@61.14.86.23) |
| 02:03.44 | booray | quell: you're right |
| 02:03.51 | hal2385612 | nice one, qwell!!! service zaptel restart |
| 02:04.29 | hal2385612 | thank you for your generosity! |
| 02:04.53 | sumasuma | if i terminate with gsm and connect to a zap, if the milliseconds needed to convert from gsm to sline is 7 milliseconds, shall i operate 4 channels on this hardware ? |
| 02:05.51 | bkw_ | RUNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNN |
| 02:06.01 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 02:06.02 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 02:06.23 | bkw_ | JT you mean you can't run asterisk on windows? |
| 02:06.33 | bkw_ | oh why not? :P |
| 02:06.44 | hal2385612 | vmware! |
| 02:06.52 | JT | heh |
| 02:07.34 | JT | sumasuma: that question didn't make sense |
| 02:07.51 | sumasuma | ok |
| 02:08.04 | sumasuma | i have installed asterisk 1.4 in the efika board pegasos |
| 02:08.23 | sumasuma | i want to have TDM400P on that |
| 02:08.32 | sumasuma | with 4 fxo channels |
| 02:09.11 | sumasuma | the other end which i connect is on GSM (codec) |
| 02:09.41 | bkw_ | you can't run asterisk in vmware |
| 02:09.42 | sumasuma | which factor i need to check to have the four channels simultaneously running |
| 02:09.43 | JT | you mean over sip, not the fxo? |
| 02:09.46 | bkw_ | it just won't play nice |
| 02:09.51 | *** join/#asterisk ryguillian (i=rhayes@numbertwo.midphase.com) |
| 02:09.59 | sumasuma | JT: it is over the IAX |
| 02:10.14 | JT | so the codec on IAX is gsm? |
| 02:10.19 | sumasuma | yes |
| 02:10.21 | ctooley | bkw_, I run Asterisk in VMWare every day |
| 02:10.33 | JT | please specify more details on the hardware specifications |
| 02:10.34 | bkw_ | ctooley, ah why on earth hehe |
| 02:10.46 | sumasuma | what conversion is needed to give the voice signal to the FXO ? |
| 02:10.52 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 02:10.52 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 02:10.57 | bkw_ | sumasuma, what do you mean? |
| 02:11.12 | sumasuma | GSM(codec) to FXO ? what conversion is needed, |
| 02:11.17 | JT | sumasuma: the card uses slin internally i think |
| 02:11.20 | JT | well that's gsm to slin |
| 02:11.26 | alrs | I have * running on a VPS 24/7 |
| 02:11.34 | hal2385612 | guys - is the source at http://svn.digium.com/svn/asterisk/trunk stable? At what stage is it? |
| 02:11.36 | sumasuma | that is what i meant before |
| 02:11.37 | alrs | not POTS connection, obviously |
| 02:11.38 | JT | alrs: xen? |
| 02:11.42 | ctooley | bkw_, no cards though |
| 02:11.46 | alrs | It's on openvz on vpslink |
| 02:12.06 | sumasuma | to convert from gsm to slin, core show translation it shows 7 milliseconds |
| 02:12.12 | JT | ok |
| 02:12.24 | JT | never heard of openvz |
| 02:12.28 | sumasuma | is it possible to run simultaneously 4 fxo channels on that hardware ? |
| 02:12.45 | JT | you haven't told us what the damn hardware specs are yet |
| 02:12.47 | JT | i already asked |
| 02:12.49 | alrs | http://openvz.org/ |
| 02:12.53 | sumasuma | what specs you want ? |
| 02:13.00 | alrs | http://vpslink.com |
| 02:13.02 | JT | PROCESSOR SPEED, RAM, ETC |
| 02:13.05 | JT | ARCH |
| 02:13.07 | sumasuma | there are 1000s |
| 02:13.10 | danp | not a big fan of vpslink |
| 02:13.25 | danp | the memory management isn't very good |
| 02:13.39 | *** join/#asterisk hohum (n=dcorbe@c-71-62-76-68.hsd1.va.comcast.net) |
| 02:13.43 | alrs | danp: cheap is the reason for the season |
| 02:14.01 | JT | well it'd be nice to know which, sumasuma, so it says it will add 7ms latency to each channel, doesn't say how many it would handle |
| 02:14.03 | alrs | danp: it's been a good place for me to play with * without having to deal with NAT madness |
| 02:14.30 | alrs | danp: do you have a favorite vps provider? I'm not wed to vpslink in any way. |
| 02:14.32 | bkw_ | ctooley, do you have meetme working on vmware? |
| 02:14.48 | danp | it just sucks that if you're running something like mysql that has a big virtual size you get "charged" for it |
| 02:15.00 | danp | there's no swapping out |
| 02:15.05 | JT | sumasuma: the question you have basically asked is "how many calls can i get on UNKNOWN hardware" without details, it's impossible to even guess |
| 02:15.09 | sumasuma | JT: i did core show translation |
| 02:15.15 | JT | yes i know |
| 02:15.16 | JT | so what |
| 02:15.22 | JT | that only tells you the latency is 7ms |
| 02:15.35 | JT | doesn't tell you concurrent sessions on a given piece of hardware |
| 02:15.35 | sumasuma | yes |
| 02:15.41 | JT | how many times to i need to reiterate that? |
| 02:15.52 | sumasuma | can't we multiply * 4 |
| 02:15.56 | JT | no |
| 02:16.00 | JT | you are talking crack |
| 02:16.01 | sumasuma | i c |
| 02:16.10 | JT | it is the cpu time latency |
| 02:16.15 | JT | nothing more |
| 02:16.42 | sumasuma | if i supply 4 gsm codec data then it will take 7 * 4 milliseocnds ? |
| 02:16.46 | sumasuma | am i wrong here ? |
| 02:16.50 | sumasuma | 4 frames |
| 02:16.55 | *** join/#asterisk adeeln (n=adeeln@c-24-7-133-237.hsd1.ca.comcast.net) |
| 02:17.02 | k-man__ | what is the externip in sip.conf? is that my networks external ip or is it something else? |
| 02:17.11 | JT | anyway, there's a page on the wiki about asterisk performance, have a look at what others have acheived, i can't even guess until you tell me what your hardware is |
| 02:17.31 | JT | sumasuma: not necessarily, really depends on your system |
| 02:17.33 | wnfaknd | where is that page? |
| 02:17.49 | sumasuma | http://www.genesippc.com/efika.php |
| 02:17.53 | sumasuma | this is the hardware |
| 02:18.03 | sumasuma | i remember i told you it is efika before |
| 02:18.10 | JT | http://www.voip-info.org/wiki/view/Asterisk+dimensioning |
| 02:18.11 | sumasuma | i c |
| 02:18.12 | hal2385612 | guys - is the source at http://svn.digium.com/svn/asterisk/trunk stable? At what stage is it, please? |
| 02:18.47 | JT | i'm not going to the web page sumasuma, sorry, but you need to do your own ground work and tell US what the processor MHz is, the architecture, the ram, etc |
| 02:19.07 | bkw_ | sumasuma, what is your goal? |
| 02:19.11 | JT | people here are busy |
| 02:19.14 | hal2385612 | bkw_ said >asterisk at any level isn't stable |
| 02:19.22 | bkw_ | show translation is bunk anyway |
| 02:19.24 | bkw_ | hal2385612, yep. |
| 02:19.25 | JT | and have better things to do than go researching everything for someone else for free |
| 02:19.30 | hal2385612 | why did you say that bkw? |
| 02:19.32 | bkw_ | it all depends on what you're doing with it |
| 02:19.40 | bkw_ | their are things its well suited for |
| 02:19.41 | hal2385612 | I have used it in production and it is extremely stable |
| 02:19.43 | bkw_ | some its not |
| 02:19.51 | bkw_ | hal2385612, max calls and how long do you let it run for? |
| 02:19.55 | sumasuma | bkw_: I want to have 4 port fxo with IAX :) |
| 02:20.02 | sumasuma | and without a PC |
| 02:20.15 | bkw_ | well go get one of those embeded boxen |
| 02:20.20 | bkw_ | problem solved |
| 02:20.29 | sumasuma | bkw_: you know of any ? |
| 02:20.34 | bkw_ | JT, if people were really busy they wouldn't be in here. |
| 02:20.52 | JT | yeah, he's suggested the efika, but is too lazy to tell us its specs |
| 02:21.07 | JT | bkw_: heh, well some questions are easier than others :) |
| 02:21.12 | bkw_ | sumasuma, http://www.rowetel.com/ucasterisk/ |
| 02:21.33 | bkw_ | JT, all questions are easy.. it just depends on how lazy you want to be :P |
| 02:21.37 | sumasuma | bkw_: yes, i'm working on that too, but no box or enclosed ones, efika has boxes .. |
| 02:21.48 | bkw_ | sumasuma, just use a cheap crap PC |
| 02:21.52 | bkw_ | and call it a day :P |
| 02:21.55 | JT | bkw_: yeah, i think the basic legwork should be done by the asker though |
| 02:22.08 | bkw_ | true |
| 02:22.17 | sumasuma | bkw_: it is shipped to developing countries, so running a PC is wasting much of energy |
| 02:22.30 | k-man__ | jt, should i be able to telnet to port 5060 on the local host when astersik is running? |
| 02:22.31 | bkw_ | hal2385612, honestly asterisk isn't stable by my standards... |
| 02:22.42 | bkw_ | I don't accept having to restart it daily to keep it working |
| 02:22.43 | sumasuma | also frequent reboots will happen, i don't think PC can handle frequent reboots |
| 02:22.58 | bkw_ | sumasuma, you run windows? if so then you know a PC can take that |
| 02:23.09 | bkw_ | isn't that how windows is? reboot.. reboot.. reboot.. |
| 02:23.16 | hal2385612 | ok, go to Cisco's IRC channel then - I am hoping to get answers to my specific question - not superfluous opinions |
| 02:23.26 | JT | then go |
| 02:23.35 | sumasuma | running PC is out of question, i bought Asterisk Developer Kit, it works fine but very costly ! |
| 02:23.36 | JT | the world is made of opinions |
| 02:23.38 | bkw_ | hal2385612, I can tell you trunk is not stable |
| 02:23.43 | JT | if you can't weigh them up against fact |
| 02:23.43 | bkw_ | thats why its trunk |
| 02:23.47 | JT | then you are in for a shock |
| 02:24.03 | *** join/#asterisk b1ch0 (n=b1ch0@static-200-105-209-46.acelerate.net) |
| 02:24.24 | JT | the answer is, it's very stable sometimes, very stable other times |
| 02:24.28 | JT | depends what you are doing |
| 02:24.35 | JT | but you should not be using trunk in production |
| 02:24.39 | bkw_ | making a few calls.. works great |
| 02:24.44 | hal2385612 | bkw - please can you just keep your opinions to yourself - I would rather speak to someone with experience |
| 02:24.45 | JT | unless you understand it is likely not stable |
| 02:24.47 | bkw_ | loading it up to the max.. not so stable |
| 02:24.52 | bkw_ | hal2385612, I have experience |
| 02:24.57 | bkw_ | mucho experience |
| 02:25.01 | JT | hal2385612: bkw likely has more experience than you can likely fathom |
| 02:25.14 | bkw_ | it can be stable depending on what you want to accomplish |
| 02:25.34 | JT | hal2385612: so to answer your question, it can NOT be recommended to use svn trunk in production |
| 02:25.38 | bkw_ | and their are ways to do some tasks in a ways that are more stable |
| 02:26.34 | hal2385612 | ok, I appreciate that answer, JT - thank you |
| 02:26.54 | bkw_ | well I did answer the exact same question. |
| 02:27.12 | JT | personally i only recommend 1.2.x, but go 1.4.x if there is a feature that you specifically require, and understand it's a very new release |
| 02:27.16 | hal2385612 | no, you didn't bkw - you told me that asterisk wasn't stable at all |
| 02:27.20 | bkw_ | "<bkw_> hal2385612, I can tell you trunk is not stable" |
| 02:27.27 | orlock | bkw_: WHats your opinion on running say, and smtp server on the same system as asterisk? |
| 02:27.41 | bkw_ | orlock, I wouldn't recommend it unless it was to send email out from the box ONLY |
| 02:28.04 | *** join/#asterisk hilario_vw (n=hilario@128.Red-81-39-245.dynamicIP.rima-tde.net) |
| 02:28.04 | orlock | bkw_: due to CPU or bandwidth? |
| 02:28.17 | bkw_ | orlock, it depends on what you're doing |
| 02:28.20 | orlock | yeah |
| 02:28.24 | orlock | guess it always does |
| 02:28.24 | orlock | :) |
| 02:28.26 | bkw_ | the issue is ifyou run a full email system on it |
| 02:28.33 | bkw_ | then you have a chance of your system getting hozed by smtp |
| 02:28.37 | bkw_ | and phone calls suffer |
| 02:28.41 | orlock | bkw_: say, half a dozen accounts |
| 02:28.42 | orlock | yeah |
| 02:28.42 | *** part/#asterisk hilario_vw (n=hilario@128.Red-81-39-245.dynamicIP.rima-tde.net) |
| 02:28.43 | hal2385612 | I wanted to know, in the context of Asterisk, whether the trunk was stable - not whether I should use it at all |
| 02:28.51 | bkw_ | 1.2 is what I would stick with |
| 02:29.05 | bkw_ | you can at the very least work around any issues because they are well documented |
| 02:29.20 | bkw_ | hal2385612, what exactly are you wanting to use it for? |
| 02:29.39 | sumasuma | bkw_: you know of any hardware similar to asterisk appliance ? |
| 02:29.52 | anthonyl | hal2385612, trunk may or may not work.. dont use it unless your planning on doing development or testing it to report bugs to mantis |
| 02:30.06 | hal2385612 | Yes, thank you anthony, I understand. |
| 02:30.35 | hal2385612 | Is there a place where patches are stored, and can be searched, to resolve specific issues with the stable release? |
| 02:30.49 | Qwell | bugs.digium.com |
| 02:30.51 | bkw_ | their is an asterisk backports thingy |
| 02:30.53 | bkw_ | where isthat? |
| 02:31.00 | bkw_ | didn't someone do backports for 1.2? |
| 02:31.13 | bkw_ | stuff digium refused to backport to 1.2 |
| 02:32.26 | hal2385612 | I presume zaptel 1.4 is not compatible with asterisk 1.2? |
| 02:33.09 | bkw_ | nope |
| 02:33.42 | hal2385612 | was that to me, bkw? Thank you! |
| 02:34.03 | bkw_ | yep |
| 02:34.16 | hal2385612 | Lastly, is there a webpage that clearly presents what the additional features are for asterisk 1.4 over 1.2? |
| 02:35.14 | bkw_ | not sure.. I stopped following asterisk about a 1.5 years ago. But you should check the wiki |
| 02:36.52 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 02:39.24 | *** join/#asterisk CrashHD (n=crashhd@c-76-20-22-240.hsd1.ca.comcast.net) |
| 02:39.50 | hal2385612 | thank you bkw - I have now found it |
| 02:41.30 | bkw_ | np |
| 02:41.34 | bkw_ | I knew their was something out there |
| 02:41.59 | oys | any point in getting a x100p card for use at home? or am I just wasting money? |
| 02:42.41 | `Sauron | bkw: If you stopped following asterisk, what did you switch to? |
| 02:42.43 | J4k3 | oys: x100p worked for me, everyone else's milage may vary. |
| 02:43.57 | sumasuma | bkw_: you know of any hardware similar to asterisk appliance ? |
| 02:43.59 | alrs | is the x100p the clone, or the real thing? I have one of the clones and it does phone things |
| 02:44.04 | alrs | as advertised |
| 02:44.13 | hal2385612 | well, I have to agree, bkw, that there are a lot of changes related to things being depreciated, but little in the way of "must have" features |
| 02:44.20 | hal2385612 | with regard to 1.4 |
| 02:44.22 | oys | J4k3: which one did you get? there are 'authentic' ones and clones. ebay? |
| 02:45.16 | bkw_ | sumasuma, not sure.. just google |
| 02:45.23 | bkw_ | hal2385612, thats how I see it.. very few new things |
| 02:45.25 | JT | alrs: there are NO real ones for sale new anymore |
| 02:45.44 | hal2385612 | so this leaves me with a problem - I have installed zaptel 1.4 - but now I need to install zaptel 1.2. Can I just install zaptel 1.2 over the top, or do I have to remove all the changes? |
| 02:45.47 | *** join/#asterisk map7 (n=map7@teksup41.lnk.telstra.net) |
| 02:48.27 | oys | JT: do some clones work better than others. if so, which? |
| 02:50.15 | JT | oys: impossible to tell really, some claim to be better, but at the end of the day, i've heard nothing independant to verify |
| 02:50.20 | JT | it's pretty much a gamble |
| 02:50.30 | JT | if it "must work" the x100p is not what to get |
| 02:50.56 | wnfaknd | JT, would you knwo why im unable to run AST_conf_update.pl? |
| 02:51.07 | wnfaknd | it says unknown remote host |
| 02:51.45 | orlock | wnfaknd: DNS? |
| 02:51.56 | wnfaknd | its using localhost |
| 02:52.07 | wnfaknd | which resolves fine |
| 02:52.16 | JunK-Y | AGI Rx << GET VARIABLE GROUP_COUNT(2001) |
| 02:52.16 | JunK-Y | AGI Tx >> 200 result=1 (0) |
| 02:52.16 | JunK-Y | any idea why this is always 0 ? |
| 02:52.20 | *** join/#asterisk JT_ (n=jon@unaffiliated/jt) |
| 02:52.55 | *** join/#asterisk xezz (n=xez@trust-it.gr) |
| 02:53.12 | JT | wnfaknd: hmm, what on earth is AST_conf_update.pl? |
| 02:53.59 | wnfaknd | part of vicidial, a GUI client |
| 02:54.04 | wnfaknd | maybe you are not familiar with it |
| 02:54.12 | xezz | hello , i've installed asterisk on ubuntu , first time after install runned /usr/sbin/asterisk and asterisk -r connected fine , then reboot and now asterisk refuse to start |
| 02:54.40 | hal2385612 | make sure the zaptel drivers are loading |
| 02:54.53 | JT | wnfaknd: this is #asterisk, why do you assume people will know about scripts that are part of something totally different? :) |
| 02:54.58 | hal2385612 | the dummy is required, even if you don't have any telephony hardware installed |
| 02:55.31 | oys | jt: it doesn't have to work, but I don't want to throw away money |
| 02:55.32 | wnfaknd | JT, cause no one answers in #asterisk-gui |
| 02:55.33 | wnfaknd | :) |
| 02:55.41 | xezz | asterisk -r outputs : does /var/run/asterisk.. .. .. but yes it exists |
| 02:55.45 | wnfaknd | i figured i'd give it a try |
| 02:55.50 | JT | is it part of #asterisk-gui? |
| 02:56.02 | JT | and that'd be because no-one uses asterisk-gui ;) |
| 02:56.13 | *** join/#asterisk genz (n=erdo@im.jobdig.com) |
| 02:56.16 | wnfaknd | hehehe, c'mon, it aint so bad |
| 02:56.36 | oys | think I'll try getting this one: http://cgi.ebay.com/Professional-OEM-X100P-FXO-PCI-Digium-Asterisk-IP-PBX_W0QQitemZ160081205835QQihZ006QQcategoryZ61841QQssPageNameZWDVWQQrdZ1QQcmdZViewItem#ShippingPayment |
| 02:56.46 | JT | is vicidial even a part of asterisk-gui? |
| 02:56.47 | oys | then I'll sue you all if it doesn't work :) |
| 02:56.52 | oys | thank you :) |
| 02:57.07 | genz | Would anybody be willing to help me with a problem involving dialing out on our T1 only using 4 channels and then locking up? |
| 02:57.09 | wnfaknd | no, not really |
| 02:57.24 | wnfaknd | but enough, i get your point :P |
| 02:57.54 | JT | genz: more details? |
| 02:58.40 | genz | Standard T1 PRI, but dialing out starts giving "all circuits busy" after 4 lines are tied up. It seems to be a problem with the dialplan, but I can't seem to narrow it down. |
| 03:00.46 | JT | ok, then more details in the form of stuff pasted to pastebin.ca :) |
| 03:00.55 | JT | most importantly zaptel.conf and zapata.conf |
| 03:01.15 | JT | maybe extensions.conf too |
| 03:01.20 | JT | is it a full pri or fractional? |
| 03:01.25 | genz | full |
| 03:01.28 | JT | also pastebin the exact error message |
| 03:02.01 | genz | like from verbose 10 via the cli? |
| 03:02.50 | JT | yeah, with pri intense debug turned on too, would be nice |
| 03:03.42 | genz | want all my extensions files? |
| 03:04.19 | genz | (how do i turn on pri intense debug?) |
| 03:05.30 | *** join/#asterisk kuku5 (n=kuku5@71.201.219.72) |
| 03:05.49 | kuku5 | so is 1.4 much better than 1.2 ? |
| 03:06.01 | JT | you type pri intense debug <span> |
| 03:06.44 | JT | sorry |
| 03:06.52 | JT | pri intense debug span <span> |
| 03:06.54 | JT | umm anyway |
| 03:07.01 | JT | you can find that out by typing |
| 03:07.05 | JT | help pri intense debug |
| 03:07.15 | genz | thank you |
| 03:07.23 | xezz | how can i see error logs from asterisk ? the only error i can see (var/run/asterisk/asterisk.ctl?exists?) yes exists , also asterisk.pid |
| 03:07.38 | JT | i only need extension files relevant to calls on the pri, i doubt the problem lies in them though |
| 03:10.03 | genz | http://pastebin.ca/342133 |
| 03:10.49 | genz | If you need it formatted better, please let me know. |
| 03:10.52 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 03:10.52 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 03:12.05 | JT | argh, friggen AMP |
| 03:12.28 | genz | I'm in the wrong channel? I don't mean to waste your time. |
| 03:13.09 | JT | ~freepbx |
| 03:13.17 | jbot | rumour has it, freepbx is unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 03:13.21 | JT | well, you really shouldn't be running a T1 on AMP |
| 03:13.28 | JT | i assume it's important |
| 03:13.43 | JT | AMP has disgusting dialplans, as i can see, and is very hard to maintain |
| 03:14.13 | xezz | can you help me with a matter of asterisk plz ? |
| 03:14.23 | genz | Yes, it is. I just don't have the resources to deploy the system gradually and our local vendor preferred we went with that solution. |
