| 00:05.03 | *** join/#asterisk inv_arp[work] (i=junya@c-71-206-88-100.hsd1.fl.comcast.net) |
| 00:10.43 | *** join/#asterisk inv_arp[work] (n=junya@c-71-206-88-100.hsd1.fl.comcast.net) |
| 00:12.43 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
| 00:12.53 | *** part/#asterisk karmatronic (n=karmatro@84.77.183.224) |
| 00:20.37 | *** join/#asterisk Deciphan (n=Erick@c-24-7-130-249.hsd1.ca.comcast.net) |
| 00:21.35 | *** join/#asterisk inv_arp[work] (n=junya@c-71-206-88-100.hsd1.fl.comcast.net) |
| 00:27.41 | *** join/#asterisk Skarmeth (n=Skarmeth@201009090248.user.veloxzone.com.br) |
| 00:31.01 | *** join/#asterisk inv_arp[work] (n=junya@c-71-206-88-100.hsd1.fl.comcast.net) |
| 00:34.47 | *** join/#asterisk Globetrotter (n=eric@205.211.239.11) |
| 00:37.12 | *** join/#asterisk karmatronic (n=karmatro@84.77.183.224) |
| 00:38.43 | *** join/#asterisk inv_arp[work] (n=junya@c-71-206-88-100.hsd1.fl.comcast.net) |
| 00:45.04 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 00:46.44 | *** join/#asterisk inv_arp[work] (n=junya@c-71-206-88-100.hsd1.fl.comcast.net) |
| 00:52.02 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
| 00:56.48 | *** join/#asterisk burnproof (n=jsharryp@58.69.32.129) |
| 00:58.20 | *** join/#asterisk inv_arp[work] (n=junya@c-71-206-88-100.hsd1.fl.comcast.net) |
| 00:59.11 | *** part/#asterisk Ebola (n=Ebola@host81-152-204-157.range81-152.btcentralplus.com) |
| 00:59.24 | *** join/#asterisk h0 (n=fakhir@unaffiliated/fakhir) |
| 01:00.27 | *** join/#asterisk Nukemizer (n=Nuke@160.7.249.15) |
| 01:01.09 | *** join/#asterisk javo (n=nemitzj@c-67-190-212-182.hsd1.mn.comcast.net) |
| 01:05.12 | javo | I'm having kind of a strange issue with asterisk. When people call in over my SIP provider, it doesn't play the ringing sound for them. |
| 01:07.09 | *** join/#asterisk inv_arp[work] (n=junya@c-71-206-88-100.hsd1.fl.comcast.net) |
| 01:12.28 | *** join/#asterisk inv_arp[work] (i=junya@c-71-206-88-100.hsd1.fl.comcast.net) |
| 01:20.01 | *** join/#asterisk inv_arp[work] (n=junya@c-71-206-88-100.hsd1.fl.comcast.net) |
| 01:21.28 | *** join/#asterisk Pyrodogg (n=pyrodogg@pin-broadband-ws-496.dsl.redwing.net) |
| 01:23.32 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
| 01:24.56 | *** join/#asterisk tim0123 (n=cash247@adsl-75-39-213-70.dsl.rcsntx.sbcglobal.net) |
| 01:25.43 | *** join/#asterisk inv_arp[work] (n=junya@c-71-206-88-100.hsd1.fl.comcast.net) |
| 01:25.43 | tim0123 | Anyone had a problem with trixbox losing network connection out of no where |
| 01:26.06 | [TK]D-Fender | ~trixbox |
| 01:26.07 | jbot | rumour has it, trixbox is is NOT supported here! Trixbox is extensively modified with patches, custom configuration files, and so on. FreePBX (trixbox's GUI) includes giant creeping thing-like macros which invade your dialplan, making debugging and customization extremely difficult. People using it should join #trixbox or #freepbx for help. |
| 01:26.07 | perd | do you see media disconnect in your dmesg? |
| 01:26.16 | tim0123 | let me check |
| 01:28.38 | tim0123 | well ive also had the same prob with my asterisk box |
| 01:30.07 | [TK]D-Fender | tim0123 : eitehr way not an * problem. try asking in #linux though I suspect it'd more likely be a problem with your switch if 2 boxes have gone down. |
| 01:30.40 | tim0123 | I have a client that has like 30 simotaneous calls that come in and it never fails evvery day it seems the network fails cause the computer stays on |
| 01:31.12 | *** join/#asterisk inv_arp[work] (n=junya@c-71-206-88-100.hsd1.fl.comcast.net) |
| 01:31.35 | *** part/#asterisk Pyrodogg (n=pyrodogg@pin-broadband-ws-496.dsl.redwing.net) |
| 01:32.48 | *** join/#asterisk infernix (i=nix@spirit.infernix.net) |