| 03:14.26 | *** join/#asterisk DrukenLPY (n=jdumais@CPE000854ddcdb1-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 03:15.36 | genz | Where would you suggest that I start weeding out the complexity/overhead? Or do I have to rebuild from scratch? |
| 03:15.36 | JT | do you have the resources to meticulously go through it when it doesn't work? (like now) |
| 03:16.34 | genz | A great point. But I have to fly by the "hit by a bus" theory so my employer demanded we go with a solution that a local company would support if I died:) |
| 03:17.03 | JT | easier to start from blank files, look at it, 40 lines of commands to execute one phone call, that really is horrible |
| 03:17.04 | genz | And I would much rather give my time/energy to the Asterisk community than something like Shoretel or Siemens. |
| 03:17.13 | JT | i usually have half a dozen lines max on a basic calls |
| 03:17.33 | JT | lines of apps in the dialplan execution, that is |
| 03:17.43 | JT | it must be terribly inefficient too |
| 03:18.16 | genz | I've been really happy with the performance, so far. 50 phones and the load never goes over 1 and its really polite to my network. |
| 03:18.45 | JT | anyway, the problem is you should be dialling with Zap/g1 or Zap/G1 (depending on what order you want to use your lines in) |
| 03:18.51 | wnfaknd | JT, what output should i be seeing out of telnet localhost 5038 ? |
| 03:18.59 | wnfaknd | cause im getting nothing |
| 03:19.04 | JT | also, how long is the cable between the t1 card and csu/dsu? |
| 03:19.39 | genz | Around 80 feet. |
| 03:20.01 | genz | I could get it down to 50 if that would make a difference, it just involves a lot of dust. |
| 03:20.21 | genz | So you would suggest I start cleaning/leaning the extensions.conf area, correcT? |
| 03:21.02 | *** join/#asterisk mog_home (n=rah@c-71-207-215-93.hsd1.al.comcast.net) |
| 03:21.02 | *** mode/#asterisk [+o mog_home] by ChanServ |
| 03:21.45 | JT | you don't need to get the cable length changed |
| 03:21.55 | JT | you need to get the span= correct in zaptel.conf |
| 03:22.12 | genz | 1,1,0? |
| 03:22.36 | JT | yes |
| 03:23.03 | genz | (time to ditch local vendor, that was his first "omg you have to change that") |
| 03:23.20 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
| 03:23.20 | *** mode/#asterisk [+o angler] by ChanServ |
| 03:24.15 | JT | "omg you have to change that"? |
| 03:24.44 | JT | don't quite follow :) |
| 03:24.51 | *** join/#asterisk dasenjo (n=dasenjo@190.5.196.211) |
| 03:24.53 | genz | When he looked at my settings he said that having 1,1,0 was more than likely the problem. |
| 03:25.10 | JT | well that's wrong, heh |
| 03:25.31 | JT | i think the main problem you are experiencing is you are dialling the channel INCORRECTLY |
| 03:25.40 | JT | a pretty fundamental mistake if your vendor did it |
| 03:25.51 | dasenjo | Hi! I really need help with a warning message ... |
| 03:26.05 | dasenjo | I have a Dial string that works ok |
| 03:26.17 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 03:26.47 | genz | Nope. That problem falls completely on being my fault. |
| 03:26.54 | JT | ah |
| 03:26.55 | dasenjo | if I call the authenticate app before the Dial I got WARNING[15445]: channel.c:2049 ast_indicate: Unable to handle indication 3 for 'SIP/avatar_dasenjo-b548' |
| 03:27.15 | JT | also, in zapata.conf, you specify group 1 twice, delete one instance |
| 03:27.22 | dasenjo | and have no ring tone for the caller |
| 03:27.41 | dasenjo | I do have an /etc/asterisk/inidications.conf .. |
| 03:27.43 | genz | This "group => 1 |
| 03:27.50 | dasenjo | and I'm going insane .. :p |
| 03:27.51 | genz | "? The vendor had me add that today too. |
| 03:28.16 | JT | genz: you are only trying to dial channel 1 |
| 03:28.17 | JT | genz: you must group dial |
| 03:28.36 | JT | => and = are pretty much the same, but you have both |
| 03:28.50 | JT | put only one in, it's undefined behaviour if you have 2 |
| 03:28.54 | genz | Good to know, thank you. |
| 03:29.27 | genz | I'll let the vendor know this stuff too. |
| 03:29.34 | JT | you must dial Zap/g1 or Zap/G1 |
| 03:29.44 | JT | depending on what order you want it to dialout in |
| 03:29.46 | genz | Where is that defined? |
| 03:30.00 | JT | there are also round-robin group dialling methods too |
| 03:30.16 | JT | i dunno, probably somewhere in AMP, can't help you there |
| 03:30.28 | genz | Alright, just curious if it was a .conf setting |
| 03:30.30 | flenders | JT: how are you mate? |
| 03:30.35 | JT | flenders: not bad |
| 03:31.00 | flenders | is there a way to dial using SIP in case all my Zap channels are busy? |
| 03:31.16 | JT | genz: possibly, amp might possibly also overwrite it if you change it manually and it later resynchs |
| 03:32.01 | genz | Where would you define it in a non-amp system? |
| 03:32.11 | JT | in extensions.conf |
| 03:33.27 | *** join/#asterisk Carp1 (n=none@cpe-24-92-37-135.nycap.res.rr.com) |
| 03:34.50 | k-man__ | i have discovered that when i use the "qualify=yes" command, i get an error "method not allowed" when i reload sip.conf |
| 03:35.05 | k-man__ | do i need to have qualify=yes? |
| 03:35.50 | [TK]D-Fender | flenders : Put your SIP dial as the next priority |
| 03:36.13 | [TK]D-Fender | k-man_ : Set it for now to your ITSP's peer entry |
| 03:38.49 | Carp1 | insecure=very fixed all my problems :) |
| 03:39.31 | orlock | k-man__: nope |
| 03:40.01 | orlock | k-man__: its for checking latency |
| 03:41.07 | genz | JT, thank you. I think I got it. I'm going to go run around like a maniac and try to busy out our 23 lines. |
| 03:41.22 | genz | I changed it from |
| 03:41.23 | genz | exten => s,n,Dial(${OUT_${DIAL_TRUNK}}/${OUTNUM},300,${DIAL_TRUNK_OPTIONS}) |
| 03:41.30 | genz | to exten => s,n,Dial(ZAP/g1/${OUTNUM},300,${DIAL_TRUNK_OPTIONS}) |
| 03:41.52 | genz | (and I'm going to disable the freePBX stuff for the time being) |
| 03:42.36 | JT | alright |
| 03:42.57 | genz | Do you have a Paypal account? I'd love to compensate you for your help. |
| 03:43.26 | JT | but does it work? :) |
| 03:43.40 | genz | I'll have that answer to you in 3 minutes, be right back. |
| 03:44.13 | *** join/#asterisk osiris (n=osiris@71.205.27.131) |
| 03:46.13 | hal2385612 | guys - I have all these drivers running - wcusb,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wct4xxp,tor2. How do I disable all those that I don't need? |
| 03:47.40 | hal2385612 | oh, it's OK, I have found the file - /etc/default/zaptel |
| 03:47.41 | hal2385612 | thanks |
| 03:50.48 | *** part/#asterisk xezz (n=xez@trust-it.gr) |
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| 03:58.33 | genz | JT? |
| 03:59.59 | k-man__ | [TK]D-Fender, what is my ITSP? |
| 04:00.21 | orkid | what is an ITSP? |
| 04:00.58 | JT | Internet Telephony Service Provider |
| 04:02.14 | orkid | ooh. |
| 04:02.37 | orkid | what is a good website or place to learn about internet telephony architecture/layout/protocols/etc? |
| 04:02.47 | [TK]D-Fender | orkid : |
| 04:02.49 | [TK]D-Fender | ~book |
| 04:02.56 | jbot | rumour has it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 04:03.19 | orkid | okay, one of the later chapters then. i started reading it already. thanks [TK]D-Fender |
| 04:03.47 | [TK]D-Fender | orkid : You may want to review that "index" thingy ;) |
| 04:04.05 | joe_k | anyone tried to build zaptel-1.4 on freebsd-6.2 ? |
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| 04:07.57 | joe_k | im getting compiler errors |
| 04:08.52 | *** part/#asterisk genz (n=erdo@im.jobdig.com) |
| 04:09.44 | orkid | hmm, not so many people here today... at least not now. are most irc users european?! |
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| 04:18.55 | flenders | [TK]D-Fender: thanks mate! |
| 04:19.55 | trelane | who else has the sangoma s518? |
| 04:20.19 | dasenjo | In http://pastebin.ca/342204 are the two conditions .. the first has the error, the second works perfectly and gives ring tone |
| 04:20.38 | dasenjo | ¿can someone give me just a light, please? |
| 04:20.39 | vvortex3 | im unclear about this, how do you ask the user to enter a sequence of numbers, and pass those numbers to agi? |
| 04:22.30 | *** part/#asterisk joe_k (i=joe@adsl-65-67-81-98.dsl.ltrkar.swbell.net) |
| 04:23.00 | dasenjo | :( |
| 04:26.05 | dasenjo | why don't you like me? :p |
| 04:26.22 | InHisName | anyone using the AC211 ata with the asterisk ? |
| 04:27.35 | orlock | trelane: thats a PCI DSL card? |
| 04:27.40 | trelane | orlock, yeah |
| 04:27.45 | orlock | sort of |
| 04:27.50 | orlock | i have a Traverse Pulsar card |
| 04:27.59 | orlock | which is physycally the same, but different drivers |
| 04:28.00 | trelane | I have determined that it is the ADSL card of doom, and a friend asked if it was vorpal and I wanted a second opinion |
| 04:28.18 | orlock | trelane: GPL drivers, its a full hardware card.. whats the problem? |
| 04:28.23 | orlock | trelane: i have hundreds in use |
| 04:28.44 | trelane | orlock, would you consider it to be the "ADSL card of doom" or the "Vorpal ADSL Card of Doom"? |
| 04:28.53 | orlock | Nope. |
| 04:29.10 | orlock | though we did just swap out 5 to resolve some vop quality issues, but they were not the problem |
| 04:30.00 | orlock | trelane: what problems? |
| 04:30.42 | trelane | none |
| 04:30.47 | trelane | they just freaking work |
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| 04:31.52 | orlock | why adsl card of doom then? |
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| 04:33.05 | trelane | orlock, doom = good |
| 04:33.08 | trelane | (in this case) |
| 04:33.14 | trelane | demise of all others |
| 04:33.16 | trelane | not itself |
| 04:33.20 | trelane | think more RPG |
| 04:33.27 | orlock | ahh |
| 04:33.37 | orlock | trelane: yes |
| 04:33.46 | tim0123 | Anyone know where a good script is to combine in and out file from monitor |
| 04:33.47 | orlock | i have had some problems with them, but have been able to get them resolved |
| 04:34.03 | orlock | even took a system out to the place that makes them to debug one of the few issues i had |
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| 04:37.53 | dasenjo | I'm not getting dial tone before calling the Authenticate app ... |
| 04:37.57 | dasenjo | ¿what can I do? |
| 04:38.09 | dasenjo | see http://pastebin.ca/342204 |
| 04:39.04 | dasenjo | it happends in the first block, in the second one (without auth) everything is alright |
| 04:39.15 | dasenjo | any idea? |
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| 04:55.20 | vvortex3 | grrr |
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| 04:59.47 | vvortex3 | lol how am i gonna almost have dialplan voicemail working on my first asterisk install |
| 04:59.54 | vvortex3 | gettin there |
| 05:00.08 | vvortex3 | good documentation++ |
| 05:00.37 | vvortex3 | but how come "SAY NUMBER 0" doesnt work, any number but 0 works |
| 05:03.46 | *** join/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
| 05:03.57 | BSDTech | ok any one here on 1.4 |
| 05:04.05 | BSDTech | that has flite working |
| 05:04.40 | dj-fu | vvortex3, did you do a manual install/ I thought you were doing trixbox yesterday |
| 05:06.34 | vvortex3 | uh i am using aah |
| 05:06.39 | vvortex3 | and php/mysql/agi |
| 05:06.50 | BSDTech | aah is old |
| 05:06.53 | vvortex3 | truly very little of the work is actually in asterisk |
| 05:06.54 | BSDTech | get trixbox |
| 05:06.58 | vvortex3 | meh, working |
| 05:07.31 | BSDTech | I have a scipt to do a full asterisk install and freepbx |
| 05:07.41 | BSDTech | with out all the trixbox crap |
| 05:07.52 | BSDTech | trixbox has become bloated |
| 05:08.02 | tim0123 | bsdtech i have a question? |
| 05:08.25 | BSDTech | ? |
| 05:08.55 | dj-fu | manual asterisk install wins |
| 05:09.07 | dj-fu | it only took me ~6 hours yesterday with no prior knowledge |
| 05:10.03 | tim0123 | On the trixbox when recording inboung calls it stores the audio files combined what give it that functinality |
| 05:10.06 | [TK]D-Fender | trelane : I have an S518. Lovely card |
| 05:10.20 | trelane | that was pretty much my opinion as well :) |
| 05:10.45 | BSDTech | a script |
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| 05:10.58 | BSDTech | I beliebe its a perl script |
| 05:12.05 | vvortex3 | why does "SAY NUMBER 0" specifically say nothing? |
| 05:12.33 | BSDTech | wrong codec |
| 05:12.52 | tim0123 | i have the trixbox tarball but i cant find it |
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| 05:13.30 | BSDTech | I am about to look at the asterisk gui on a embedded system |
| 05:13.40 | BSDTech | see if we can get it better off |
| 05:13.54 | BSDTech | I like the fact it does not use sql |
| 05:14.20 | BSDTech | but its a matter of learnign how they wrote it |
| 05:14.21 | BSDTech | or look at third lane maybe |
| 05:16.49 | wnfaknd | i need help with the asterisk call manager |
| 05:16.57 | wnfaknd | i keep getting this: |
| 05:17.22 | wnfaknd | Response: Error |
| 05:17.23 | wnfaknd | Message: missing action in request |
| 05:17.31 | vvortex3 | omfg why does it have to be so hard just to make it say number 0 through agi |
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| 05:35.12 | gmfm | anyone know of a queue reporting system that will report calls by DID? We have different marketing campaigns for each DID# and like to track the effectiveness of each |
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| 05:41.31 | BSDTech | Playback(zero) |
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| 05:44.30 | k-man__ | [TK]D-Fender and jt, i just want to tell you that i found the problem i was having... i had somehow incorrectly entered the nodephone user id (which is the phone number) into sip.conf |
| 05:44.41 | k-man__ | i fixed that and it worked fine |
| 05:44.49 | k-man__ | what a drongo i am |
| 05:44.53 | k-man__ | i can't believe i did that |
| 05:45.04 | k-man__ | anyway, thanks for both of your help over the last week |
| 05:45.18 | [TK]D-Fender | k-man_ : np |
| 05:45.46 | k-man__ | sometimes its those stupid things that stump you |
| 05:45.48 | *** join/#asterisk booray (n=booray@static-66-15-145-196.bdsl.verizon.net) |
| 05:45.56 | [TK]D-Fender | k-man_ : yup... |
| 05:46.05 | k-man__ | i guess what threw me was that it didn't really give me an error that indicated it was a login problem |
| 05:46.22 | k-man__ | anyway |
| 05:46.26 | k-man__ | it works nicely now |
| 05:46.30 | JT | k-man__: i see |
| 05:46.34 | k-man__ | very clear and smooth |
| 05:46.49 | JT | didn't you get something working in the meantime or am i confusing you with someone else? |
| 05:47.08 | k-man__ | jt, i manage to get my sip phone to talk to asterisk |
| 05:47.20 | k-man__ | i could do an echo test in asterisk fine |
| 05:47.27 | k-man__ | but i never got anything else going |
| 05:51.47 | *** join/#asterisk BugKhaM (n=LAMER@ppp-58.8.2.85.revip2.asianet.co.th) |
| 05:53.03 | *** join/#asterisk bmg505 (n=leon@c1-174-5.rndf.isadsl.co.za) |
| 05:53.17 | BugKhaM | is there a way not to let another user hear our DTMF within meetme? |
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| 06:06.43 | k-man__ | ok, so now i can make calls... what should i do next? |
| 06:07.13 | rue_mohr | phone for help? |
| 06:07.18 | rue_mohr | :) |
| 06:07.43 | k-man__ | hehe |
| 06:07.44 | k-man__ | yeah |
| 06:08.27 | rue_mohr | hye your farther than I am, I still have to dig out a computer |
| 06:08.39 | k-man__ | what are peoples thoughts on policies like dial 0 to get an outside line? |
| 06:08.48 | k-man__ | rue_mohr, vmware is your friend ;) |
| 06:09.01 | x86 | k-man_: usually (in the US) you dial 7 or 9 |
| 06:09.18 | x86 | k-man_: it's almost never 0, as 0 is reserved for the operator |
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| 06:13.14 | k-man__ | x86, oh, interesting, over here (in AUS) it is often 0 |
| 06:13.20 | k-man__ | as 0 is never opperator |
| 06:13.36 | k-man__ | there is an interesting side effect though |
| 06:13.46 | k-man__ | in au. you dial 0011 for idd |
| 06:13.59 | k-man__ | so if you are used to dialing 0 0011 to call idd from work |
| 06:14.27 | k-man__ | then you go home and dial 0 0011... you dial 000 by accident which is equivalent to 911 over here |
| 06:16.05 | k-man__ | apparantly that is one of the biggest causes of incorrect use of 000 over here |
| 06:16.11 | k-man__ | in case you wanted to know |
| 06:20.48 | x86 | what an odd combination for an emergency dialing code |
| 06:21.09 | x86 | here we dial 011 for idd |
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| 06:25.58 | [TK]D-Fender | k-man_ : Prefixes to dial out is so 80's.... |
| 06:26.13 | [TK]D-Fender | Screw them all |
| 06:26.30 | [TK]D-Fender | k-man_ : make your dialplan as close to natural dialing as possible |
| 06:31.24 | [TK]D-Fender | ok, checkout time... later all |
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| 06:46.21 | booray | so after a queue times out, is there any reason a caller would be resubmitted rather than being pushed to the next level in the dialplan? |
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| 06:53.33 | k-man | what is freepbx? |
| 06:54.13 | CunningPike | ~freepbx |
| 06:54.21 | jbot | rumour has it, freepbx is unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 06:54.54 | k-man | ooh |
| 06:54.54 | k-man | scary |
| 06:55.02 | k-man | i was only asking what it was, not for support |
| 06:55.18 | k-man | so its based on asterisk though? |
| 06:55.25 | k-man | or its compleetly different? |
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| 06:56.10 | CunningPike | k-man: It's based on asterisk - it basically provides a "PBX in a box" with a web-based front end |
| 06:56.27 | k-man | ok, thanks |
| 06:56.52 | CunningPike | k-man: I didn't mean the definition to be so terse........ sorry |
| 06:56.59 | k-man | CunningPike, its ok |
| 06:57.05 | k-man | :) |
| 06:57.14 | k-man | warm fuzzy feelings allround |
| 06:57.17 | CunningPike | lol |
| 07:03.16 | orkid | is there hardware out ther eother than digium (and clones) that work with asterisk? |
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| 07:04.21 | zeeesh | hi |
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| 07:15.03 | adeeln | hey all, i'm using asterisk 1.2.14 on gentoo...and i'm running into a problem where i can't bridge calls, resulting in only unidirectional conversation...that is, i can't hear the other person talking, but they can hear me....i have no idea how to fix it...i've searched around online, and can't seem to find anything useful |
| 07:15.47 | CunningPike | orkid: Sangoma |
| 07:18.24 | CunningPike | adeeln: Any CLI messages? |
| 07:19.55 | adeeln | CunningPike: nothing in particular that stands out...i've seen a few times it say that the bridging failed...anything your looking for specificall? |
| 07:20.49 | CunningPike | adeeln: Is this SIP-to-SIP? There are lots of things that cause one-way audio - NAT, codec mismatch |
| 07:22.05 | adeeln | CunningPike: i've mainly been using SIP-to-SIP...only thing is, that my box is on a static ip, somewhere not behind a nat, but i'm at home behind a nat |
| 07:22.23 | *** join/#asterisk evgeni (i=evgeni@212-7-20-235.stv.ee) |
| 07:22.51 | CunningPike | adeeln: So, one of the endpoints is behind NAT? |
| 07:22.53 | adeeln | CunningPike: calling between extensions works just find, irrespective of where they are physically (behind nat, etc) |
| 07:23.13 | CunningPike | adeeln: OK- so, what's not bridging then? |
| 07:23.18 | adeeln | CunningPike: yeah, one is behind nat...but it also doesn't seem to work even if it is not behind nat |
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| 07:24.02 | adeeln | CunningPike: if i dial out to a DID, or even if i call into a DID, it won't bridge the audio for the local extension & trunk |
| 07:24.32 | *** part/#asterisk TekAdmin20XX (n=prodroom@c-68-55-105-37.hsd1.md.comcast.net) |
| 07:25.15 | CunningPike | adeeln: So, ZAP to SIP then |
| 07:25.41 | adeeln | CunningPike: well i don't have any zap channels defined, i was using a soft-sip phone to make the calls |
| 07:26.55 | CunningPike | adeeln: OK - what codecs are you using at each end? |
| 07:27.34 | *** join/#asterisk shinux__ (n=shinux@196.220.27.131) |
| 07:30.16 | adeeln | CunningPike: ulaw/alway |
| 07:30.19 | adeeln | er ulaw/alaw |
| 07:30.41 | x86 | adeeln: which one? |
| 07:31.27 | adeeln | let me pastebin the cli sip debug output |
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| 07:34.44 | adeeln | http://www.pastebin.ca/342363 |
| 07:35.55 | *** part/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 07:39.21 | adeeln | is it a codec mismatch? |
| 07:39.44 | adeeln | from what i make of the logs, it seems that i'm using ulaw |
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| 07:46.55 | adeeln | x86: here's a more verbose output pastebin....http://www.pastebin.ca/342371 |
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| 07:53.39 | tzafrir | adeeln, is the call actually rejected there? |
| 07:54.05 | adeeln | tzafrir: the call is received & accepted, but i can't get full duplex conversation going |
| 07:54.39 | tzafrir | adeeln, so this is probably not a codec problem |
| 07:55.05 | tzafrir | is the rtp going through Asterisk or directly between the parties? |
| 07:55.29 | adeeln | tzafrir: i'm pretty sure it's going through Asterisk, but is there an easy way to double check that? |
| 07:55.43 | tzafrir | If you get audio even in one direction, this is not a codec issue |
| 07:56.00 | adeeln | is there a range of ports for RTP? |
| 07:56.02 | tzafrir | sounds more like nat/firewall related |
| 07:56.32 | tzafrir | <PROTECTED> |
| 07:56.46 | adeeln | alright, let me check that |
| 07:58.19 | *** join/#asterisk misc-- (n=misc@203.87.183.134) |
| 07:59.34 | adeeln | tzafrir: i opened the ports, and still no change |
| 08:00.22 | misc-- | does asterisk@home support sata disks? |
| 08:00.54 | tzafrir | misc--, astersk@home is centos 4.2 or 4.3 |
| 08:01.06 | misc-- | hmm so it should then |
| 08:01.08 | tzafrir | generally it should support most of them |
| 08:01.16 | misc-- | for some reason it's not detecting the disk but the bios can see it |
| 08:01.30 | tzafrir | misc--, ask on #freepbx |
| 08:01.37 | tzafrir | or on #centos |
| 08:01.48 | misc-- | ok no probs |
| 08:05.04 | adeeln | tzafrir: any other ideas as to what could be wrong? |
| 08:05.20 | adeeln | tzafrir: i've tried using a stun server as well on my end (since i'm behind a nat) |
| 08:06.38 | *** join/#asterisk lorinc (n=ang@caracas-0862.adsl.interware.hu) |
| 08:07.34 | tzafrir | adeeln, I believe that asterisk does not work well with stun |
| 08:07.42 | tzafrir | try siproxyd or whatever |
| 08:08.27 | adeeln | tzafrir: what's siproxyd? |
| 08:09.31 | adeeln | tzafrir: does alsa have to be working? would that affect asterisk? |
| 08:09.57 | tzafrir | hmmm... I believe I got the name wrong |
| 08:10.03 | tzafrir | ignore that |
| 08:10.45 | *** join/#asterisk Fr0zen_ (i=Fr0zen_@c-67-175-92-171.hsd1.il.comcast.net) |
| 08:10.48 | Fr0zen_ | sup all |
| 08:10.52 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 08:10.52 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 08:13.24 | adeeln | tzafrir: know of anything that might be able to help? |
| 08:13.52 | tzafrir | voip-info.org has a big summary page for asterisk, sip and NAT |
| 08:14.32 | tzafrir | should probably be linked from http://voip-info.org/wiki/view/Asterisk+config+sip.conf |
| 08:14.43 | adeeln | tzafrir: yeah i've read that...and that is mainly for people who's box is behind a NAT/router |
| 08:14.55 | adeeln | my box is open, but my endpoint is behind a router |
| 08:15.12 | tzafrir | so set nat=yes for the relevant account |
| 08:15.38 | adeeln | it's mainly that i can't transmit from my endpoint behind the router, to the box |
| 08:16.10 | adeeln | tzafrir: nat=yes is already set for the endpoint |
| 08:18.38 | *** join/#asterisk uwe (n=uwe@dogbert.palnet.com) |
| 08:26.13 | *** join/#asterisk inspired (n=mikael@cl-330.sto-01.se.sixxs.net) |
| 08:27.09 | Fr0zen_ | I have a question to those in here. What exactly will a Asterisk server allow a home user to do using a service like via talk with a Cisco 7961G IP phone? |
| 08:31.16 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 08:33.32 | *** join/#asterisk expat_iain (n=iain@217.15.107.200) |
| 08:34.01 | dj-fu | Fr0zen_, it'll allow the home user to use his phone |
| 08:34.41 | dj-fu | or rather, it'll allow the home users IP phone to make an IP connection to the asterisk server which is running your service, and route calls through that |
| 08:35.04 | dj-fu | it'll also have the capability of doing many other things, but that shouldn't be up to the home user, really |
| 08:35.44 | Fr0zen_ | i se |
| 08:35.49 | gerhard7 | i have a problem |
| 08:35.52 | Fr0zen_ | I see, the ip phone supports SIP |
| 08:36.04 | Fr0zen_ | it can connect directly to via talks sip servers |
| 08:36.09 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 08:36.13 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 08:36.15 | Fr0zen_ | but a few users of the service told me to setup a asterisk box |
| 08:36.26 | Fr0zen_ | to i can use the full features of the box and use the SCCP firmware. |
| 08:36.31 | gerhard7 | i have two telephone numbers from the same voip provider |
| 08:36.55 | dj-fu | Fr0zen_, what does the sccp firmware do |
| 08:37.03 | gerhard7 | but i can't seperate them into different contexts .. they enter the context of the other |
| 08:37.04 | dj-fu | Fr0zen_, the full features of the box are only what you make of it |
| 08:37.33 | gerhard7 | i think due to the fact that they come from the same ip |
| 08:37.57 | dj-fu | it should handle two different phone numbers into two conexts quite easily |
| 08:38.06 | dj-fu | are they incoming sip connections or two way? |
| 08:38.17 | gerhard7 | dj-fu: sip |
| 08:38.37 | dj-fu | yeah, are you going to be calling out on both of them, or just calling out on one? |
| 08:38.43 | gerhard7 | dj-fu: am i making a mistake somewhere? |
| 08:39.03 | Fr0zen_ | so an asterisk box basically allows small businesses a sort of gateway for ip phones and such to connect to so that the gateway can connect to the net and other sip servers? |
| 08:39.11 | dj-fu | if you're not calling out on both of them, then setup the second sip connection as a different type=, make it just so that it's incoming. I forget if it's user or peer. I' say peer |
| 08:39.13 | gerhard7 | du-fu: on both |
| 08:39.17 | dj-fu | oh |
| 08:39.19 | dj-fu | nevermind then |
| 08:39.34 | dj-fu | sorry, no idea, perhaps you need to define your sip stuff a little better |
| 08:39.57 | dj-fu | Fr0zen_, yes, more importantly, it's a free open source PBX. you don't need a pbx for one phone |
| 08:40.05 | gerhard7 | dj-fu: one number is for my daughter |
| 08:40.09 | dj-fu | there are some good features for single phone setups though from what I've read |
| 08:40.18 | dj-fu | a menu, voicemail, that kind of stuff |
| 08:40.30 | dj-fu | gerhard7, sorry I don't know, have you read "the book"? |
| 08:40.32 | dj-fu | ~thebook |
| 08:40.39 | jbot | methinks thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 08:40.46 | gerhard7 | dj-fu: yes, i read it |
| 08:41.00 | dj-fu | Fr0zen_, i'd suggest you take a read through that too |
| 08:41.04 | gerhard7 | yes, this book :) |
| 08:41.25 | dj-fu | gerhard7, there is a section regarding sip connections. How are you currently registering your sip connections? |
| 08:41.36 | dj-fu | with a register => user:password@host line? |
| 08:42.07 | gerhard7 | yes, i registered them so |
| 08:42.44 | dj-fu | perhaps you should try registering them in another method, one that allows you to specify a context for each regitsered sip connection |
| 08:43.02 | gerhard7 | register => 071514xxxx:secret:071514xxxx@tiscali2 |
| 08:43.15 | dj-fu | wtf |
| 08:43.35 | dj-fu | that's why they're going into the same context. you should have two register statements |
| 08:44.06 | gerhard7 | register => 071785xxx:secret:071785xxx@tiscali1 is the other |
| 08:44.23 | gerhard7 | yes, two |
| 08:44.37 | dj-fu | oh |
| 08:44.38 | dj-fu | cool |
| 08:44.44 | gerhard7 | what would be another method? |
| 08:44.55 | dj-fu | same way you register sip phones I'd say, i'm not sure |
| 08:45.14 | dj-fu | perhaps you can specify what context it goes onti, I don't know |
| 08:45.31 | dj-fu | I started with asterisk yesterday, I don't even have an incoming sip connection |
| 08:45.37 | gerhard7 | yes, i do that |
| 08:45.56 | gerhard7 | you can try freeworlddialup |
| 08:46.04 | gerhard7 | i got that working |
| 08:46.39 | Fr0zen_ | thanks dj-fu, i think asterisk is a bit more then I really need, espacially for 1-2 phones. It would be great for a business with 30-40 phones though. Voicemail, music on hold, good stuff. |
| 08:47.08 | dj-fu | Fr0zen_, where I work, I installed asterisk because the phones are nice, functional, and we can have calls directed to different departments |
| 08:47.18 | dj-fu | we've only got four phones, but it's kinda helpful |
| 08:47.45 | *** join/#asterisk af_ (n=getsmart@ip-179-53.sn1.eutelia.it) |
| 08:53.01 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:53.28 | jeremy_g | morning! |
| 08:53.29 | Fr0zen_ | i see |
| 08:53.39 | Fr0zen_ | is * similar to Cisco's call manager? |
| 08:53.52 | jeremy_g | damn, do i have create the tables myself while setting up asterisk real time (extconfig stuff) |
| 08:53.53 | jeremy_g | ;( |
| 08:54.06 | jeremy_g | Fr0zen_:yes |
| 08:54.08 | *** join/#asterisk DrukenLPY (n=jdumais@CPE000854ddcdb1-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 08:54.19 | Fr0zen_ | where it receives calls from ip phones and sends them to a gateway to a PSTN |
| 08:54.22 | Fr0zen_ | i see jeremy_g |
| 08:54.25 | Fr0zen_ | voip is so new to me |
| 08:54.31 | Fr0zen_ | i've been reading about it all night |
| 08:54.38 | Fr0zen_ | i think i have a decent understand of how it works now |
| 08:55.58 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 08:57.17 | *** join/#asterisk Dibbler_XP (n=Dibbler@host217-45-198-229.in-addr.btopenworld.com) |
| 08:59.08 | J4k3 | don't be so sure :D |
| 08:59.09 | J4k3 | hehe |
| 09:05.17 | InHisName | anyone using the AC211 ata with the asterisk ? |
| 09:06.11 | *** join/#asterisk newbie1 (n=Main@203.208.196.140) |
| 09:06.24 | newbie1 | hello everybody |
| 09:06.32 | newbie1 | does asterisk support PRBT? |
| 09:06.42 | newbie1 | I mean ring back tone |
| 09:10.52 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 09:10.52 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 09:13.42 | *** join/#asterisk jm|work (n=jm@sentry.flags.co.uk) |
| 09:15.35 | newbie1 | anybody there? |
| 09:15.51 | InHisName | maybe not, gotta hit sack |
| 09:16.05 | newbie1 | sounds all are dead ! |
| 09:17.02 | InHisName | groans and moans or just plain nothing ? |
| 09:18.27 | x86 | anyone ever setup paging with polycom IP601's? |
| 09:18.42 | newbie1 | does asterisk support PRBT? |
| 09:18.48 | newbie1 | I mean ring back tone |
| 09:23.53 | Chris-NB | hi |
| 09:24.11 | Chris-NB | anyone can explain pridialplan and prilocaldialplan to me? |
| 09:24.23 | jeremy_g | newbie1>:deos #asterisk support dumb questions |
| 09:36.30 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 09:36.34 | *** join/#asterisk zeeesh (i=zeeesh@202.38.55.125) |
| 09:36.48 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 09:36.53 | zeeesh | hi |
| 09:38.04 | dj-fu | what's usually the cause of audio only going one way on a sip connection? |
| 09:38.41 | *** join/#asterisk _mh (n=largo@202.5.145.13) |
| 09:40.28 | Chris-NB | dj-fu, problems with a firewall/nat |
| 09:42.34 | adeeln | Chris-NB: i have my asterisk box outside my nat, and i'm behind a nat...now i can hear the callers, but they can't hear my responses, and i've tried mapping the RTP ports and doing the STUN server stuff, but to no avail |
| 09:43.28 | Chris-NB | have you defined nat for your sip device? |
| 09:43.33 | adeeln | yes |
| 09:44.14 | Chris-NB | have you entered asterisk as the default outgoing proxy? |
| 09:44.38 | adeeln | where would i do that? |
| 09:44.46 | Chris-NB | what sip device are u using? |
| 09:44.59 | adeeln | i was using a few different sip software clients |
| 09:45.17 | adeeln | i've tried sjphone, bolsipphone, phonerlite |
| 09:45.27 | Chris-NB | within xlite u can define default outgoing proxy |
| 09:45.47 | adeeln | would that just be the ip of my asterisk box? |
| 09:45.50 | Chris-NB | trace on asterisk box if our rtp stream 1. comes from your phone to server 2. where the server sends it |
| 09:45.59 | Chris-NB | jep |
| 09:46.07 | Chris-NB | or dns if you have/prefere |
| 09:46.33 | adeeln | i have my box already defined as my proxy/registrar |
| 09:46.40 | adeeln | and the username/pass is correct |
| 09:47.11 | adeeln | should i put anything for the realm/domain? |
| 09:48.46 | *** join/#asterisk booray (n=booray@adsl-71-156-53-250.dsl.irvnca.sbcglobal.net) |
| 09:49.36 | Chris-NB | jep. what you put in your sip.conf |
| 09:49.39 | *** join/#asterisk phearless (n=phear@host81-138-68-106.in-addr.btopenworld.com) |
| 09:49.48 | adeeln | let me pastebin it |
| 09:50.20 | Chris-NB | but afair, it is not necessary for xlite |
| 09:50.22 | Chris-NB | k |
| 09:52.10 | adeeln | hmmm....i don't see anything for realm/domain in the sip.conf or sip_additional.conf files |
| 09:54.16 | *** join/#asterisk UlbabraB (n=salama@host241-43-static.72-81-b.business.telecomitalia.it) |
| 09:55.35 | jeremy_g | adeeln:kia musla hai ? |
| 09:55.44 | jeremy_g | :D |
| 09:55.48 | adeeln | haha |
| 09:56.07 | jeremy_g | there my vocabulary ends |
| 09:56.09 | adeeln | jeremy_g: i can't get my asterisk box to apparently bridge the 2 legs together so i can have 2 way voice |
| 09:56.23 | adeeln | jeremy_g: i can speak it, but i can't write it |
| 09:56.32 | jeremy_g | waht? |
| 09:56.37 | jeremy_g | are you a newbyie or sth |
| 09:56.39 | BrokenNoze_ | anyone had problems with DTMF and Polycom 600 series? I can't get asterisk to register DTMF from these phones |
| 09:57.09 | adeeln | not really a newbie...been using it for about 6 months...just upgraded to the asterisk 1.2.14 thinking it'd be stable and easy to use |
| 09:57.40 | adeeln | any my setup is a little odd too...my box is on the outside of a nat, while i sit behind a nat |
| 09:58.00 | jeremy_g | um myself pretty scared of 1.4 |
| 09:58.44 | Chris-NB | adeeln, I'm also behind NAT and my * has a official IP outside and I've no problems with that |
| 09:58.44 | adeeln | i'm working on getting 1.4 working..but that's later on...i just can't seem to get 1.2.14 to give me full duplex conversations...i can usually get audio only in one direction |
| 09:59.08 | jeremy_g | adeeln:don't listen to Chris-NB, he lies around just to promote wrong things |
| 09:59.11 | jeremy_g | :P |
| 09:59.13 | BrokenNoze_ | adeeln: check your firewall settings |
| 09:59.19 | Chris-NB | ??? |
| 09:59.20 | jeremy_g | yep |
| 09:59.25 | jeremy_g | check them |
| 09:59.33 | adeeln | BrokenNoze_: i've set and reset the firewall, but that doesn't seem to do anything |
| 09:59.41 | Chris-NB | adeeln, this is for the realm in sip.conf: realm=mydomain.tld in general section |
| 09:59.48 | BrokenNoze_ | firewall on your router AND on Linux? |
| 10:00.06 | BrokenNoze_ | and the linux security? I had problems with that switched on |
| 10:00.22 | jeremy_g | BrokenNoze_:which distro? suse? |
| 10:00.27 | jeremy_g | or fedora prolly |
| 10:00.30 | adeeln | BrokenNoze_: no extra linux security enabled |
| 10:00.33 | BrokenNoze_ | I'm fedora |
| 10:00.37 | adeeln | i'm using gentoo |
| 10:00.59 | Chris-NB | adeeln, are u rinning iptables on that box? |
| 10:01.05 | BrokenNoze_ | Um.. I had the same problem. u using a NAT? |
| 10:01.17 | *** join/#asterisk nilkanthp (n=nilkanth@dialpool-210-214-10-245.maa.sify.net) |
| 10:02.39 | adeeln | no, no iptables |
| 10:03.06 | Chris-NB | have you checked if the rtp stream from your sip client reaches the server? |
| 10:03.09 | Chris-NB | tcpdump |
| 10:03.44 | adeeln | yeah i've checked it |
| 10:05.18 | BrokenNoze_ | is it one director OR the other? randomly? |
| 10:05.59 | BrokenNoze_ | I assume you've set nat=yes on your Sip.conf? |
| 10:06.04 | adeeln | yup |
| 10:06.10 | adeeln | let me pastebin my CLI logs |
| 10:06.17 | adeeln | http://www.pastebin.ca/342468 |
| 10:06.45 | adeeln | i have a few of htem |
| 10:06.46 | adeeln | http://www.pastebin.ca/342363 |
| 10:07.08 | adeeln | http://www.pastebin.ca/342371 |
| 10:07.50 | BrokenNoze_ | Ah. have you set disallow=all |
| 10:07.53 | BrokenNoze_ | allow=alaw |
| 10:07.56 | BrokenNoze_ | allow-ulaw |
| 10:08.00 | adeeln | that's set in my sip.conf |
| 10:08.24 | adeeln | although, i have allow=ulaw, then allow=alaw |
| 10:08.55 | BrokenNoze_ | I had a problem where one of the devices I weas using didn't support a specific format |
| 10:09.36 | BrokenNoze_ | hence the other device could hear, while the failing device couldn't interpret the returned stream |
| 10:09.49 | adeeln | well, i've tried 3 seperate SIP software clients, 3 different SIP providers, and i'm trying an IAX client now |
| 10:09.56 | adeeln | all with no luck |
| 10:10.06 | BrokenNoze_ | Ah. |
| 10:10.35 | BrokenNoze_ | and your router def has firewall open for rtp |
| 10:10.37 | adeeln | i used a trunk that i have setup at home (stanaphone) and tested it on my home box, it worked |
| 10:10.49 | adeeln | then took it to the new box, after unregistering it from my home box, and no audio worked at all |
| 10:10.52 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 10:10.52 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 10:12.11 | BrokenNoze_ | Do internal calls work? |
| 10:12.28 | adeeln | yup, internal sip calls work no problems |
| 10:12.53 | BrokenNoze_ | then it's GOT to be a firewall issue. what ports do you have open? |
| 10:13.00 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 10:13.18 | adeeln | 10001-20000 |
| 10:13.24 | adeeln | 5004-5082 |
| 10:13.34 | adeeln | 4569,4570 |
| 10:14.15 | BrokenNoze_ | 5060? |
| 10:14.19 | BrokenNoze_ | UDP |
| 10:14.28 | BrokenNoze_ | Oh sorry, yeah |
| 10:14.41 | adeeln | it doesn't give me a choice to choose between udp or tcp |
| 10:14.44 | BrokenNoze_ | well my audio ports start at 10000 not 10001 |
| 10:15.09 | adeeln | 10000 is webmin's port |
| 10:15.58 | adeeln | i'm setting up an asterisk machine here locally, behind my nat to see if it's a software issue, or something with the connection |
| 10:16.00 | *** join/#asterisk A-Tuin|work (n=A-Tuin@212.41.185.81) |
| 10:16.37 | adeeln | i wonder if running gentoo has anything to do with it |
| 10:17.07 | BrokenNoze_ | how can it? internal calls are working |
| 10:17.30 | adeeln | that's what i thought |
| 10:17.41 | adeeln | but it's either a flipping config problem or god knows what |
| 10:17.46 | BrokenNoze_ | source of the data should be irrelivent. the problem your having is the data coming on from the NAT |
| 10:18.03 | adeeln | what do you mean? |
| 10:18.08 | *** join/#asterisk StyleWarz (i=stylewar@gayanalfisting.biz) |
| 10:18.14 | StyleWarz | Morning |
| 10:18.37 | BrokenNoze_ | Well from what your saying your only failing to get calls in from the outside world right? |
| 10:18.46 | BrokenNoze_ | via a SIP provider? |
| 10:18.52 | adeeln | yep |
| 10:19.01 | BrokenNoze_ | I assume its inbound audio thats failing |
| 10:19.28 | StyleWarz | I got a question belonging to Asterisk Realtime. For some tables i saw in examples like 'voicemail' i noticed that it has only fields for holding voicemail-user-records but no default settings which i have in the config |