| 01:32.59 | *** join/#asterisk TheCops (n=henri@206-248-136-187.dsl.teksavvy.com) |
| 01:33.08 | TheCops | Hi |
| 01:33.59 | TheCops | If i have a T1 as data service, can I split this T1 to get 8 line or something like that? |
| 01:34.12 | *** join/#asterisk RoyKa (n=roy@217-175-39.100710.adsl.tele2.no) |
| 01:35.20 | [TK]D-Fender | TheCops : You can split voice/data, sure |
| 01:35.54 | TheCops | [TK]D-Fender, I have bell canada as supplier of the T1, how it work? |
| 01:36.11 | TheCops | Do I have to get a special equipment or something else |
| 01:37.06 | *** join/#asterisk inv_arp[work] (n=junya@c-71-206-88-100.hsd1.fl.comcast.net) |
| 01:38.05 | [TK]D-Fender | TheCops : Its in the WIKI,a nd if I'm not mistaken, in the book. |
| 01:38.11 | TheCops | :) |
| 01:38.25 | [TK]D-Fender | TheCops : Just take in your T1 into a digital card and you'll get a virtual device for it |
| 01:42.30 | *** join/#asterisk inv_arp[work] (n=junya@c-71-206-88-100.hsd1.fl.comcast.net) |
| 01:48.08 | *** join/#asterisk inv_arp[work] (n=junya@c-71-206-88-100.hsd1.fl.comcast.net) |
| 01:48.54 | *** join/#asterisk DocHolliday (i=RogerRab@gateway/gpg-tor/key-0x0E4F6D6C) |
| 01:53.27 | *** join/#asterisk SwK (n=Silik0nJ@12-214-191-109.client.mchsi.com) |
| 01:53.34 | *** join/#asterisk oQPa (n=roque@198.Red-83-41-117.dynamicIP.rima-tde.net) |
| 01:53.45 | *** join/#asterisk inv_arp[work] (n=junya@c-71-206-88-100.hsd1.fl.comcast.net) |
| 01:55.22 | *** join/#asterisk DocHolliday (i=RogerRab@gateway/gpg-tor/key-0x0E4F6D6C) |
| 01:57.23 | *** join/#asterisk coppice (n=chatzill@129.168.17.210.dyn.pacific.net.hk) |
| 01:58.44 | *** join/#asterisk envoy (n=user@000-046-882.area3.spcsdns.net) |
| 02:01.22 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 02:04.37 | *** join/#asterisk inv_arp[work] (n=junya@c-71-206-88-100.hsd1.fl.comcast.net) |
| 02:04.40 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-153-99-104.red.bezeqint.net) |
| 02:07.55 | *** join/#asterisk predder (n=predder@203.220.55.70) |
| 02:12.27 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 02:13.46 | DocHolliday | how powerful does the hardware have to be to manage a network of 100-150 phones? |
| 02:15.44 | *** join/#asterisk inv_arp[work] (i=junya@c-71-206-88-100.hsd1.fl.comcast.net) |
| 02:16.08 | [TK]D-Fender | DocHolliday : Depends if you're doing any transocing, heavy recording, and PSTN connectivity/concurrency |
| 02:16.42 | bkruse_home | ya, thats a really hard question to ask |
| 02:16.50 | bkruse_home | complex IVR's or just playback |
| 02:17.48 | [TK]D-Fender | transcoding* |
| 02:18.02 | bkruse_home | [TK]D-Fender: agreed |
| 02:19.30 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 02:20.56 | *** join/#asterisk inv_arp[work] (i=junya@c-71-206-88-100.hsd1.fl.comcast.net) |
| 02:24.04 | rbd | hey guys...doesn't asterisk manager have a file based interface (i.e. I create a file at a certain location with my manager command...as an alternative to a telnet connection) |
| 02:24.04 | Nugget | telnet is eeeeeeevil! |
| 02:24.10 | rbd | it is |
| 02:25.41 | perd | you could easily do that with a perl script |
| 02:26.10 | bkruse_home | rbd: you could use a call file |
| 02:26.13 | rbd | perd: I remember there being a location under /var where I could throw a file containing an asterisk command to execute |
| 02:26.29 | perd | you can dial like that, rbd |
| 02:26.32 | rbd | well bascally I want to make an ORIGINATE invocation |
| 02:27.05 | rbd | okay is dial and originate functionally the same? it will be using a local channel type to basically bridge one extension in the dialplan to another... |
| 02:27.32 | perd | you could do that |
| 02:27.38 | bkruse_home | rbd: thats not hard at all |
| 02:27.44 | bkruse_home | look at the sample.call file in ur source dir |