| 10:19.48 | StyleWarz | So do i still have to set defaults in config files? |
| 10:19.52 | adeeln | BrokenNoze_: oddly enough, inbound works, it's the outbound audio |
| 10:20.09 | BrokenNoze_ | Ah. ok. |
| 10:20.10 | StyleWarz | Or at least, can i set defaults in configfiles and dynamic parts real-time? |
| 10:20.44 | BrokenNoze_ | thats a little more confusing! |
| 10:21.06 | BrokenNoze_ | is your server in the DMZ? have you tried it? |
| 10:21.19 | *** join/#asterisk saftsack (n=oliver@p54A7E6CE.dip.t-dialin.net) |
| 10:21.59 | adeeln | don't have a DMZ |
| 10:22.02 | *** join/#asterisk ivanfm (n=ivanfm@c93481ec.virtua.com.br) |
| 10:22.05 | adeeln | this router sucks my ass |
| 10:22.23 | StyleWarz | $this->router->suck(&$ass); ? :) |
| 10:22.29 | adeeln | haha |
| 10:24.22 | BrokenNoze_ | Well all I can suggest is turn firewall off on the router though i know you said you'd tried that. U do need UDP and TCP open too |
| 10:24.32 | saftsack | is it possible to let the polycom phones make a internal dialtone before 0 is pressed? |
| 10:24.34 | adeeln | yeah, i know |
| 10:25.20 | BrokenNoze_ | safesack: Hi, don't suppose you know haow to set up a Polycom for DTMF to work? can't work it out for life of me |
| 10:26.37 | saftsack | no i dont know but im at the manual here and i saw a dtmf tone generator chapter |
| 10:26.42 | BrokenNoze_ | saftsack: and I don't quite know what you mean by your question |
| 10:27.05 | saftsack | if you raise your handset you hear a tone. and if you press a 0 then you will hear another dialtone |
| 10:27.28 | BrokenNoze_ | Nope, not on mine. |
| 10:27.51 | BrokenNoze_ | but I hear a dialtone, soon as I hit 0 it goes silent |
| 10:28.13 | BrokenNoze_ | not a dialplan thing you've got setup? |
| 10:28.41 | saftsack | dont know thats the reason why i am at the manual. i just wanted to look if there are some guys here which did this the last few days |
| 10:30.09 | Chris-NB | adeeln, have you tried to tcpdump on the server to see where the rtp streams are going to? or, if the rtp stream from your sip client does not reach the server? |
| 10:30.41 | Chris-NB | adeeln, tcpdump -s0 -w /file/to/dump <- make dump, place call, hangup, stop tcpdump and open dump file with wireshark |
| 10:32.10 | adeeln | Chris-NB: i've tried the tcpdump and the logs seem to be showing data going both ways |
| 10:32.27 | adeeln | there was time where some of the packets weren't going both ways, but |
| 10:33.26 | Chris-NB | adeeln, strange!so your rtp data from your sip client reaches the server? |
| 10:33.37 | Chris-NB | adeeln, and the server sends it ... to pstn? |
| 10:33.49 | adeeln | yeah, in one test the data reached the server, and the server showed it forwarding to the SIP provider |
| 10:33.55 | adeeln | but there was no audio from me being heard |
| 10:34.34 | adeeln | there was another test where my data reached the * server, but the * didn't forward it |
| 10:34.46 | Chris-NB | adeeln, is there a firewall between the server and ... internet/your provider? |
| 10:35.04 | Chris-NB | adeeln, didn't forwarded? strange behaviour |
| 10:35.04 | adeeln | not that i know of |
| 10:35.16 | Chris-NB | k, pastebin your sip.conf |
| 10:35.23 | adeeln | sure |
| 10:35.36 | adeeln | well, i'm using freepbx to manage...so it's split the sip.conf |
| 10:35.49 | adeeln | Chris-NB: if you scroll up, i've pastebinned a few of my logs |
| 10:36.02 | Chris-NB | adeeln, I've ssen that. but not sip.conf or? |
| 10:36.23 | Chris-NB | adeeln, have you set realm and externip for your box? |
| 10:37.47 | adeeln | let me pastebin the sip.conf |
| 10:38.22 | Chris-NB | k |
| 10:38.48 | adeeln | http://www.pastebin.ca/342515 |
| 10:39.51 | Chris-NB | tos=0x68 <-- does nothing if asterisk is not running as root |
| 10:40.15 | Chris-NB | have you set externalip and realm? |
| 10:40.27 | adeeln | in which file? |
| 10:40.49 | adeeln | asterisk is running as asterisk:asterisk....so the tos is useless? |
| 10:40.54 | Chris-NB | jep |
| 10:41.04 | Chris-NB | cause u need to be root to set tos values |
| 10:41.13 | Chris-NB | I do it with iptables |
| 10:41.19 | adeeln | ah....okay...do i need to have a sip_nat.conf ? |
| 10:41.35 | adeeln | the * box is not behind a router afaik |
| 10:41.48 | *** join/#asterisk jarco (i=hardcode@177.Red-217-125-128.staticIP.rima-tde.net) |
| 10:41.56 | Chris-NB | I don't know freepbx so I don't know what parameters are in which files |
| 10:42.19 | Chris-NB | I've added nat=yes to my sip client sections, not in the general section |
| 10:42.29 | jarco | hi (hola) |
| 10:42.36 | Chris-NB | externip and realm should be in one of your config files |
| 10:42.44 | Chris-NB | hola jarco |
| 10:43.24 | adeeln | let me grep it real quick |
| 10:44.04 | jarco | Chris-NB hola |
| 10:44.28 | Chris-NB | adeeln, k. |
| 10:45.04 | BrokenNoze_ | Anyone help with my polycom DTMF problem? |
| 10:46.21 | adeeln | what should the realm be? |
| 10:46.42 | Chris-NB | normaly, set it to your domain name |
| 10:46.49 | Chris-NB | or set it to asterisk |
| 10:47.05 | adeeln | ok |
| 10:47.15 | Chris-NB | you have to set it in the sip client as well |
| 10:47.28 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 10:47.55 | adeeln | YAY!!!!!!!!! |
| 10:47.58 | adeeln | it finally works |
| 10:48.09 | adeeln | only after 15 hours of struggling with this damn thing |
| 10:48.24 | *** join/#asterisk apardo (n=apardo@87.217.145.218) |
| 10:49.55 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
| 10:49.55 | *** mode/#asterisk [+o angler] by ChanServ |
| 10:52.22 | adeeln | interesting...now when i call the DID associated with * box, i get a busy signal instead of it going through |
| 10:54.09 | Chris-NB | adeeln, what did you change to made it work? |
| 10:54.43 | adeeln | i added the externip, nat=yes |
| 10:54.49 | Chris-NB | fine : D |
| 10:54.50 | adeeln | and realm for good measure |
| 10:54.54 | Chris-NB | k |
| 10:55.11 | adeeln | but now i can't get it to accept my inbound calls associated with * box |
| 10:55.16 | adeeln | it just gives me a busy signal |
| 10:55.24 | Chris-NB | what says the CLI ? |
| 10:55.30 | adeeln | checking that now |
| 10:56.14 | adeeln | http://www.pastebin.ca/342527 |
| 10:56.27 | evgeni | Have anybody connected as5300-radius-asterisk ? |
| 10:59.12 | Chris-NB | adeeln, I'll look at it after lunch : D |
| 10:59.36 | adeeln | alrighty |
| 11:01.50 | adeeln | i keep seeing this, and i don't know what it means, or how to get rid of it |
| 11:01.53 | adeeln | <-- SIP read from 4.78.224.161:5060: |
| 11:01.53 | adeeln | SIP/2.0 409 Conflict |
| 11:01.53 | adeeln | Via: SIP/2.0/UDP 4.78.224.161:5060;branch=z9hG4bK6f6be117;rport |
| 11:01.53 | adeeln | From: <sip:9048664201@192.168.67.2>;tag=as2c7cb12d |
| 11:01.54 | adeeln | To: <sip:9048664201@192.168.67.2> |
| 11:01.54 | adeeln | Call-ID: 13acb2a04c5d46e00166553205913f1c@127.0.0.1 |
| 11:01.56 | adeeln | CSeq: 104 REGISTER |
| 11:01.58 | adeeln | Content-Length: 0 |
| 11:03.38 | *** join/#asterisk yassine (n=yassine@dsl.voicint.com) |
| 11:10.52 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 11:10.52 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 11:11.39 | *** join/#asterisk Dibbler_XP_ (n=Dibbler@host217-45-198-229.in-addr.btopenworld.com) |
| 11:11.46 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 11:13.00 | *** join/#asterisk Tebi (n=rantis@gw.aller.fi) |
| 11:13.25 | zoa | yo yo attila |
| 11:14.41 | ruied | If I place a call from the internal phone (fxs channel or sip account) asterisk picks the call imediately, but if it is from the outside operator, asterisk picks after two or three rings... |
| 11:16.33 | *** join/#asterisk kelch (n=chatzill@124.217.71.116) |
| 11:17.44 | kelch | who here have successfully integrated 1.4 with cacti? |
| 11:18.05 | kelch | documentation on res_snmp is scarce |
| 11:21.30 | *** join/#asterisk penguinFunk (n=penguin@87.224.86.46) |
| 11:22.07 | adeeln | so i now have a new problem....i can get the full duplex audio...and i can call outbound...but i can't get any inbound calls to actually be accepted...the CLI just says 'congestion' and gives me a busy signal |
| 11:22.26 | adeeln | all the while, this was working when the other wasn't |
| 11:32.50 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 11:34.39 | kelch | adeeln, have you reviewed your contexts? |
| 11:36.25 | adeeln | yeah, i did |
| 11:36.36 | adeeln | once i deleted all the extensions, and remade them, it worked |
| 11:40.00 | Chris-NB | adeeln, now it's working? |
| 11:40.57 | jarco | ZT_CHANCONFIG failed on channel 1: No such device or address (6) |
| 11:41.37 | jarco | this is with digital:/etc# ztcfg -vv |
| 11:41.48 | jarco | x100p clone :( |
| 11:43.20 | *** join/#asterisk coppice (n=chatzill@55.157.17.210.dyn.pacific.net.hk) |
| 11:46.34 | *** join/#asterisk drako (n=ljd@unaffiliated/luisjose) |
| 11:46.40 | Chris-NB | jarco, cat /proc/zaptel/* |
| 11:46.46 | Chris-NB | jarco, are there any spans? |
| 11:47.16 | jarco | Charles[NS] ok one moment |
| 11:47.27 | jarco | im newbie sorry :( |
| 11:47.43 | jarco | digital:/etc# cat /proc/zaptel/* |
| 11:47.44 | jarco | cat: /proc/zaptel/*: No existe el fichero o el directorio |
| 11:47.45 | Chris-NB | jarco, np. |
| 11:47.59 | jarco | is a debian sarge |
| 11:48.05 | Chris-NB | ok, then make lsomd and grep for zap |
| 11:48.09 | jarco | asterisk and zaptel 1.2 |
| 11:48.15 | Chris-NB | what card are u using? |
| 11:48.38 | jarco | the card with chip motorola |
| 11:48.42 | jarco | x100p |
| 11:48.53 | jarco | digital:/etc# dmesg | grep -i zap |
| 11:48.53 | jarco | Zapata Telephony Interface Registered on major 196 |
| 11:48.53 | jarco | Zaptel Version: SVN-branch-1.2-r2086 |
| 11:48.54 | jarco | Zaptel Echo Canceller: KB1 |
| 11:49.24 | Chris-NB | lsmod | grep -i zap |
| 11:49.44 | jarco | digital:/etc# lsmod | grep -i zap |
| 11:49.44 | jarco | zaptel 179748 1 wcfxo |
| 11:49.45 | jarco | crc_ccitt 2432 1 zaptel |
| 11:50.06 | Chris-NB | have you configured /etc/zaptel.conf to match your config? |
| 11:50.22 | jarco | fxsks=1 |
| 11:50.23 | jarco | #loadzone=es |
| 11:50.23 | jarco | #defaultzone=es |
| 11:50.35 | kelch | exit |
| 11:50.45 | *** part/#asterisk kelch (n=chatzill@124.217.71.116) |
| 11:50.48 | jarco | im from spain (europe) |
| 11:51.01 | drako | this is kind of survey, which Linux distro do you guys use for asterisk? |
| 11:51.02 | Chris-NB | jarco, thought that : D (es) |
| 11:51.11 | Chris-NB | drako, debian |
| 11:51.21 | drako | Chris-NB, sarge or etch? |
| 11:51.27 | Chris-NB | drako, both |
| 11:51.31 | jarco | Chris-NB irqs or bios ? |
| 11:52.10 | Chris-NB | drako, older sys are sarge, newer ones etch (cause of driver problems -> broadcom netxtrem, e1000) |
| 11:52.19 | Chris-NB | jarco, what do you mean with that? |
| 11:52.20 | jarco | Chris-NB i compile zaptel only |
| 11:52.27 | jarco | i didnt make udevs |
| 11:52.37 | drako | Chris-NB, asterisk from source or packages? |
| 11:52.56 | Chris-NB | drako, both : D older from source, newer from packages (etch) |
| 11:53.09 | jarco | Chris-NB i have read in tfot asterisk about irqs problems |
| 11:53.21 | jarco | (sorry for my poor english) |
| 11:53.36 | drako | jarco, which card are you using? |
| 11:53.48 | jarco | drako x100p clone |
| 11:53.51 | jarco | motorola chip |
| 11:54.22 | Chris-NB | jarco, I think you either didn't configured /etc/zaptel.conf correct or didn't load the correct modules |
| 11:54.23 | Chris-NB | jarco, lspci shows the card? |
| 11:54.49 | jarco | 0000:00:09.0 Network controller: RaLink Ralink RT2500 802.11 Cardbus Reference Card (rev 01) |
| 11:55.05 | jarco | is this ? |
| 11:55.17 | Chris-NB | nop, don't think so |
| 11:55.28 | Chris-NB | 802.11 is WiFi |
| 11:55.40 | adeeln | finally |
| 11:55.44 | jarco | yes |
| 11:55.47 | adeeln | asterisk is working like it should |
| 11:55.48 | jarco | pues no sale |
| 11:56.01 | jarco | sorry, so it dont show :( |
| 11:56.10 | adeeln | for some reason, after i made the nat=yes change in my sip.conf file, my extensions got corrupted/won't working |
| 11:56.20 | Chris-NB | adeeln, what was missing? ... ok : / |
| 11:56.54 | adeeln | Chris-NB: well i pretty much just needed to add nat=yes...everything else turned out to be irrelevant...but after adding it, my extensions stopped receiving calls |
| 11:57.10 | adeeln | i could make outgoing calls, but nothing incoming was going through...and none of the messages were helpful |
| 11:58.00 | adeeln | so i just deleted all the ring groups, and extensions, and readded just 1, and things started working |
| 11:58.00 | Chris-NB | adeeln, strange. but now, when everyting is working find ... : D |
| 11:58.00 | adeeln | yeah, so i'm making myself a nice tarball image |
| 11:58.01 | Chris-NB | *hrhr |
| 11:58.14 | adeeln | next step is setting up a Certificate Authority & making my own certs |
| 11:58.35 | Chris-NB | with asterisk |
| 11:58.37 | Chris-NB | ? |
| 11:58.47 | adeeln | no, with openssl |
| 11:59.07 | adeeln | and then maybe figure out how to do implicit trusting/encrypting calls between * boxes |
| 11:59.08 | Chris-NB | jea, but using it with asterisk ? |
| 11:59.41 | Chris-NB | hehe, hf with it : / afaik is there very less support for SRTP/SIPS in asterisk |
| 12:00.13 | *** join/#asterisk HeinrichSA (n=hh@196.25.68.131) |
| 12:00.36 | adeeln | haha that's always the fun part |
| 12:00.43 | HeinrichSA | Hi all |
| 12:00.45 | Chris-NB | mhm |
| 12:00.45 | adeeln | i might consider switching to openpbx or sipx later on |
| 12:00.55 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 12:01.08 | adeeln | all i know right now is that i've been sitting in this chair for over 14 hours now |
| 12:01.21 | adeeln | trying to get the asterisk machine to work |
| 12:02.13 | Chris-NB | ok. I've set up two machines this morning : D |
| 12:02.33 | HeinrichSA | Does anyone know if you can use Asterisk as a voicemail system? I want to be able to forward unanswerd calls from my Mobile phone to my asterisk box? |
| 12:02.56 | adeeln | HeinrichSA: i don't think you can do that with * |
| 12:03.15 | adeeln | unless your mobile phone was an IP phone or somehow a client to the * |
| 12:03.57 | HeinrichSA | HI adeeln, thanx |
| 12:04.02 | *** join/#asterisk Ebola (n=Ebola@host81-151-91-139.range81-151.btcentralplus.com) |
| 12:04.12 | jarco | bye |
| 12:04.26 | adeeln | HeinrichSA: actually |
| 12:04.41 | adeeln | HeinrichSA: if your cell has a call forwarding feature, you could do that |
| 12:04.48 | HeinrichSA | Well i need a system like that... I want to forward voicemails to a different location |
| 12:05.11 | HeinrichSA | ok explain? |
| 12:05.46 | HeinrichSA | But it wont be only one person. It needs to be able to handle several mailboxes |
| 12:06.12 | adeeln | well, if your phone has a call forward feature, you can have it programmed to forward the call to your asterisk box |
| 12:06.33 | adeeln | and then you can have asterisk do CID matching to then route the call to a user extension which has a voicemailbox |
| 12:07.04 | adeeln | you only need 1 DID for the asterisk box, and a few minutes of thinking through the user setup...and a cell phone that can do the call forwarding |
| 12:07.07 | adeeln | most phones can |
| 12:07.40 | adeeln | my phone...the sony ericcson t639 can (it's about 4 years old now) |
| 12:08.04 | HeinrichSA | Yes i think allmost all phones can do that, but it's a network feature |
| 12:08.35 | HeinrichSA | The problem is how will i know what mailbox to use if i only have one DID? |
| 12:08.56 | adeeln | you set that up in asterisk...the inbound route rules will do that for you |
| 12:09.00 | HeinrichSA | I cant use CID as the same person could be phoning different people |
| 12:09.42 | adeeln | it all depends on how the call forwarding feature works |
| 12:10.07 | adeeln | if it's your cell phone calling another number, and then bridging the conversation so it's a 3 way...you can then just do CID matching for your users phones |
| 12:10.40 | adeeln | if it's something different, then, well, you can either do pin sets, or have extensions for each user (like the last 4 digits of the users cell # or something) |
| 12:10.52 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 12:10.52 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 12:11.20 | HeinrichSA | If i understand correcly with normal voicemail systems there is signalling that takes palce between the pbx and the voicemail system to indicate what mailbox it is |
| 12:11.30 | HeinrichSA | pin sets? |
| 12:11.43 | adeeln | people type in a predifined pin number or something |
| 12:12.13 | adeeln | HeinrichSA: the signalling will still transmit some kind of data, so depending on the data stream, you can figure something out |
| 12:12.15 | *** join/#asterisk merbzt (n=banan@136.240.13.217.in-addr.dgcsystems.net) |
| 12:12.50 | HeinrichSA | you mean I should devlop something to read the signalling? |
| 12:13.38 | adeeln | the signalling will probably be some standard...ss7 or SIP or something |
| 12:13.57 | adeeln | so you won't need to actually do any real development...maybe a little regex or pattern matching |
| 12:13.58 | HeinrichSA | I am more than willing to do that... |
| 12:14.15 | adeeln | you just need to get some more information on what the signalling is |
| 12:14.24 | HeinrichSA | Yes i am guessing ss7 but i dont know the first thing of ss7 |
| 12:14.44 | adeeln | how soon do you need this kind of system? |
| 12:14.59 | HeinrichSA | I still need to find a the best way to do this. |
| 12:15.16 | HeinrichSA | Well like with everying in life (these days) asap |
| 12:15.25 | adeeln | haha |
| 12:15.58 | HeinrichSA | yeah so i have a handful to digest |
| 12:16.24 | HeinrichSA | Have you done any devlopment in asterisk? |
| 12:16.28 | adeeln | it should be possible with asterisk...your best bet might be to get someone to do the development for you |
| 12:16.39 | adeeln | i work for a company that does development for asterisk based stuff |
| 12:16.54 | HeinrichSA | what company? |
| 12:16.58 | adeeln | www.equivox.net |
| 12:17.12 | HeinrichSA | so you should be able to do something like this? |
| 12:17.26 | adeeln | possibly...there should be a link on the website where you can ask for an RFP |
| 12:18.01 | adeeln | fill out the form and try to be pretty thorough in your requirements, and we'll get back to you in a day or two in how we think it can be done |
| 12:18.32 | HeinrichSA | is there anyone else i should talk to? |
| 12:18.46 | HeinrichSA | someone who knows allot about voicemail? |
| 12:19.03 | adeeln | i don't know of any other companies that do custom development for asterisk based stuff...i know of a few that do asterisk installs and all, but dunno about what your asking for |
| 12:19.19 | HeinrichSA | thanks i'll give it a try/ |
| 12:19.27 | adeeln | there are voicemail systems out there, but i don't think they'll be able to do what your looking for |
| 12:19.47 | *** join/#asterisk olsen (n=diego@200.61.236.33) |
| 12:21.02 | HeinrichSA | All the large network opperators have such voicemail systems. So someone should be able to help me |
| 12:21.45 | adeeln | HeinrichSA: i don't think they have it for their cell phones though |
| 12:23.10 | HeinrichSA | Isn't it a GSM network function? |
| 12:23.32 | *** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53) |
| 12:23.38 | HeinrichSA | So any divert/forward/voicemail function is done by the network switches? |
| 12:24.11 | HeinrichSA | I dont really know much about all this... this is why i am asking for help. |
| 12:24.46 | HeinrichSA | The phone itself does nothing? |
| 12:24.57 | adeeln | i'm not too sure either...but i'm pretty sure that the cell phone company charges you for 2 calls when you do call forwarding |
| 12:25.23 | *** join/#asterisk Ahrimanes (n=ma@81.7.159.2) |
| 12:25.38 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 12:25.40 | HeinrichSA | yes they do... but not for voicemail |
| 12:26.28 | adeeln | it might be carrier specific...i couldn't tell you off the top of my head...i'd have to do some testing/research |
| 12:26.36 | adeeln | but it should be possible |
| 12:26.54 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 12:26.56 | *** join/#asterisk whocares123 (n=asdas@wbs-41-208-216-205.wbs.co.za) |
| 12:27.05 | whocares123 | Hi all |
| 12:27.31 | whocares123 | I need some assistance with a digium FXO card installation in asterisk |
| 12:27.58 | whocares123 | Is there anybody that can help? please! |
| 12:28.26 | adeeln | whocares123: what part of installation? |
| 12:28.33 | *** join/#asterisk eject_ck (n=eject@ll-110.213.223.85.sovam.net.ua) |
| 12:28.37 | HeinrichSA | thanks adeeln... will find out more, please give me shout if you find anything |
| 12:29.04 | whocares123 | Getting asterisk to listen on the digium card (TDM02B) |
| 12:29.13 | adeeln | HeinrichSA: i'd definitely recommend filling out that RFP on www.equivox.net, we'll return a full proposol |
| 12:29.15 | eject_ck | can anybody recommend good Digital Interface Card |
| 12:29.33 | adeeln | whocares123: so you've compiled the module and done all the zaptel config? |
| 12:29.51 | whocares123 | yip and the zttool says all is ok |
| 12:30.11 | whocares123 | but when I try to load the module chan_zap.so I get a error |
| 12:30.19 | whocares123 | Error loading module 'chan_zap.so': libpri.so.1.0: cannot open shared object file: No such file or directory |
| 12:30.46 | adeeln | whocares123: you need to install the libpri library |
| 12:30.46 | whocares123 | but the asterisk ./configure command did pick it up |
| 12:31.15 | whocares123 | libpri is installed |
| 12:31.50 | adeeln | issue a 'find / -name libpri*' |
| 12:32.04 | adeeln | and see if there is a libpri.so.1.0 file |
| 12:32.49 | whocares123 | got file here /usr/lib/libpri.so.1.0 |
| 12:33.33 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 12:33.36 | adeeln | okay....try changing into the directory where chan_zap.so is (i think /lib/asterisk/blah) and do a 'ldd chan_zap.so' |
| 12:34.19 | whocares123 | ldd chan_zap.so |
| 12:34.20 | whocares123 | <PROTECTED> |
| 12:34.20 | whocares123 | <PROTECTED> |
| 12:34.20 | whocares123 | <PROTECTED> |
| 12:34.20 | whocares123 | <PROTECTED> |
| 12:34.35 | adeeln | so it's not finding your libpri.so.1.0 file |
| 12:35.07 | whocares123 | any way I can specify the path> |
| 12:35.53 | adeeln | usually you can just copy the missing library into /usr/lib or /lib or anywhere in your PATH environment variable |
| 12:37.07 | adeeln | try making a link in /lib to your libpri.so.1.0 file |
| 12:37.21 | adeeln | also, what are the permissions on your libpri.so.1.0 file? |
| 12:38.01 | *** join/#asterisk HarryR (n=Harry@host-83-146-53-46.bulldogdsl.com) |
| 12:39.04 | whocares123 | -rwxr-xr-x 1 root root 306510 Jan 31 14:30 libpri.so.1.0 |
| 12:39.56 | adeeln | odd...that should be fine |
| 12:40.08 | *** join/#asterisk Tebi (n=rantis@gw.aller.fi) |
| 12:40.55 | whocares123 | whats weird is when I reinstall asterisk from scratch the ./configure cmd yes it does find libpri |
| 12:43.12 | adeeln | try re-installing libpri, then zaptel, then asterisk |
| 12:44.02 | whocares123 | will try |
| 12:45.43 | adeeln | alright, well i'm out...good luck |
| 12:45.51 | *** join/#asterisk newbie1 (n=Main@203.208.196.140) |
| 12:47.52 | *** join/#asterisk littleball (n=littleba@bb220-255-152-63.singnet.com.sg) |
| 12:53.24 | whocares123 | thanks |
| 12:54.58 | *** join/#asterisk rbw_ (n=rowi@slave.rixport80.se) |
| 13:02.53 | *** join/#asterisk orkid (n=orkid@bas1-barrie18-1242375312.dsl.bell.ca) |
| 13:08.38 | newbie1 | Can I set Ring back tone in Asterisk? |
| 13:10.52 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 13:10.52 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 13:13.50 | newbie1 | Hello |
| 13:13.53 | newbie1 | Does asterisk support Ring back tone? |
| 13:14.51 | RoyK | newbie1: you can script that up |
| 13:15.02 | drako | anyone using asterisk with sugar? |
| 13:19.24 | drako | i can't figure how its going to dial through a webpage... |
| 13:20.11 | newbie1 | RoyK: Can you show me an example? |
| 13:23.09 | *** join/#asterisk DrukenHME (n=jdumais@CPE000854ddcdb1-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 13:23.27 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
| 13:23.27 | *** mode/#asterisk [+o angler] by ChanServ |
| 13:24.27 | RoyK | newbie1: Dial(SIP/something) and then GotoIf(${HANGUPCAUSE}=BUSY) |
| 13:24.32 | RoyK | erm. that may be numeric |
| 13:24.43 | drako | nv, |
| 13:24.57 | drako | nvm |
| 13:24.59 | RoyK | but anyway - check HANGUPCAUSE and go on after that |
| 13:25.01 | drako | i found out. |
| 13:29.17 | drako | Why my asterisk is not listening at 5038 port? |
| 13:31.47 | Corydon76-home | drako: because it's not set to listen in manager.conf |
| 13:31.56 | Corydon76-home | drako: you need to enable it |
| 13:32.48 | drako | Corydon76-home, yes thanks |
| 13:33.24 | drako | Corydon76-home, 2 more questins, they are simple, have you had any problem with TMD cards on Debian Etch? |
| 13:33.30 | *** join/#asterisk DonnieNLAPD (n=don@ip565044ea.direct-adsl.nl) |
| 13:33.38 | drako | erm |
| 13:33.38 | Corydon76-home | TDM? |
| 13:33.52 | drako | Digium TDM400 |
| 13:33.56 | Corydon76-home | I've never used Debian Etch |
| 13:34.26 | DonnieNLAPD | Hello all, can someone help me with the following problem: Transfering calls with Cisco 7940 results in ZOMBIE channels since Asterisk 1.4 |
| 13:34.31 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:34.42 | DonnieNLAPD | Other phones are working all ok |
| 13:34.45 | drako | Chris-NB, are you there? |
| 13:34.56 | Chris-NB | drako, jep. |
| 13:34.56 | drako | Corydon76-home, what about installing ARI? |
| 13:35.10 | drako | Chris-NB, have you ever have any problem with Etch and TDM cards? |
| 13:35.10 | Corydon76-home | drako: ??? |
| 13:35.54 | Chris-NB | drako, tdm400 I've never used them |
| 13:36.48 | drako | Chris-NB, which cards do you use? what about ARI (the recording system for voicemails etc) |
| 13:36.57 | *** join/#asterisk ctooley (n=ctooley@jc1-111.moment.net) |
| 13:37.30 | Chris-NB | drako, I'm using Sangoma (PRI) and Beronet (multiBRI) or hfc based cards (single BRI) |
| 13:39.25 | DonnieNLAPD | Someone else have problems with Zombie channels in the newest release of Asterisk? (1.4) both on Debian Sarge and Etch |
| 13:40.21 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-140-124.ks.ok.cox.net) |
| 13:41.12 | DonnieNLAPD | I have these problems with all the protocols (SIP IAX ZAP CAPI) |
| 13:41.33 | *** join/#asterisk ToyMan (n=Stuart@72.168.167.241) |
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| 14:00.47 | *** join/#asterisk orkid (n=orkid@bas1-barrie18-1242375312.dsl.bell.ca) |
| 14:03.34 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
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| 14:18.18 | anonymouz666 | recording a call with gsm format |
| 14:18.31 | anonymouz666 | 1 min will be what size? |
| 14:18.39 | *** join/#asterisk strav (n=strav@dsl-142-138.aei.ca) |
| 14:18.42 | anonymouz666 | 100k? |
| 14:20.27 | *** join/#asterisk RoyK (n=roy@213.160.242.90) |
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| 14:20.56 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:24.45 | strav | hi there. I've tried several hours yesterday to enable fax receiving through sip on asterisk, the only point I'm seeking to acheive now is to get the tif file created. For that, I created a directory with complete read/write access, installed spandsp, libtiff v. 3.7.1, along with a patched version of rxfax (and txfax) and recompiled asterisk. On my tests, I can see that rxfax is properly loaded (still, trying to enable debugging with | debug as a rxfa |
| 14:24.46 | strav | x argument never worked). Upon an incoming fax, the servers seems to properly run the dialplan context I assigned (answer, wait 3 secs, set timeout, rxfax to a test .tif file, etc). Problem is my .tif file is never created and I have no clue why... any suggestion? |
| 14:26.05 | strav | also note: I tried different sip service go get the best connection quality possible... |
| 14:26.28 | strav | any help would be much appreciated. |
| 14:29.13 | *** join/#asterisk gerhard7 (n=gerhard@82-171-117-191.dsl.ip.tiscali.nl) |
| 14:31.16 | tzafrir | strav, faxing ans SIP are not the best of friends. Did you try T.38 or standard fax? |
| 14:31.39 | tzafrir | strav, is it over LAN? |
| 14:34.00 | strav | nope. I've read about this instability about sip and faxing... Actually our box is not configured to answer to anything else then sip I'm afraid. Although, with the sip service I'm using now, several users reported that they did succeeded in receiving a fax... |
| 14:35.15 | *** join/#asterisk h0 (i=fakhir@unaffiliated/fakhir) |
| 14:37.04 | strav | say, not to hide anything, this is the first time I'm playing with asterisk. And well there's something I'm finding quite odd... The server is actually configured to answer anything as a fax, although, when I directly phone to it, I do not hear any fax tone or anything similar.... |
| 14:37.15 | *** join/#asterisk sasch (n=sasch@host102-30-static.107-82-b.business.telecomitalia.it) |
| 14:37.25 | *** join/#asterisk DaveCanoe (n=Dave@dhcp3022.nanog.merit.net) |
| 14:37.25 | sasch | hi all i have one problem with zaptel.conf |
| 14:37.38 | sasch | i live in italy and I want that asterisk speak italian ... |
| 14:37.43 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 14:37.45 | sasch | in my zaptel.conf i add |
| 14:37.51 | vvortex3 | no italiano! |
| 14:38.04 | sasch | defaultzone=it |
| 14:38.05 | sasch | loadzone=it |
| 14:38.34 | sasch | and i add this line in zaptel.conf |
| 14:38.50 | DonnieNLAPD | sasch] isn't that used for busy tones, dial tones etc? like the indications.conf file? |
| 14:38.52 | *** join/#asterisk littleball (n=littleba@bb220-255-152-63.singnet.com.sg) |
| 14:39.02 | sasch | but when i recevied a call asterisk speak english (i make /it/ sound in /var/lib/asterisk/sound) |
| 14:39.29 | shido6 | so set the language in the dialplan. |
| 14:39.40 | strav | I've also read about a module called something like nvfax used for fax detection, can it be necessary to implement it even if my actual settings are to answer anything as a fax? |
| 14:39.41 | shido6 | do you have other language sound files? |
| 14:40.08 | sasch | i have italy sound |
| 14:40.15 | shido6 | in what directory? |
| 14:40.15 | *** join/#asterisk jarg (n=jarg@200.56.225.61) |
| 14:40.44 | tzafrir | sasch, language= in zapata.conf |
| 14:41.11 | sasch | i add this line |
| 14:41.12 | sasch | [channels] |
| 14:41.12 | sasch | defaultzone=it |
| 14:41.12 | sasch | loadzone=it |
| 14:41.12 | sasch | language=it |
| 14:41.16 | sasch | but don't work |
| 14:41.19 | sasch | in zapata.conf |
| 14:41.31 | shido[NuFone] | what directory are you "italy" sounds in? |
| 14:41.46 | shido[NuFone] | your... |
| 14:41.49 | shido[NuFone] | damn "r" |
| 14:41.52 | shido[NuFone] | :) |
| 14:42.00 | tzafrir | sasch, wait, there is more: there is still the kerenl parameter opermode, and there is still progzone in zapata.conf . Both use different set of countries :-( |
| 14:42.34 | tzafrir | sasch, there is /etc/zaptel.conf and /etc/asterisk/zapata.conf . two different files |
| 14:42.37 | sasch | i have all sound in /var/lib/asterisk/sound/it |
| 14:42.52 | shido[NuFone] | excellent. |
| 14:42.53 | sasch | <tzafrir> I edit all |
| 14:43.03 | shido[NuFone] | do you make reference to "it" in your dialplan? |
| 14:44.18 | shido[NuFone] | ${LANGUAGE()}) , perhaps? |
| 14:44.39 | sasch | ?? |
| 14:44.49 | shido[NuFone] | http://www.voip-info.org/wiki-Asterisk+multi-language |
| 14:45.06 | shido[NuFone] | http://www.voip-info.org/wiki/view/Asterisk+sound+files+international |
| 14:45.29 | jeremy_g | there is iodbc and unixodbc then where the hell this res_odbc.conf come from? |
| 14:45.45 | DonnieNLAPD | Is there someone with some knowhow about my problem: see -> http://forums.digium.com/viewtopic.php?p=43220#43220 |
| 14:46.12 | shido6[NuFone] | ooooooh |
| 14:46.15 | shido6[NuFone] | zombies |
| 14:46.26 | slima | In ,,sip show peers'' command i have 'Unmonitored', what turn it on? |
| 14:46.33 | shido6[NuFone] | hehe |
| 14:46.35 | shido6[NuFone] | qualify |
| 14:46.36 | DonnieNLAPD | yeah, zombies suck :) |
| 14:46.46 | shido6[NuFone] | silma, "qualify" |
| 14:46.47 | sasch | this is my dial plan http://pastebin.com/876705 |
| 14:46.50 | *** join/#asterisk wunderkin (i=kev@ip72-208-3-221.ph.ph.cox.net) |
| 14:47.04 | shido6[NuFone] | look in...... sip.conf.samples there's a chart of what you can use in a peer stanza and a user stanza |
| 14:47.23 | slima | thx |
| 14:48.19 | sasch | because if i call with my sip phone asterisk speak italian and when i call my pstn (ZAP/1) asterisk speak english |
| 14:48.37 | shido6[NuFone] | I dont see any reference to "language" in that dialplan, sasch |
| 14:49.00 | jeremy_g | will some bastard answer my question too :D |
| 14:49.19 | DonnieNLAPD | wait in line jeremy ;) |
| 14:49.45 | shido6[NuFone] | unixodbc is what you want if u want to use that. |
| 14:49.55 | *** join/#asterisk mercestes (n=merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 14:49.59 | shido6[NuFone] | i think freetds and unixodbc can be used together |
| 14:50.12 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
| 14:50.12 | *** mode/#asterisk [+o angler] by ChanServ |
| 14:50.13 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 14:50.13 | DonnieNLAPD | yes that is possible |
| 14:50.13 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:50.19 | shido6[NuFone] | one of my comrades uses those for his weird billing system |
| 14:50.46 | jeremy_g | :) |
| 14:51.00 | *** part/#asterisk strav (n=strav@dsl-142-138.aei.ca) |
| 14:51.37 | DonnieNLAPD | shido6, you know something more about Zombies? in asterisk not in real life I mean ;) |
| 14:51.45 | shido6[NuFone] | yes. |
| 14:52.06 | shido6[NuFone] | a lot of the people I live with are Zombies. |
| 14:52.07 | jeremy_g | :) Thanks DonnieNLAPD and shido6[NuFone] |
| 14:53.37 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-72-65-23-44.bflony.east.verizon.net) |
| 14:54.01 | DonnieNLAPD | Well, watchout you don't become one |
| 14:56.17 | DonnieNLAPD | Cool title for a new movie: Shawn of the dead phone zombies :| |
| 14:58.36 | booray | good morning all |
| 14:59.06 | *** join/#asterisk oej (n=olle@c-24-5-132-169.hsd1.ca.comcast.net) |
| 14:59.21 | Nugget | I just read Stephen King's "Cell" which is about phone zombies. |
| 15:00.07 | booray | quick question to start the day -- would there be a good reason a Queue() would resubmit itself after a timeout rather than continue in the dialplan? |
| 15:01.13 | anonymouz666 | great when dialing from asterisk 2 to asterisk 1 sometimes I got this message: DEBUG[27513]: app_dial.c:1635 dial_exec_full: Exiting with DIALSTATUS=CONGESTION. and sometimes WORKS... only god knows why |
| 15:01.39 | DonnieNLAPD | I am sorry anonymouz666, even I don't know it |
| 15:01.51 | anonymouz666 | failed to auth on INVITE... |
| 15:02.02 | anonymouz666 | its strange because it WORKS sometimes |
| 15:03.39 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 15:04.21 | *** join/#asterisk ToyMan (n=Stuart@12.23.30.130) |
| 15:10.28 | shido6[NuFone] | heh |
| 15:10.52 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 15:10.52 | shido6[NuFone] | well sometimes isnt going to cut it |
| 15:10.52 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 15:11.04 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) |
| 15:11.07 | shido6[NuFone] | keep similar versions talking to each other :) |
| 15:11.18 | shido6[NuFone] | upgrade 1 or downgrade the other. |
| 15:12.03 | shido6[NuFone] | I sometimes get 911 |
| 15:12.36 | *** join/#asterisk in-pt (n=lokesh@estrela.nortenet.pt) |
| 15:12.36 | shido6[NuFone] | hows that for motivation :) |
| 15:12.56 | *** join/#asterisk JoNate (n=noone@mail.wmelec.com) |
| 15:14.11 | *** join/#asterisk funxion (n=nunya@63.214.236.169) |
| 15:17.50 | *** join/#asterisk saftsack (n=oliver@p54A7FE44.dip.t-dialin.net) |
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| 15:28.28 | b11d | . |
| 15:30.14 | JoNate | ? |
| 15:31.05 | b11d | yeah.. deal with it |
| 15:31.06 | b11d | :) |
| 15:31.12 | JoNate | ! |
| 15:33.33 | saftsack | has somebody a polycom telephone with a serverside telephonebook? |
| 15:34.19 | *** join/#asterisk marv[work] (n=timr@24.214.206.254) |
| 15:35.57 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 15:41.39 | StyleWarz | Anyone here using Asterisk Realtime? I want to know how to deal with default context. Can i leave the general stuff in my configfiles (like in voicemail.conf) and then just put the stuff from default in the db? |
| 15:42.12 | StyleWarz | I really don't get it. I only find shitty howto's like http://www.asteriskguru.com/tutorials/realtime_pgsql.html that only explain a little part like how to create the voicemail_users table but nothing how to set emailbody, charset and stuff |
| 15:43.03 | danp | that stuff is still done in voicemail.conf, yes |
| 15:43.24 | StyleWarz | great |
| 15:44.46 | StyleWarz | thanks danp :) |
| 15:45.05 | StyleWarz | danp: Is it still possible to hold those defaults in the database? |
| 15:45.35 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 15:45.50 | danp | only if you set them in each row |
| 15:46.02 | danp | there's no replacement for [general] type sections in the database |
| 15:46.07 | newbie1 | Anyone work with ring back tone? |
| 15:46.13 | danp | at least i don't think there is |
| 15:46.53 | jeremy_g | can i put macros in the db to be read by asterisk realtime (extconfig) |
| 15:46.54 | jeremy_g | ? |
| 15:47.00 | jeremy_g | does it support macros |
| 15:47.22 | *** join/#asterisk shinux__ (n=shinux@196.220.27.131) |
| 15:47.45 | jeremy_g | and complex expressions in extensions for extension app |
| 15:48.05 | jeremy_g | StyleWarz:bro, we are on the same ship |
| 15:48.15 | jeremy_g | :D |
| 15:48.23 | jeremy_g | StyleWarz:yeah that sucks, so much |
| 15:49.35 | *** join/#asterisk dlynes_laptop (n=dlynes@S0106001346f7843f.vc.shawcable.net) |
| 15:49.38 | jeremy_g | StyleWarz:i just said following a couple of minutes ago on -dev |
| 15:49.38 | jeremy_g | jeremy_g> damn |
| 15:49.38 | jeremy_g | <jeremy_g> i just discovered that the best documentation for asterisk lies within the doc files as README.damnthing-i-wanna-learn |
| 15:49.38 | jeremy_g | <jeremy_g> so to the point and concise, better than those bloody howtos that talk about the color of howto-author's undergarments, his favourite beer brand, etc. |
| 15:53.32 | StyleWarz | *ouch* :) |
| 15:53.44 | StyleWarz | Okay i have the docs emerge'd |
| 15:53.52 | StyleWarz | bbl |
| 15:55.14 | *** join/#asterisk Gr1ncheux (n=devine@unaffiliated/gr1ncheux) |
| 15:56.00 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
| 15:56.25 | ttrygve | does a lower bitrate codec like g.729 also help with high latency connections, or does it only help when there are bandwidth constraints causing quality problems? |
| 15:58.25 | *** join/#asterisk moprilo (n=felipe@196.40.45.12) |
| 15:58.33 | *** join/#asterisk cpatry (n=junky@modemcable105.205-56-74.mc.videotron.ca) |
| 16:01.27 | *** join/#asterisk toerkeium (i=oo@201.216.206.221) |
| 16:01.55 | saftsack | <PROTECTED> |
| 16:04.43 | zoa | ttrygve: only helf with bandwidth constraints |
| 16:05.15 | zoa | helf -> helps |
| 16:05.16 | anonymouz666 | In [general] I got context=sip-sip I also have a user [user1] with context=menu-sip... dialing from another box using user1 credentials this call with reach sip-sip or menu-sip context? I think that is sip-sip |
| 16:06.35 | *** join/#asterisk Flauto (n=zhao@ppp-68-251-53-224.dsl.chcgil.ameritech.net) |
| 16:06.47 | ttrygve | thanks zoa |
| 16:10.31 | *** join/#asterisk luchshiy (n=anonymou@d212-53-104-193.cust.tele2.ch) |
| 16:10.53 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 16:10.53 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 16:15.03 | *** join/#asterisk Vec (n=Vector@dsl-243-124-97.telkomadsl.co.za) |
| 16:15.35 | Vec | Anyone know of a way to link asterisk to oulook 2007 built in phone dialer ? |
| 16:15.55 | *** part/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
| 16:16.28 | zoa | Idefisk biz :p |
| 16:17.50 | jeremy_g | zoa:promoting ur stuff :P |
| 16:18.04 | jeremy_g | only 19.9$ :P Vec |
| 16:18.12 | jeremy_g | err Euro |
| 16:20.04 | *** join/#asterisk McLazarus (n=mcallist@pool-72-78-34-148.phlapa.east.verizon.net) |
| 16:21.09 | McLazarus | hi, anyway from the console to kick a "caller" out of a queue? in this case the is no actual active channel that I can see. If I do a show channels concise this channel doesn't show up. |
| 16:21.45 | McLazarus | also the caller seems to have been waiting for 1219:38 |
| 16:23.21 | Flauto | anyone is using asterisk with gtalk? |
| 16:23.42 | funxion | on a tdm400p how can I detect whether there is a valid line connected to it or not and autoroll to the next available channel? |
| 16:25.30 | Flauto | i edited gtalk.conf and jabber.conf but i dont' even see the account is registered with gtalk |
| 16:26.24 | Jason99 | I have a D-Link ATA connected to my * server and I need to make it ring with distinctive ring.. can't get it to go |
| 16:29.28 | StyleWarz | Any ideas where to find bristuff for 1.4? |
| 16:29.44 | mercestes | McLazarus: Maybve softhangup |
| 16:30.08 | McLazarus | mercestes: thanks, I'll give it a shot. |
| 16:31.33 | McLazarus | hmm not a known channel. |
| 16:31.43 | McLazarus | maybe I just need to bounce that Queue, the queue is out of realtime |
| 16:33.20 | *** join/#asterisk Crescendo (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
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| 16:41.17 | *** join/#asterisk [[blah]asfd (n=ckwall@63.149.122.93) |
| 16:41.56 | [[blah]asfd | looking for some help with a feature in voicemail. is there a way for someone to transfer a call directly to a persons voicemail without it ringing the phone for specified number of rings? |
| 16:42.38 | file | [[blah]asfd: are you using FreePBX, Trixbox, something like that? |
| 16:42.59 | [[blah]asfd | no. just asterisk |
| 16:43.17 | file | interesting... yes it's possible, just write dialplan logic that goes directly to someone's voicemail |
| 16:44.19 | [[blah]asfd | hmmm.... |
| 16:44.23 | [[blah]asfd | that could be way messy. |
| 16:44.39 | [[blah]asfd | i am picturing a second extension for every phone. i have about 200 |
| 16:45.19 | file | depending on how you map extensions to mailbox it could be done in 2 lines of dialplan logic |
| 16:48.15 | mercestes | [[blah]asfd: Just do a _*8X.,1,VoiceMail(${EXTEN:2} or something. Pretty simple. Anything beginning with "*8" goes to the person's voicemail. |
| 16:48.42 | mercestes | I wouldn't use *8 btw. Just as an example. |
| 16:48.43 | [[blah]asfd | yeah thats a good idea.. thanks |
| 16:48.49 | mercestes | do something random like 69. |
| 16:48.59 | [[blah]asfd | right |
| 16:49.50 | jeremy_g | StyleWarz;wow, you are gonna run realtime on 1.4 |
| 16:50.32 | jeremy_g | what would the table look like for asterisk real time with macros in extensions.conf |
| 16:51.02 | *** part/#asterisk [[blah]asfd (n=ckwall@63.149.122.93) |
| 16:51.23 | *** join/#asterisk luchshiy (n=anonymou@d212-53-104-193.cust.tele2.ch) |
| 16:53.34 | funxion | on a tdm400p how can I detect whether there is a valid line connected to it or not and autoroll to the next available channel? |
| 16:53.47 | jeremy_g | how do we define global variables in asterisk real time mysql table |
| 16:53.59 | jeremy_g | e.g. MYSUPEREXTEN=1234; |
| 16:54.19 | jeremy_g | where do i store it in the db |
| 16:54.19 | jeremy_g | ? |
| 16:54.41 | jeremy_g | answer it damn it! |
| 16:55.16 | jeremy_g | or do all of you read from files? |
| 16:56.40 | jeremy_g | :p |
| 16:56.43 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
| 16:56.43 | *** mode/#asterisk [+o angler] by ChanServ |
| 16:57.03 | *** join/#asterisk Assid (n=assid@202.88.132.238) |
| 16:57.08 | Assid | heya |
| 16:57.13 | defend | can some one look a a dialplan gotoif cmd for me real quick please it is not doing what i expect and i think i have a syntax screw up |
| 16:57.15 | defend | http://pastebin.ca/342899 |
| 16:57.21 | Assid | anyone here using voicepulse? |
| 16:57.27 | Assid | i cant seem to make a 1900 call |
| 16:58.38 | jeremy_g | what are you trying to do here |
| 16:59.15 | jeremy_g | oh ok |
| 16:59.57 | defend | i am trying to make it so that we can have multiple operator extentions for voice mail |
| 17:00.19 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 17:01.08 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 17:03.28 | jeremy_g | defend:using 10,1:20,1 |
| 17:03.40 | Assid | err.. anyone know if voipjet allows 1900 calls? |
| 17:03.41 | jeremy_g | i mean try adding the priority too |
| 17:03.54 | Assid | or any other service |
| 17:05.15 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 17:05.28 | *** join/#asterisk ucfMethod (n=ucfmetho@c2.efb7d1.client.atlantech.net) |
| 17:05.55 | *** join/#asterisk ctooley (n=ctooley@jc1-111.moment.net) |
| 17:05.56 | jeremy_g | nopes |
| 17:05.59 | jeremy_g | i gave wrong advice |
| 17:06.04 | ping2921 | How can I have a playback of who the callers is before answering a call? |
| 17:06.06 | *** join/#asterisk CrashSys (n=kumba@208.177.233.66.ptr.us.xo.net) |
| 17:06.25 | jeremy_g | exten => o,4,GotoIf($["${VMCONTEXT}" = "test"]?99:101) |
| 17:06.25 | jeremy_g | exten => o,99,dial(SIP/336) |
| 17:06.25 | jeremy_g | exten => o,100,dial(SIP/302) |
| 17:06.25 | jeremy_g | exten => o,101,Hangup |
| 17:06.28 | jeremy_g | try this |
| 17:06.34 | jeremy_g | i gotta go |
| 17:06.37 | jeremy_g | good luck |
| 17:07.41 | *** join/#asterisk anthonyl (n=anthonyl@outbound-chi.piperrudnick.com) |
| 17:08.13 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 17:10.51 | *** join/#asterisk tecnico (n=tecnico@24.96.146.69) |
| 17:10.51 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 17:10.52 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 17:10.53 | *** join/#asterisk russellb (i=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 17:10.53 | *** mode/#asterisk [+o russellb] by ChanServ |
| 17:11.37 | Assid | anyone suggest a good/decent outgoing provider? |
| 17:13.02 | sweeper | Assid: les.net has been pretty solid for me over the past couple of months |
| 17:13.03 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 17:13.19 | *** join/#asterisk UlbabraB (n=salama@host241-43-static.72-81-b.business.telecomitalia.it) |
| 17:13.24 | sweeper | and he's got a 3-letter domain, which is cool |
| 17:14.05 | Assid | haha |
| 17:16.20 | Assid | whats a good codec to use? |
| 17:16.30 | Qwell[] | Assid: any |
| 17:16.33 | Assid | for a bandwith thats limited to 768kbit (for 4 users) |
| 17:16.46 | Assid | these guys keep having issues and i cant figure out the problem |
| 17:17.15 | sweeper | Assid: ulaw is a good place to start |
| 17:17.41 | *** join/#asterisk anthonyl (n=anthonyl@outbound-chi.piperrudnick.com) |
| 17:17.54 | Assid | sweeper: it starts cracking away |
| 17:18.12 | Qwell[] | Assid: sounds like you need qos, or a smaller codec |
| 17:18.25 | Qwell[] | but then, even with a smaller codec, a big download could screw it up again |
| 17:18.56 | mafkees | heya |
| 17:18.58 | mafkees | :) |
| 17:19.09 | Assid | nah.. seperate line for voip |
| 17:19.26 | Qwell[] | then you have jitter |
| 17:19.41 | Qwell[] | 4 ulaw calls over 768k should be fine |
| 17:19.42 | *** join/#asterisk Xen^ (n=linux@unaffiliated/lnux/x-10290) |
| 17:20.37 | Assid | yeah.. but they just start cracking up.. if it goes to 5th/6th.. sometimes even 3 calls |
| 17:20.45 | mafkees | Qwell[]: 3 days without chan_sccp now .... |
| 17:20.49 | Assid | avg is 4 simultanous |
| 17:20.52 | Qwell[] | mafkees: congrats |
| 17:21.05 | mafkees | :) |
| 17:21.10 | mafkees | lots of issues tho |
| 17:21.19 | Qwell[] | mafkees: forking yet again was just not the right solution |
| 17:21.27 | mafkees | Qwell[]: I agree |
| 17:21.42 | mafkees | and sergio is gone |
| 17:21.43 | Qwell[] | I mean, somebody could've made a cool $1500 with just a couple days work of fixing chan_skinny |
| 17:21.56 | Qwell[] | then all parties involved would be sated |
| 17:21.59 | mafkees | no reply to mail nor fora |
| 17:21.59 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 17:22.17 | Qwell[] | Sergio is dead as far as I'm concerned. :) |
| 17:22.32 | mafkees | gheh, looks like it |
| 17:22.36 | mafkees | I cant reach him |
| 17:23.01 | mafkees | Qwell[]: what are you working on in chan_skinny right now ? |
| 17:23.13 | mafkees | dont want to solve stuff you are already fixing |
| 17:23.21 | Qwell[] | mafkees: not a whole lot, actually |
| 17:23.25 | mafkees | I have a couple of issues I really want to be solved |
| 17:23.30 | Qwell[] | haven't had much time lately |
| 17:23.43 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 17:23.47 | mafkees | incoming calls show no callerid |
| 17:23.47 | Qwell[] | I would like to do some of the SLA stuff when Russell is done with it |
| 17:23.58 | mafkees | speeddials are not working |
| 17:24.38 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
| 17:24.38 | *** mode/#asterisk [+o angler] by ChanServ |
| 17:24.43 | mafkees | those 2 are the main issues for my home setup |
| 17:24.57 | Qwell[] | mafkees: 7960's should be the most "solid" phone |
| 17:25.10 | mafkees | I have 1 7960 and 1 7905 |
| 17:25.19 | Qwell[] | ooo, nice |
| 17:25.27 | Qwell[] | I have... |
| 17:25.41 | Qwell[] | 7960, 12sp+, 30vip, and a 7910 I think |
| 17:25.46 | mafkees | ah |
| 17:25.55 | mafkees | you have callerid on the 7960 ? |
| 17:26.08 | Qwell[] | I don't recall... I haven't actually even unpacked my phones since I moved |
| 17:26.12 | *** join/#asterisk oej (n=olle@204.11.231.198.static.etheric.net) |
| 17:26.16 | mafkees | ah |
| 17:26.25 | Qwell[] | that was...3-4 months ago :P |
| 17:26.51 | Assid | Qwell: aany suggestions to better this? |
| 17:26.52 | anthony] | I use a Compaq presario laptop + xlite! |
| 17:26.54 | anthony] | Im cool. |
| 17:27.01 | file | Qwell[]: you have unpacked *some* though... right? |
| 17:27.09 | Qwell[] | file: computer and bed |
| 17:27.10 | Assid | and no any other decent provider |
| 17:27.26 | file | Qwell[]: makes sense |
| 17:28.01 | *** join/#asterisk punkgode (n=punkgode@rev-200-40-119-222.netgate.com.uy) |
| 17:28.54 | punkgode | hello, anyone knows if the h extension is supported in AEL ? I can't make it work with dialplans writen in AEL (v1) |
| 17:30.37 | mog | mitcheloc, ping |
| 17:32.18 | *** join/#asterisk shinux__ (n=shinux@196.220.26.214) |
| 17:33.24 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-154-54-119.red.bezeqint.net) |
| 17:34.18 | *** join/#asterisk olinux (n=olinux@starbucks.wellspublishing.net) |
| 17:34.59 | mafkees | Qwell[]: indeed no callerid on incoming calls |
| 17:35.27 | mafkees | heya tzafrir_laptop |
| 17:36.10 | Assid | okay seriously.. is vp having an issue |
| 17:36.15 | tzafrir_laptop | hi |
| 17:36.27 | *** join/#asterisk CrazyTux (n=CrazyTux@64.95.219.140) |
| 17:41.23 | mercestes | hi, tza |
| 17:42.00 | *** join/#asterisk Crescendo (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
| 17:45.42 | *** join/#asterisk orkid_ (n=orkid@dataq2.utias.utoronto.ca) |
| 17:47.18 | toot | hey folks - i keep getting chan_zap.c:3926 in zt_handle_event: Ring/Off-hook in strange state 7 on channel - any ideas welcome :) |
| 17:49.14 | *** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
| 17:49.24 | FuriousGeorge | hey all |
| 17:49.48 | *** join/#asterisk Taadow (i=JimmyJam@S0106000ea68f0431.va.shawcable.net) |
| 17:49.53 | mafkees | hhmm |
| 17:50.02 | mafkees | I think I need to read more sources |
| 17:51.47 | FuriousGeorge | if i'd forgotten to plug the molex into my tdm400p w/ fxs modules, would i be able to load the wctdm module, or start asterisk? would the channels show up in zap show channels? i have no dial tone (or voltimeter), but i dont think forgetting the molex is my problem |
| 17:52.15 | Taadow | I have, on average, 20,000 North American sip based minutes monthly I'm looking for termination on. 70% us, 30% canadian blend. We rarely exceed 5-6 concurrent channels but are expanding. Looking for a good rate, anyone? |
| 17:52.15 | mafkees | Qwell[]: what's the funcion if a new call comes in on skinny phone ? |
| 17:53.14 | FuriousGeorge | hmmm.... ive had an issue like this caused by uising apic on a uni-cpu 32-bit platform... maybe this is caused by kernel support for apic (though this is a dual-core opteron) |
| 17:53.51 | FuriousGeorge | anyone got any ideas? i need to swap a tdm but i dont have one |
| 17:53.58 | FuriousGeorge | err a spare one |
| 17:54.02 | FuriousGeorge | this was my spare |
| 17:56.36 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 17:58.06 | reber | hi |
| 17:58.17 | reber | do you know any scriptable softphones ? |
| 17:59.58 | *** join/#asterisk KuJaX (n=one@customtrading.dsl.xmission.com) |
| 18:00.35 | sweeper | reber: plz to be being more verbose |
| 18:00.51 | KuJaX | I currently have two VOIP accounts into my Asterisk box. I want to get a couple (atleast one) ANALOG POT lines into the box also. I have heard of digium PCI cards and also standalone ATA cards that work well. Does anyone here have any recommendations to convert the analog signal to digital signal for our Asterisk server? |
| 18:01.44 | Assid | Taadow: if you find a good provider lemme know |
| 18:02.58 | sweeper | Taadow: voxee.com has absolute cheapest minutes at 0.011, but is flaky. les.net is pretty stable, and charges 0.015. Junction networks is "enterprise |
| 18:03.06 | sweeper | " and charges 0.029 |
| 18:03.29 | Assid | les.net doesntg have the charges mentioned correctly |
| 18:03.38 | sweeper | Assid: sure it does |
| 18:03.43 | Assid | if you scroll down.. alot of US states are mentioned |
| 18:03.57 | sweeper | http://les.net/products/product_iptermnorth.php |
| 18:04.11 | reber | sweeper, i'd like a softphone where i could from a bash script, tell wich number to call. |
| 18:04.13 | sweeper | you're just looking at the a-z list... |
| 18:04.25 | Taadow | Thanks sweeper... much obliged for the info. |
| 18:04.26 | reber | sweeper, and the call starts automatically. |
| 18:04.50 | sweeper | reber: aha. sounds like you want a phone with autoanswer |
| 18:05.02 | sweeper | then you can script the calls from asterisk |
| 18:05.18 | sweeper | and connect to the phone, and the phone will pick up automagically |
| 18:05.51 | *** join/#asterisk J4k3 (i=jsuter@dhcp-12-197-128-58.intrastar.net) |
| 18:06.03 | Assid | sweeper: works prepaid ? |
| 18:06.12 | Assid | i wonder how les compares to voipjet |
| 18:06.22 | *** join/#asterisk codefreeze (n=steve_mu@216.166.159.235) |
| 18:06.24 | *** join/#asterisk anthonyl (n=anthonyl@outbound-chi.piperrudnick.com) |
| 18:06.50 | sweeper | Assid: yea, I use prepaid |
| 18:06.57 | reber | sweeper, not really, i just want a softphone (so from the client sofphone, not the asterisk server) to have a param wich could be a number. For example : thewantedsoftphone --call 5342. |
| 18:07.12 | sweeper | OH |
| 18:07.19 | sweeper | mmm |
| 18:08.04 | sweeper | no clue :) |
| 18:08.09 | *** join/#asterisk menfin (n=cray@LNeuilly-152-21-127-49.w193-253.abo.wanadoo.fr) |
| 18:08.16 | menfin | hello |
| 18:08.21 | sweeper | I guess you could check the manpages of the appropriate sotwares |
| 18:08.51 | orkid_ | how come DID's in canada are so much more than in the USA? |
| 18:08.58 | *** join/#asterisk Lann (i=Lanniste@adsl-63-200-88-82.dsl.scrm01.pacbell.net) |
| 18:09.03 | Lann | hey hey |
| 18:09.04 | Assid | infrastructural costs? |
| 18:09.06 | reber | yeap of course... I just wonder if anyone knows a softphone like that (with a param wich is a call number) |
| 18:09.18 | orkid_ | I think it's Bell Canada's monopoly |
| 18:09.29 | orkid_ | whatever's left of it perhaps |
| 18:09.31 | StyleWarz | Which of the G.729 Codecs seems to be "best"? |
| 18:09.47 | Assid | sweeper: they support 729 ?? |
| 18:10.01 | Lann | so im almost done with this damn diaplan based voicemail...i know how to pass variables from Asterisk AGI to PHP, but how can i pass something from php back to asterisk so that i can make Flite($string) ? |
| 18:10.52 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 18:10.52 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 18:10.56 | *** join/#asterisk orkid_ (n=orkid@dataq2.utias.utoronto.ca) |
| 18:11.28 | BrokenNoze_ | ChanSpy, anyone had problems using it on internal calls? |
| 18:12.08 | Lann | stop snooping your coworkers |
| 18:12.10 | Lann | :-p |
| 18:12.19 | BrokenNoze_ | lol, Na it's for a client |
| 18:12.23 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 18:12.28 | BrokenNoze_ | ambulance service |
| 18:12.44 | anthony] | same thing I said, until I got arrested ;P |
| 18:12.58 | Lann | grand theft asterisk? |
| 18:13.05 | BrokenNoze_ | works fine on inbound/ outbound. just not internal |
| 18:13.21 | BrokenNoze_ | Yeah. u have a good point |
| 18:13.43 | BrokenNoze_ | u think it's been left out on purpose? |
| 18:13.49 | Lann | everyone i know thats any good with network security has been arrested, wtf |
| 18:14.25 | Lann | a guy i used to know had a satellite pointed at a colo facility and was on their wifi lol |
| 18:18.23 | sweeper | .... |
| 18:18.26 | sweeper | really now? |
| 18:18.32 | BrokenNoze_ | anyone? |
| 18:18.53 | sweeper | and how did he get this satellite to recieve the 2.4ghz range, FROM SPACE? |
| 18:19.06 | *** join/#asterisk in-pt (n=lokesh@estrela.nortenet.pt) |
| 18:19.09 | Assid | sweeper: so do they support 729 ? |
| 18:19.15 | sweeper | Assid: they who? |
| 18:19.25 | Assid | les |
| 18:19.38 | Assid | and they got iax? |
| 18:20.24 | Assid | oh yeah they got iax |
| 18:20.25 | KuJaX | I currently have two VOIP accounts into my Asterisk box. I want to get a couple (atleast one) ANALOG POT lines into the box also. I have heard of digium PCI cards and also standalone ATA cards that work well. Does anyone here have any recommendations to convert the analog signal to digital signal for our Asterisk server? |
| 18:20.28 | orkid_ | a satellite dish i suppose, not a real satellite |
| 18:20.34 | zoa | reber, on linux or windows ? |
| 18:21.08 | sweeper | orkid_: yea, I like to nitpick, being a satellite tech :P |
| 18:21.14 | J4k3 | KuJaX: realize you've been asking this for days... You could have spent less than half the time at american minimum wage paying for BOTH devices by now. |
| 18:21.32 | StyleWarz | Is maxmsg in voicemail.conf in 1.2.14 still limited to 9999 ? |
| 18:21.32 | reber | zoa, ideally both, but linux in priority. |
| 18:21.38 | sweeper | Assid: they got g.711, GSM, and G.726 |
| 18:22.02 | zoa | idefisk biz does it on windows through some API (And command line parameters) for linux im unsure |
| 18:23.03 | Assid | hrmm k |
| 18:24.36 | KuJaX | J4k3 - BOTH devices? |
| 18:24.44 | Jason99 | I'm trying to use RealTime asterisk.. I've added the lines to extconfig.conf and res_mysql.conf but when I do a 'sip show peers' it doesn't show any entries... |
| 18:25.03 | KuJaX | J4k3 - I haven't got an answer from anyone yet. |
| 18:25.17 | KuJaX | or atleast common answer. |
| 18:25.43 | *** join/#asterisk lowlevel (n=Stuart@CPE00145176d140-CM000f9f1e356e.cpe.net.cable.rogers.com) |
| 18:25.48 | orkid_ | yeah seems there are many similarities between the communities in sattech and telephony, and that;s not a comliment. |
| 18:25.56 | orkid_ | compliment |
| 18:26.06 | wunderkin | from polycom sip 2.1.0 release notes: • 14759: Added microbrowser support to the SoundPoint IP 501 platform • 14760: Added microbrowser support to the SoundPoint IP 430 platform !! |
| 18:29.22 | cpatry | wunderkin: neat. |
| 18:29.33 | *** join/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
| 18:29.34 | *** join/#asterisk IPmonger (n=ipmonger@c-68-84-208-206.hsd1.pa.comcast.net) |
| 18:29.59 | sweeper | KuJaX: get both, test, return the one you like least |
| 18:36.51 | KuJaX | what PCI and what ATA have you had experience with sweeper? |
| 18:40.02 | *** join/#asterisk sasch (n=sasch@host102-30-static.107-82-b.business.telecomitalia.it) |
| 18:40.31 | sasch | hi all for recompile asterisk ?? i unload kernel module ?? |
| 18:42.43 | sasch | <PROTECTED> |
| 18:42.53 | sasch | modprobe -r zaptel don't run |
| 18:42.59 | sasch | FATAL: Module zaptel is in use. |
| 18:43.03 | sasch | but asterisk don't work |
| 18:46.25 | *** join/#asterisk Obergandhi (n=Obergand@P57ca.p.pppool.de) |
| 18:47.00 | *** join/#asterisk FarrisG (n=lckirk@gateway.wiquest.com) |
| 18:47.28 | FarrisG | Can anyone help me figure out why I'm getting a lot of DTMF mis-reads with my sip trunks from bandwidth.com? |
| 18:48.52 | Obergandhi | kennt jmd nen sehr gutes deutsches howto über asterisk mit misdn inkl extensions.conf |
| 18:49.06 | file | FarrisG: and how is DTMF being sent? |
| 18:49.37 | Qwell[] | Obergandhi: nein...oder ja |
| 18:50.10 | Obergandhi | ? |
| 18:50.16 | Qwell[] | Obergandhi: das Google... |
| 18:50.25 | Qwell[] | ist gut |
| 18:50.36 | Obergandhi | jaja das kleine google beantwortet mir nicht meine fragen |
| 18:50.41 | Obergandhi | leider |
| 18:51.28 | Qwell[] | jbot: translate de en beantwortet |
| 18:51.38 | *** join/#asterisk PMantis (n=pmantis@cpe-69-207-130-14.rochester.res.rr.com) |
| 18:51.40 | Qwell[] | ah hah |
| 18:51.43 | Qwell[] | jbot: translate de en fragen |
| 18:52.10 | FarrisG | file: I'm really not sure. My itsp says rfc2833, but if I set my inbound context dtmf/dtmfmode to anything but "auto" my trunks will not work at all |
| 18:52.16 | PMantis | Is there was way in the dialplan (prefer no AGI) to tell if an agent is part of a queue or not? |
| 18:52.36 | *** join/#asterisk giasai68 (n=giasai@ip-240-130.sn2.eutelia.it) |
| 18:52.36 | Qwell[] | Obergandhi: das google ist Ihr Fruend |
| 18:52.39 | *** join/#asterisk pythos (i=lanebob@unaffiliated/pythos) |
| 18:53.03 | giasai68 | hello |
| 18:53.22 | giasai68 | can I to have a help regard asterisk setup? |
| 18:54.21 | PMantis | giasai68, Just ask your question, and if someone here knows the answer, they'll respond. |
| 18:55.09 | Obergandhi | also misdn erkennt die karten, ne misdn-init.conf habe ich. eine karte in nt - eine in te. ist ab dann für asteriskt nurnoch der port interessant oder auch die nummer der karte |
| 18:55.52 | giasai68 | PMantis, ok... thank you :) |
| 18:55.57 | Obergandhi | quasi die ausgabe von misdnportinfo Port1 ... Port2 ... |
| 18:56.14 | Obergandhi | ? |
| 18:56.22 | hardwire | what? |
| 18:57.16 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-154-54-119.red.bezeqint.net) |
| 18:58.36 | giasai68 | I have installed asterisk 1.4 with zaptel 1.4 in ubuntu server.... if I run asteriosk -vvvvfc and digit help I dont see zapata command... any reason? |
| 18:58.41 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 18:59.07 | Obergandhi | ne andere frage ist.. in der misdn.conf gebe ich der TE karte die MSNs der Telekom, die werden jedoch beim asterisk start angemeckert |
| 19:00.35 | *** join/#asterisk h0 (i=fakhir@unaffiliated/fakhir) |
| 19:00.37 | Jason99 | I'm trying to use RealTime asterisk.. I've added the lines to extconfig.conf and res_mysql.conf but when I do a 'sip show peers' it doesn't show any entries... |
| 19:02.03 | bkruse | isnt connecting to your db |
| 19:02.58 | file | if you are using realtime and you do not have caching turned on, or you have caching turned on but the data has not been fetched yet, then sip show peers will not show anything |
| 19:03.00 | pythos | Hi, I got the kernel/modules done: Module 0: Installed -- AUTO FXS/DPO Module 1: Installed -- AUTO FXS/DPO Module 2: Installed -- AUTO FXO (FCC mode) Module 3: Installed -- AUTO FXO (FCC mode) Found a Wildcard TDM: Wildcard TDM400P REV E/F (4 modules) |
| 19:03.03 | pythos | etc. |
| 19:03.20 | pythos | now what? |
| 19:05.23 | *** join/#asterisk anthony] (n=anthony@175.21.188.72.cfl.res.rr.com) |
| 19:05.48 | sasch | ok i have upgrade my asterisk |
| 19:05.55 | Jason99 | bkruse: how would i know if it's connecting? |
| 19:06.08 | pythos | where is there a good 'howto' for getting config done AFTER the basic stuff like modules/kernel ?? |
| 19:07.54 | giasai68 | I have digium TE205P card |
| 19:08.11 | sasch | i have language=it |
| 19:08.28 | sasch | if i want to copy sound i make /var/lib/asterisk/sound/it |
| 19:08.31 | KuJaX | What digium card can I purchase that is PCI that will convert two POT into the asterisk server? |
| 19:08.32 | sasch | if i want to copy sound i make /var/lib/asterisk/sounds/it |
| 19:08.36 | sasch | it's ok ?? |
| 19:10.34 | perd | kujax: FXO card |
| 19:10.52 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 19:10.53 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 19:11.05 | KuJaX | perd - any specific make/model |
| 19:11.43 | perd | wel,l if you only have two pots lines coming in to your system and you dont expect moer than 4 ever, i would say a TDM400P should take care of you |
| 19:12.02 | perd | wow i spell like a pro |
| 19:12.17 | Jason99 | file: I tried to kill asterisk and restart it and it still didnt load the peer from the mysql db.. |
| 19:12.17 | KuJaX | perd- Thank you. We aren't 100% positive going analog. Would it be more cost effective to get a stand alone ATA? If so, anything you have tested that works great? |
| 19:13.04 | file | is the peer being used by a phone... or something? |
| 19:13.27 | Jason99 | file: no, it's not |
| 19:13.44 | file | that's why |
| 19:14.01 | file | the data will only be fetched if the peer is used |
| 19:14.11 | file | ie: trying to register or attempted to dial |
| 19:14.11 | Jason99 | ah let me try |
| 19:14.16 | file | and now I leave for the airport |
| 19:14.18 | file | bbl |
| 19:14.31 | mafkees | have fun file |
| 19:14.44 | giasai68 | file: can help me? |
| 19:15.28 | mafkees | giasai68: what's the problem ? |
| 19:15.28 | pythos | where to find the 'easy step by step' basic howto? |
| 19:15.44 | mafkees | pythos: there are some books around |
| 19:16.16 | mafkees | pythos: look here: http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 19:17.15 | giasai68 | mafkees: thanks for your reply... can I contact you next 2 hours? |
| 19:17.22 | *** join/#asterisk qdk (n=qdk@0x535eae17.boanxx9.adsl-dhcp.tele.dk) |
| 19:17.33 | mafkees | you mean 2 hours from now ? |
| 19:17.35 | mafkees | dont know |
| 19:17.46 | mafkees | I think not |
| 19:18.08 | giasai68 | and tomorrow are u here? |
| 19:18.16 | mafkees | yeah, I will be |
| 19:18.25 | mafkees | and maybe later today |
| 19:18.28 | mafkees | no idea yet |
| 19:18.38 | *** join/#asterisk southtel (n=slester@75.138.69.109) |
| 19:18.53 | giasai68 | so I can contact you tomorrow if is possible |
| 19:18.58 | giasai68 | thanks |
| 19:19.39 | giasai68 | bye |
| 19:20.19 | SuPrSluG | is there a way to put a sound file into a meetme room? maybe like a through .call file |
| 19:21.01 | SuPrSluG | or through manager api |
| 19:25.32 | *** join/#asterisk zapp-branigan (n=zapp-bra@81-202-140-56.user.ono.com) |
| 19:28.49 | Obergandhi | i cant reach my asterisk under the numer i gave him in the misdn.conf [tecard]msns.. from extern. what do i need in de extensions.conf that a internal msn gets the call over the hfc card in nt mode |
| 19:30.54 | *** join/#asterisk topping (n=topping@h-67-100-91-18.snfccasy.covad.net) |
| 19:33.39 | *** join/#asterisk znoG_ (n=gs@97-228-126-200.fibertel.com.ar) |
| 19:36.51 | BrokenNoze_ | whats the syntax for #include in sip.conf? can I use it? |
| 19:39.59 | SuPrSluG | #include sip_nat.conf |
| 19:40.28 | SuPrSluG | that would include sip_nat.conf in sip.conf |
| 19:41.01 | *** join/#asterisk RoyK (n=roy@216-182-163.0506.adsl.tele2.no) |
| 19:42.46 | PMantis | Chanspy just kiled the whole Asterisk 1.4.0 process... if I spy over and over, I can reproduce. What to submit bug? |
| 19:42.55 | RoyK | wow. asterisk is stable after '0 * * * * asterisk -rx "restart now"' |
| 19:43.09 | PMantis | s/What to/where to/ |
| 19:43.22 | RoyK | PMantis: bugs.digium.com is a nice place to file bug reports |
| 19:43.43 | PMantis | I with the Asterisk full log showed more... but... |
| 19:43.46 | RoyK | PMantis: and if asterisk is built with -O0 and started with -g you may even have a change to create a good backtrace |
| 19:44.07 | RoyK | PMantis: a backtrace is usually needed to locate the bug quickly |
| 19:44.35 | PMantis | RoyK, Can you explain -O0 ? is that configure -O0 ? |
| 19:44.48 | RoyK | make dont-optimize |
| 19:44.50 | RoyK | iirc |
| 19:44.56 | RoyK | do a make clean first |
| 19:45.06 | CunningPike | There's a nice coincidence with someone called PMantis asking where to file asterisk bugs :D |
| 19:45.07 | RoyK | make dont-optimize also does a make install |
| 19:45.08 | PMantis | oh, sure... thanks. |
| 19:45.17 | RoyK | CunningPike: hehe |
| 19:45.30 | PMantis | CunningPike, heh. |
| 19:47.28 | PMantis | ROFL |
| 19:47.43 | PMantis | CunningPike, and the digium bugtracker is: Copyright © 2000 - 2005 Mantis Group |
| 19:48.05 | RoyK | PMantis: it's called mantis, yes |
| 19:49.29 | PMantis | Ahhh |
| 19:50.09 | PMantis | file, You are assigned the bug... I can confirm (#8922) |
| 19:50.49 | PMantis | Hmmm, except this is for 1.2.14 |
| 19:50.55 | *** join/#asterisk phearless (n=phear@host81-138-68-106.in-addr.btopenworld.com) |
| 19:53.04 | *** join/#asterisk Araluccl0 (n=ciccio@adsl-ull-230-7.46-151.net24.it) |
| 19:53.04 | RoyK | PMantis: never file a bug for anything except latest release. if you do, it'll be rejected. if you can't reproduce it with latest 'stable' version from svn, it'll also be rejected. and it's quite possible that you'll have to try svn trunk, possibly with a few patches. You know, Asterisk Is Stable, It Is Just Ruined By Configuration Errors And Bad Users |
| 19:53.04 | De_Mon | when I make outgoing calls its through SIP/bandwidth but when I get incoming calls for the same friend it comes in as Natalia, |
| 19:53.04 | De_Mon | We are still having a problem calling. We marked your case for call |
| 19:53.04 | De_Mon | back, use the directions below to reach us directly. |
| 19:53.07 | De_Mon | To reach your support engineer directly, your case has been setup for |
| 19:53.08 | De_Mon | call back support while it is open. |
| 19:53.11 | De_Mon | To place a call to your assigned support engineer: |
| 19:53.13 | De_Mon | 1. Dial 1-800-876-1422 -or- 1-863-877-4177 |
| 19:53.16 | De_Mon | 2. at the main menu press five (5) for Case Callback (NOTE: this option |
| 19:53.18 | De_Mon | is not listed on the menu) |
| 19:53.20 | PMantis | RoyK, You can run, but you can't hide! :) |
| 19:53.21 | De_Mon | 3. At the prompt, dial the last 6 digits of your case: |
| 19:53.23 | De_Mon | 800 176 |
| 19:53.26 | De_Mon | shit |
| 19:53.28 | De_Mon | SIP/4.68.250.148 |
| 19:53.41 | De_Mon | !!! when I make outgoing calls its through SIP/bandwidth but when I get incoming calls for the same friend it comes in as SIP/4.68.250.148 |
| 19:55.43 | PMantis | RoyK, Calling ChanSpy(SIP/1002|q), and it completely crashing Asterisk when the call disconnects is a bug. I can't see how it could be perceived as a config issue. |
| 19:56.16 | *** join/#asterisk test34 (n=test34@unaffiliated/test34) |
| 19:56.22 | PMantis | RoyK, I'm using 1.4.0, but the only bug I fould referencing ChanSpy is for 1.2.14 |
| 19:56.30 | RoyK | omis there a core file? |
| 19:56.58 | *** join/#asterisk znoG (n=gs@97-228-126-200.fibertel.com.ar) |
| 19:59.52 | BSDTech | so where does 1.4 stand with pacthing know issues |
| 19:59.56 | *** join/#asterisk infernix (n=nix@spirit.infernix.net) |
| 20:00.25 | ruied | I have a digium tdm400 and it's working, but I have a problem, for testing I have a n 'Incoming' context, and the first rule is to answear the call. When I call from the outside it rings three time before * answears the call, what could be wrong? |
| 20:02.45 | mog | its getting caller id |
| 20:02.48 | mog | if you dont want that |
| 20:02.51 | mog | you can turn it off |
| 20:02.58 | mog | and asterisk will answer right away |
| 20:04.11 | ruied | ah, the caller id is the remote phone number? |
| 20:05.42 | mog | right it has to read the caller id of the person calling you |
| 20:05.42 | mog | well it doesnt have to |
| 20:05.43 | mog | but it is |
| 20:05.44 | mog | in default configuration |
| 20:05.48 | mog | as most people like callerid |
| 20:06.10 | pythos | how to know what start protocol my telco is using? |
| 20:07.04 | mog | ask? |
| 20:07.12 | pythos | heh... |
| 20:07.38 | pythos | its a nortel dsm100 switch.. US, ma-bel unit from the 80's |
| 20:08.00 | ruied | thanks |
| 20:10.52 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 20:10.52 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 20:12.13 | ruied | so, if I have callerid on than it always takes a time to answer, one of the problems is while it doesn't answer if I have a Playback(voice) it will play the 'voice' from the middle to the end, like it was playng since I place the call (in the background while it is ringing) |
| 20:13.16 | ruied | I don't know if I was clear (I'm Portuguese)... |
| 20:13.33 | *** part/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
| 20:16.31 | mog | the best you can do is turn off caller id or add a wait that is = to 3 rings or something of the sort |
| 20:20.59 | ruied | mog: done! thanks |
| 20:22.22 | PupenoR | Does anybody have any idea what program might create tables with names starting with "stats_" in the asterisk database (mysql)? I can't find it. |
| 20:24.14 | *** join/#asterisk flying_Luck (n=melifaro@ppp85-140-139-214.pppoe.mtu-net.ru) |
| 20:25.38 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 20:26.11 | *** join/#asterisk dj-fu (n=deejay@203-167-190-166.dsl.clear.net.nz) |
| 20:26.40 | *** join/#asterisk duel (i=duel@mail.asaiatm.com) |
| 20:26.52 | dj-fu | hey uh |
| 20:26.56 | dj-fu | having some slightly major issues |
| 20:27.00 | dj-fu | I've got incoming calls working ok |
| 20:27.08 | dj-fu | but whenever I dial anything it apparently rings but doesn't? |
| 20:27.09 | dj-fu | http://rafb.net/p/ixvYk997.html |
| 20:27.16 | dj-fu | is me dialing my cellphone (which isn't ringing btw..) |
| 20:28.48 | *** join/#asterisk RoyK (n=roy@216-182-163.0506.adsl.tele2.no) |
| 20:29.20 | ruied | how can I route a call from my fxo to a sip phone, I've done with an Answer() followed by a DIal() but it's not very elegant, since * answears the phone and the custommer pays for that even if the sip person didn't answer. I would like to make some kind of bridging the fxo to an fxs or sip account... |
| 20:31.21 | *** join/#asterisk elriah (i=elriah@adsl-072-149-159-016.sip.bhm.bellsouth.net) |
| 20:31.27 | simplexio | ruied: exten => _20.,1,Dial(SIP/${EXTEN},20,t) |
| 20:31.38 | elriah | Hi all. Anyone have the latest polycom sip/bootrom? 2.1.x/3.3.x? |
| 20:31.48 | sevard | dj-fu: want to paste your dialplans? |
| 20:31.51 | simplexio | ruied: i dont know does that work with normal lines.. |
| 20:31.52 | dj-fu | sec |
| 20:32.07 | dj-fu | http://rafb.net/p/IpTGuk81.html |
| 20:32.19 | simplexio | ruied: attleast here we dont use answer() at all |
| 20:32.48 | ruied | simplexio, going to try |
| 20:33.06 | dj-fu | sevard, it appears I was trying to call out on the wrong zap |
| 20:33.07 | dj-fu | doh. |
| 20:33.39 | sevard | heh |
| 20:33.42 | sevard | :) |
| 20:33.55 | dj-fu | that's a point, the naming convention Zap/x refers to what order on my card? |
| 20:34.02 | dj-fu | is Zap/x only for fxo lines? |
| 20:34.20 | dj-fu | & also, do I have to setup a context for my outgoing fxo? |
| 20:34.28 | simplexio | anyone know gpl or open source call center programs for asterisk . something like real realtime call centers ? |
| 20:34.48 | sevard | I haven't used zap in well over 5 months, but iirc you group zap channels in zapata.conf and then it's Zap/group |
| 20:34.52 | *** join/#asterisk _dreams_ (n=dreams@128.Red-83-51-103.dynamicIP.rima-tde.net) |
| 20:35.02 | sevard | you already have a context for your outbound |
| 20:35.18 | dj-fu | right |
| 20:35.21 | *** join/#asterisk tim0123 (n=cash247@adsl-75-39-213-70.dsl.rcsntx.sbcglobal.net) |
| 20:35.24 | dj-fu | but in zapata.conf, do I have to define the fxo? |
| 20:35.35 | sevard | i would put them in seperate groups |
| 20:36.08 | tim0123 | How come when i use recording on queues it doenst combine recording |
| 20:36.43 | sevard | tim0123: you have to combine them yourself with something like sox, there's an example on the wiki. each channel is recorded as a seperate file |
| 20:36.54 | elriah | Anyone? Latest polycom firmware? |
| 20:37.11 | tim0123 | Do you have a link to that wiki artical |
| 20:37.20 | sevard | tim0123: yes, http://www.google.com |
| 20:38.08 | sevard | If you're having serious issues finding it, I'll get it for you. But, you need to learn how to search the wiki. |
| 20:38.11 | sevard | it's important |
| 20:40.21 | tim0123 | I agree |
| 20:41.33 | mercestes | D-Fender isn't it? Omg. |
| 20:41.53 | mercestes | s/it?/in?/ |
| 20:42.16 | J4k3 | jbot is silly. |
| 20:42.21 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 20:42.25 | perd | what is the preferred codec to use with a cisco sip phone if you're not worried about bandwidth, alaw? |
| 20:42.34 | wunderkin | s/jbot/J4k3/ |
| 20:45.15 | elriah | Wunderkin: Thanks! Can you resend (I was on the phone) |
| 20:45.21 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
| 20:45.59 | pythos | Hi, in the exapmle: [internal] |
| 20:46.02 | pythos | exten => 611,1,Answer( ) |
| 20:46.09 | pythos | exten => 611,2,Echo( ) |
| 20:46.21 | pythos | what is the "2" referencing? |
| 20:46.29 | Moobius | pythos: step number two. |
| 20:46.31 | sevard | perd: I'd use ulaw. |
| 20:46.38 | Moobius | the second thing that happens. |
| 20:46.44 | pythos | ok |
| 20:46.53 | perd | im havign a problem with garbled audio |
| 20:47.03 | pythos | well.. I take it that is supposed to go into the zapata.conf file? |
| 20:47.06 | perd | it plays fine but for some of the prompts in voicemail and stuff it will garble |
| 20:47.07 | dj-fu | ulaw unless you're in an alaw country |
| 20:47.29 | Moobius | pythos: no. that goes in extensions.conf |
| 20:47.44 | Lann | (set caller id as caller) exten => *,1,Dial(911) exten => *,2,Flite(Fuck the police. Fuck Fuck Fuck the Police) |
| 20:47.45 | pythos | OH! ok. I was getting a busy signal.. :-() |
| 20:47.48 | dj-fu | sevard, sorry I had to play with a printer, in zapata.conf, how do I put my incoming zap properly |
| 20:47.54 | perd | i had an irq conflict but i removed the card in question and i still have the issues |
| 20:48.00 | perd | maybe i need a diff sound set |
| 20:49.42 | J4k3 | hahaha |
| 20:49.51 | *** join/#asterisk angler_ (i=angler@nat/digium/x-40dfb96adba47371) |
| 20:49.57 | J4k3 | FuckThePolice<8005551212> |
| 20:50.02 | Qwell[] | angler_: You fail at teh internets |
| 20:50.08 | perd | haha |
| 20:50.18 | pythos | Moobius: hey, that tested OK |
| 20:50.28 | Qwell[] | ^ I rest my case |
| 20:50.28 | J4k3 | I should offer "accelerated" emergency services |
| 20:50.48 | J4k3 | ie - hired gun, waterhose, first aid kit.... all at one easy to remember extension! |
| 20:50.49 | elriah | wunderkin: thanks :) |
| 20:53.06 | pythos | Moobius: OK! both fxs lines have dialtone AND test echoback good! |
| 20:53.35 | dj-fu | can anyone please tell me if I need to define anything in zapata.conf for an outgoing fxo? |
| 20:53.52 | dj-fu | wait a sec i'm getting confused |
| 20:53.59 | dj-fu | is fxo the one that gives a dialtone or expects a dialtone? |
| 20:54.14 | pythos | the o reffures to the office |
| 20:54.19 | pythos | that is the telco office |
| 20:54.27 | pythos | fxs is you rinternal line |
| 20:54.37 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 20:54.50 | pythos | the O has dialtone from the telco |
| 20:54.58 | dj-fu | so if someone rings my phone number, it'll signal on the fxo? |
| 20:54.58 | pythos | the S is your station |
| 20:55.03 | dj-fu | ah |
| 20:55.06 | dj-fu | wrong way around, righto |
| 20:55.38 | J4k3 | you turn me right round baby, right round, like a record baby |
| 20:55.45 | pythos | but the config is really messy as the way you define the signaling LOOKS backward |
| 20:55.45 | dj-fu | now if I have two fxs's, I want to use one as incoming calls and one as outgoing, do I have to define them both in zapata.conf |
| 20:55.50 | J4k3 | yes, thats what I should use for onhold music... annoying enough for most people to just hang up |
| 20:58.38 | sweeper | herm |
| 20:59.09 | sweeper | http://pastebin.ca/343185 <-- why does this thing PERSIST in going to the default context, instead of where I tell it to? |
| 20:59.14 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 21:01.29 | anonymouz666 | I got the same problem here |
| 21:01.49 | anonymouz666 | is it going to [general] context? |
| 21:04.53 | *** join/#asterisk DaveCanoe (n=Dave@dhcp3022.nanog.merit.net) |
| 21:05.17 | *** join/#asterisk docelm0 (n=vircuser@c-68-32-61-251.hsd1.de.comcast.net) |
| 21:06.57 | pythos | Moobius: hey, the echotest on the FXO modules works too! The tdm400p is working! Now I wonder what to do next.. |
| 21:07.27 | dj-fu | http://rafb.net/p/DtSsSK11.html < this is my zapata.conf, I'm no having any luck configuring the 4th module |
| 21:07.39 | dj-fu | also on a tdm400p, which side is 1, which side is 4? |
| 21:07.46 | *** join/#asterisk h0 (i=fakhir@unaffiliated/fakhir) |
| 21:08.10 | *** join/#asterisk _Vile (n=vile@bc182112.bendcable.com) |
| 21:08.24 | perd | what the hell....somehow i busted my sip setup |
| 21:08.29 | perd | i get no audio from the server |
| 21:08.30 | pythos | 1 is nearest the screw |
| 21:08.39 | pythos | oh wait.. |
| 21:08.40 | perd | but sip to sip calls work fine |
| 21:09.02 | perd | your sips boggle me |
| 21:09.20 | PMantis | file, ping |
| 21:09.30 | dj-fu | right, so 3 and four are my fxo's. damnit, asterisk doesn't wanna reconfigure 4 |
| 21:09.41 | pythos | dj-fu: ya, first network port nearest the screw, and also first module port nearest the screw |
| 21:10.08 | ruied | what is the 'i and the 't' flag for? example: i,1,Playback) t,1,Playback() |
| 21:10.18 | dj-fu | any ideas http://rafb.net/p/qEc0Xh45.html |
| 21:10.23 | dj-fu | t is timeout I think |
| 21:10.56 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 21:10.56 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 21:11.06 | tim0123 | guys why am i get this error? |
| 21:11.06 | sweeper | hoi, what's the switchtype for siemens ewsd? |
| 21:11.09 | tim0123 | soxmix: error while loading shared libraries: libmp3lame.so.0: cannot open shared object file: No such file or directory |
| 21:14.12 | danp | ruied: http://www.voip-info.org/wiki/view/Asterisk+standard+extensions |
| 21:15.55 | ruied | danp, thanks |
| 21:16.12 | dj-fu | if I have three one fxs setup in my zapata.conf and two fxo's, if I am trying to dial out over the fxs, is it Dial(Zap/1/number) or does it depend on what port the fxs is on or what |
| 21:17.59 | tzafrir_laptop | dj-fu, depends |
| 21:18.49 | dj-fu | ok, so if that particular fxs is in port 3, would it be Zap/3/number? |
| 21:20.05 | *** join/#asterisk zuez (i=steve@66.103.132.86) |
| 21:20.22 | tzafrir_laptop | a number to an FXS port? |
| 21:20.34 | tzafrir_laptop | What do you have connected there? |
| 21:22.47 | *** join/#asterisk sharp (n=sharp@c-68-46-30-7.hsd1.pa.comcast.net) |
| 21:25.09 | *** join/#asterisk anthonyl (n=anthonyl@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 21:25.41 | dj-fu | tzafrir, a pstn line |
| 21:27.10 | dj-fu | tzafrir, i also have a pstn line plugged into port 4, but asterisk doesn't wanna reconfigure it when I put it in the zapata.conf |
| 21:27.21 | dj-fu | i'd like incoming calls to come in on port 3, outgoing to go out on port 4 |
| 21:27.28 | dj-fu | but it don't wanna work |
| 21:27.49 | *** join/#asterisk postel_ (n=jp@wikimedia/Postel) |
| 21:29.09 | *** join/#asterisk X-Rob (n=Rob@ppp214-210.static.internode.on.net) |
| 21:32.35 | *** join/#asterisk lude (i=dolsen@smoke.isprime.com) |
| 21:32.55 | lude | I'm having a weird issue with asterisk |
| 21:33.09 | lude | it seems to be dropping the caller id for calls to sip phones from a certain PRI |
| 21:33.24 | lude | asterisk itself gets the caller id, it will even pass it off to another PRI |
| 21:33.47 | lude | but when i pass it to sip phones |
| 21:33.57 | lude | it just drops it and says unknown |
| 21:36.28 | *** join/#asterisk s0lid (n=jlq@203.192.188.182) |
| 21:38.46 | ruied | after a s,3,dial(SIP/ruied,10) I made a Playback(vm-nobodyavail) but * cuts part of the sound, why is that? |
| 21:39.06 | mafkees | ruied: did you Answer() the line ? |
| 21:39.15 | mafkees | ruied: what codec are you using ? |
| 21:39.43 | ruied | didn't answer |
| 21:40.21 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 21:40.30 | ruied | mafkees, the codec, I have to check I'm using the default configuration (making my firs tests) |
| 21:40.39 | mafkees | ok |
| 21:40.52 | PMantis | Is there a command that allows me to query the network for the ip that my IAXy currently has? |
| 21:41.00 | mafkees | try to answer the line, put a wait(1) and after that do the playback |
| 21:41.25 | The_DoC^ | <PROTECTED> |
| 21:41.30 | *** part/#asterisk tim0123 (n=cash247@adsl-75-39-213-70.dsl.rcsntx.sbcglobal.net) |
| 21:41.41 | The_DoC^ | wooops |
| 21:41.51 | dj-fu | does anyone know what could cause asterisk to not be able to configure my FXO in port 4? |
| 21:42.01 | ruied | mafkees, codec is in the extensions.conf? |
| 21:42.07 | PMantis | dj-fu, bad or missing module? |
| 21:42.21 | *** join/#asterisk lba (n=lba@user-12lml5g.cable.mindspring.com) |
| 21:42.45 | The_DoC^ | irq confict? |
| 21:42.53 | mafkees | ruied: codec is in sip.conf |
| 21:43.09 | lba | I can't parse this line in extensions.conf exten => _4XX,n,GotoIf(${ChanIsAvail(EXTEN)} = CHANUNAVAIL ? ${Congestion} ) |
| 21:43.15 | *** join/#asterisk rer (n=rer@cpe.atm2-0-1081072.0x50a73b36.sdbnxx1.customer.tele.dk) |
| 21:43.17 | dj-fu | The_DoC^, how can I test if there is an irq conflict |
| 21:43.23 | lba | The idea is to goto congestion if the channel does not exist. |
| 21:43.27 | perd | `lspci -v |
| 21:43.31 | dj-fu | The_DoC^, all of the other ports work |
| 21:43.34 | perd | cat /proc/interrupts |
| 21:43.53 | dj-fu | wctdm has it's own interrupt |
| 21:43.58 | ruied | mafkees, forgot, I'm dialing from the operator to an fxo (digium tdm400) |
| 21:44.08 | mafkees | ah, so no g729 |
| 21:44.10 | mafkees | that's good |
| 21:44.30 | mafkees | ruied: add: |
| 21:44.35 | *** join/#asterisk xezz (n=xez@trust-it.gr) |
| 21:44.40 | mafkees | exten => s,4,Answer() |
| 21:44.51 | mafkees | exten => s,5,Wait(1) |
| 21:45.06 | mafkees | exten => s,6,Playback(vm-nobodyavail) |
| 21:45.08 | mafkees | try that |
| 21:45.18 | *** join/#asterisk [[blah]asfd (n=ckwall@63.149.122.93) |
| 21:45.40 | xezz | hello , when i try to run /etc/init.d/asterisk start i get this error : Starting Asterisk PBX: Unable to open pid file '/var/run/asterisk.pid': Permission denied asterisk. |
| 21:46.11 | mafkees | xezz: looks like a permission issue |
| 21:46.16 | mafkees | what user is running asterisk ? |
| 21:46.32 | *** join/#asterisk anthonyl (n=anthonyl@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 21:46.32 | ruied | mafkees, ok, but why do I need delays, I do not need for sip... is it a hardware problem picking the call ? |
| 21:46.53 | xezz | i run it as root |
| 21:47.01 | mafkees | ruied: I found I need it to let rtp settle down |
| 21:47.15 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 21:47.56 | ruied | hmm, ok |
| 21:48.35 | mafkees | dont know why |
| 21:48.37 | mafkees | but it worked here |
| 21:48.46 | s0lid | anyone here have donfigured te412p connected to dialogic cards? |
| 21:50.00 | xezz | makfees , how to see what user is running asterisk ? |
| 21:51.33 | [[blah]asfd | i am getting an error when installing asterisk. I got libpri and zaptel ok, but when I run make for asterisk 1.2 I get the following output: http://pastebin.com.uk/1007 I must be missing a package on my os... can anyone help me identify? |
| 21:52.23 | rer | looking for info on bad zttest results on te110 (99.975) - irq is fine APIC is fine hdparm -t makes no difference, same card works fine on another pc |
| 21:53.18 | *** join/#asterisk RoyK (n=roy@216-182-163.0506.adsl.tele2.no) |
| 21:54.03 | mafkees | xezz: look in the init script |
| 21:54.19 | mafkees | in debian there's this option to set the user and group |
| 21:54.24 | mafkees | -U I think |
| 22:01.08 | *** part/#asterisk [[blah]asfd (n=ckwall@63.149.122.93) |
| 22:04.00 | rer | I am having clicks in audio using a te110 I am unable to determine the cause, any help is apreciated |
| 22:04.15 | *** join/#asterisk lorinc (n=ang@caracas-1559.adsl.interware.hu) |
| 22:05.17 | Bobthehunter | for frenchies |
| 22:05.18 | Bobthehunter | . Finalement pour ton NAME DISPLAY, Group Telecom dit que tu dois envoyer tes informations en HEXADECIMAL |
| 22:07.00 | perd | anyone know why asterisk would stop sending audio to sip phones? i have canreinvite=no on and it passes through audio fine, when i dial in to the voicemail system or something though, i hear no audio |
| 22:07.25 | perd | sip debuggign appears to be fine |
| 22:07.42 | perd | but the server is not sending udp data back to the phone, it hears it from the phone fine |
| 22:07.55 | perd | but like i said, when i dial another sip phone the server sends me audio without a problem |
| 22:08.07 | perd | friggen irritating me :( it worked before i rebooted |
| 22:09.22 | *** join/#asterisk Obergandhi (n=Obergand@P57ca.p.pppool.de) |
| 22:10.37 | Obergandhi | ive problems with my misdn cfg .. 1 TE card and 1 NT card ... can sb help me?? |
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| 22:11.34 | mafkees | I'm off to bed |
| 22:11.37 | mafkees | bye |
| 22:11.43 | tzafrir_laptop | mafkees, |
| 22:11.50 | mafkees | yeah ? |
| 22:11.57 | tzafrir_laptop | you should use -U to set the user |
| 22:12.07 | tzafrir_laptop | not start-stop-daemon's options |
| 22:12.18 | tzafrir_laptop | (just a random comment) |
| 22:12.20 | mafkees | or run it as root ;) |
| 22:12.33 | tzafrir_laptop | you shouldn't ;-) |
| 22:12.42 | mafkees | indeed |
| 22:13.17 | mafkees | I use the /usr/local/src/asterisk/asterisk-trunk/contrib/init/rc.debian.asterisk as skeleton |
| 22:13.21 | tzafrir_laptop | Obergandhi, not sure I can, but be more specific. What *is* wrong? |
| 22:13.24 | mafkees | put the user and group stuff in there |
| 22:13.48 | tzafrir_laptop | mafkees, get the init.d script from the deb |
| 22:13.57 | tzafrir_laptop | It actually works |
| 22:14.27 | mafkees | the contrib one as well |
| 22:14.42 | mafkees | the etch package is up-to-date ? |
| 22:14.48 | *** part/#asterisk bkw_ (n=brian@adsl-70-142-59-250.dsl.tul2ok.sbcglobal.net) |
| 22:14.54 | tzafrir_laptop | basically, yes. 1.2.13 |
| 22:15.01 | mafkees | I have to be honest: I use debian but never used their * package |
| 22:15.06 | mafkees | always compiled from source |
| 22:15.07 | tzafrir_laptop | Sid has 1.2.14 |
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| 22:15.17 | mafkees | first because I needed bristuff |
| 22:15.19 | tzafrir_laptop | I should start my own repo soon, hopefully |
| 22:15.22 | Obergandhi | i have got a HFC card for the internal ISDN phones .. that works ... |
| 22:15.26 | mafkees | but I found out the deb is bristuff |
| 22:15.55 | tzafrir_laptop | there is a bristuffed and a non-bristuffed deb |
| 22:16.00 | Lann | gah that scientology thing on digg is pissing me off |
| 22:16.07 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 22:16.48 | Lann | the "church" of scientology is having a programmer imprisoned for saying he was going to aim a "tom cruis missle" at their compound |
| 22:16.52 | Lann | *redirects all dns requests for scientology sites to xenu wiki* |
| 22:16.57 | mafkees | tzafrir_laptop: I'll check them on a vm tomorrow |
| 22:17.14 | Obergandhi | and a fritzcard for the external line .. the problem is - everytime i call the specified msn i get an error in the asterisk command line |
| 22:17.41 | xezz | tzafrir, is there a how to install asterisk-bristuff rather than run ./install.sh from jughanims bristuffs patch ? |
| 22:17.59 | mafkees | xezz: in debian there is |
| 22:18.08 | mafkees | aptitude search asterisk |
| 22:18.11 | tzafrir_laptop | xezz, the Debian packages (etch ones, e.g) |
| 22:18.14 | mafkees | install the bristuffed ones |
| 22:19.00 | tzafrir_laptop | apt-get install asterisk asterisk-bristuff |
| 22:19.25 | mafkees | I'll test tomorrow |
| 22:19.31 | mafkees | should be no problem |
| 22:19.48 | mafkees | tzafrir_laptop: I dont see your name in the packages.debian.org page about asterisk |
| 22:19.52 | mafkees | I thought it was your work |
| 22:20.23 | tzafrir_laptop | I participate there... |
| 22:20.38 | tzafrir_laptop | lokk at e.g. zaptel |
| 22:21.09 | mafkees | uhhuh |
| 22:21.11 | mafkees | that I know |
| 22:21.48 | mafkees | genzaptelconf and the xorcom stuff |
| 22:25.01 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 22:25.14 | mafkees | but I never use them |
| 22:25.21 | mafkees | because all my installs are voip only |
| 22:25.29 | mafkees | no need for zaptel here |
| 22:26.01 | *** join/#asterisk Burgwork (n=corey@ubuntu/member/burgundavia) |
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| 22:26.58 | Burgwork | anybody know of a good service for PSTN termination in Canada? |
| 22:26.59 | Nugget | you still need ztdummy at a minimum for timing source if you want to do meetme() |
| 22:27.24 | mafkees | Nugget: yeah, but app_conference fixes that :) |
| 22:27.36 | Nugget | and there was much rejoicing in the non-linux world. :) |
| 22:27.36 | mafkees | some platforms dont have ztdummy support |
| 22:27.51 | mafkees | lol |
| 22:28.00 | *** join/#asterisk adeeln (i=adeel@c-24-7-133-237.hsd1.ca.comcast.net) |
| 22:29.50 | mafkees | well, I'm glad asterisk runs on more then just linux |
| 22:29.56 | adeeln | can someone tell me the what i'm doing wrong here? i want an extension (555) to dial out when i call into it...so i've added in the dial string box '9047300911#' but nothing seems to work...it just loops it back to the extension i called from |
| 22:31.42 | mafkees | well, I'm off to have fun with the lady |
| 22:41.06 | *** join/#asterisk nosbig (n=nosbig@rrcs-70-62-223-6.central.biz.rr.com) |
| 22:43.07 | nosbig | Anyone have any luck with IAX trunking? |
| 22:43.26 | putzz | IAX trunking? whats so hard about it? |
| 22:43.42 | nosbig | I have two Asterisk servers I am trying to trunk, and it just isn't working... |
| 22:43.52 | nosbig | The secondary server registers to the main server...\ |
| 22:43.52 | mercestes | The howto's are kinda cryptic with the user v/s peer verbage. |
| 22:44.21 | *** join/#asterisk Vec (n=Vector@dsl-243-124-97.telkomadsl.co.za) |
| 22:44.31 | mercestes | nosbig: I couldn't get it to work until I basically set up my servers verbatim, and changing "master" "slave" and "secret" as it applied to my specific application. |
| 22:44.57 | nosbig | But when I use the dialplan on the secondary box to call the other box, it gives the following error message: |
| 22:45.21 | nosbig | chan_iax2.c:6907 socket_read: Rejected connect attempt from XXX.XXX.XXX.XXX, who was trying to reach '32@people' |
| 22:45.28 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqu6.cable.mindspring.com) |
| 22:46.01 | nosbig | I use IAX2/hostname/32@people as my dial string the the dialplan. |
| 22:49.03 | nosbig | When I dial from the "user" server (secondary) to the "peer" (main), which machine provides the context? |
| 22:49.42 | mercestes | pretty sure it's the main server. |
| 22:50.35 | mogorman | both servers have timing source nosbig ? |
| 22:51.22 | mogorman | or do you just meen calling back and forth |
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| 22:52.37 | nosbig | That's another issue... For now, I just want to get a single IAX call to work between them... |
| 22:52.56 | nosbig | But I need to get the ztdummy module working on the secondary server which does not have a Digium card in it. |
| 22:53.09 | nosbig | But I am just talking about calling back and forth for now. |
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| 22:57.01 | nosbig | I should also point out that the error message was from the server. |
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| 22:59.30 | Obergandhi | ive problems with my misdn cfg .. 1 TE card and 1 NT card ... can sb help me?? |
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| 23:09.10 | [hC] | any astlinux users in the house? |
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| 23:13.51 | clyrrad | So does anyone here know how to enable CLI colors from the CLI with out having to stop asterisk and restart it? I have asked this a few times on previous days but no one seems to have an answer. Anyone know if this is possible? |
| 23:15.17 | *** part/#asterisk xezz (n=xez@trust-it.gr) |
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| 23:18.59 | elriah | Hi all. With SIP phones, specifically the Polycom Soundpoint series, what is UDPOnly vs DNSnaptr and which should I use? Gut instinct says use UDP for least latency but that would be a guess. |
| 23:22.12 | The_DoC^ | clyrrad : I take it you disabled ANSI colors in the config file? |
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| 23:29.01 | clyrrad | The_DoC^: no I have not disabled it |
| 23:29.09 | clyrrad | not on purpose anyway - where can I check that? |
| 23:30.49 | Moobius | lx |
| 23:30.55 | Moobius | whoops... |
| 23:32.04 | The_DoC^ | well if you didn't disable it then it should work |
| 23:33.00 | clyrrad | it used to work |
| 23:33.04 | clyrrad | its not now |
| 23:33.14 | clyrrad | not sure what's changed |
| 23:35.32 | anthony] | anyone looking for a dedicated server to run pbx's? |
| 23:35.37 | clyrrad | so is there a way to re-enable colors from the command line? |
| 23:35.47 | elriah | anthony: Sure? |
| 23:36.43 | *** join/#asterisk coppice (n=chatzill@55.157.17.210.dyn.pacific.net.hk) |
| 23:40.25 | [hC] | hmm. how can i convert wav to g729? |
| 23:42.02 | test34 | hc, http://voip-info.org/wiki/view/Convert+WAV+audio+files+for+use+in+Asterisk |
| 23:43.12 | [hC] | doh. only a windows program available so far. |
| 23:44.56 | dj-fu | what could cause this? http://rafb.net/p/JUJPrJ35.html |
| 23:45.02 | Moobius | converts a directory of *.wav to gsm: #!/bin/sh for a in *.wav; do sox "$a" -r 8000 -c1 "`echo $a|sed -e s/wav//`gsm" resample -ql; done |
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| 23:48.32 | dj-fu | Feb 7 12:48:10 NOTICE[32766]: app_dial.c:1056 dial_exec_full: Unable to create channel of type '' (cause 66 - Channel not implemented) |
| 23:48.34 | dj-fu | ;o |
| 23:49.03 | Moobius | dj-fu: i'm guessing it's the "/703" part... |
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| 23:55.18 | CunningPike | clyrrad: Did you disable the tty9 console? |
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