| 02:27.46 | bkruse_home | and dump it to /var/spool/asterisk/outgoing |
| 02:27.53 | rbd | ok sounds good, thanks guys |
| 02:28.26 | bkruse_home | np |
| 02:28.35 | *** join/#asterisk inv_arp[work] (i=junya@c-71-206-88-100.hsd1.fl.comcast.net) |
| 02:28.46 | perd | hrmm, is it worth putting my cdr records in a database... |
| 02:29.43 | perd | i guess 'records' was kind of redundant |
| 02:29.44 | bkruse_home | YES |
| 02:30.12 | perd | i was just going to rotate logs every month and write a script to output the data by month in xls |
| 02:30.39 | bkruse_home | perd: that works too. |
| 02:30.47 | bkruse_home | reason for keeping the cdr logs? |
| 02:30.53 | perd | dunno |
| 02:30.54 | bkruse_home | records rather |
| 02:30.55 | bkruse_home | lol |
| 02:30.56 | bkruse_home | sweet |
| 02:30.57 | perd | seems like sometihng i should keep |
| 02:31.11 | perd | in case an employee gets in trouble |
| 02:31.20 | bkruse_home | gotcha |
| 02:31.22 | *** join/#asterisk DocHolliday (i=RogerRab@gateway/gpg-tor/key-0x0E4F6D6C) |
| 02:31.27 | bkruse_home | why not then |
| 02:31.30 | bkruse_home | thats not much space at all |
| 02:31.31 | perd | or we get strange calls that we want to know what the deal was |
| 02:31.32 | robl^ | perd: yeah.. like an employee is calling a phone sex line during hours |
| 02:31.43 | perd | haha no access to 900# :) |
| 02:31.58 | perd | you get tt-monkeys if you try that |
| 02:32.08 | DocHolliday | bkruse, sorry back |
| 02:32.32 | bkruse_home | DocHolliday: ha, hello ;] |
| 02:32.45 | bkruse_home | robl^: happends all the time :P |
| 02:33.07 | DocHolliday | bkruse_home, with those feature sets do you have any idea what level of hardware would be required? |
| 02:33.30 | bkruse_home | DocHolliday: what features? voicemail? |
| 02:33.38 | bkruse_home | DocHolliday: the main number i need, is how many concurrent calls. |
| 02:33.57 | DocHolliday | bkruse, assuming rougly 30 calls at once |
| 02:34.37 | DocHolliday | that might be on the low side.. |
| 02:34.59 | bkruse_home | DocHolliday: thats the number you need to worry about |
| 02:35.04 | bkruse_home | registerys not SO much.. |
| 02:35.09 | bkruse_home | (sp*) |
| 02:35.19 | DocHolliday | np |
| 02:35.28 | DocHolliday | is there a formula per call? |
| 02:35.34 | *** join/#asterisk inv_arp[work] (i=junya@c-71-206-88-100.hsd1.fl.comcast.net) |
| 02:36.03 | bkruse_home | DocHolliday: not really......would be handy |
| 02:36.09 | bkruse_home | look at voip-info.org for some info |
| 02:36.17 | bkruse_home | they had some good guesstimates |
| 02:36.46 | DocHolliday | basically what i want to know is if i have a pair of machines that are say.. Dual Xeon 3.0s with 2GB of ram and a few SCSI drives in raid will i be fine :P |
| 02:36.56 | *** join/#asterisk Bhaal (i=bhaal@freenode/unconfirmed/bhaal) |
| 02:37.39 | bkruse_home | i think so. |
| 02:37.41 | robl^ | DocHolliday: with those specs, you will barely get 2 calls. ;-) |
| 02:38.11 | DocHolliday | robl^, oh no :( |
| 02:38.11 | DocHolliday | kidding :P |
| 02:38.48 | DocHolliday | robl^, i wouldnt mind having a rack of servers though.. awfully costly though :) |
| 02:39.00 | [TK]D-Fender | DocHolliday : What would you be using for PSTN connectivity? |
| 02:39.14 | robl^ | DocHolliday: of course! 1 per call ;-) |
| 02:39.16 | bkruse_home | DocHolliday: i think you will be fine, are you using PSTN term |
| 02:39.25 | bkruse_home | robl^: barely |
| 02:40.31 | DocHolliday | [TK]D-Fender, yeah there is a geographic spread.. POTS in each location averaging 4-6 lines i'd say |
| 02:40.42 | DocHolliday | i have to be careful with rack dense servers that there are enough riser slots for the PCI cards |
| 02:40.54 | *** join/#asterisk inv_arp[work] (i=junya@c-71-206-88-100.hsd1.fl.comcast.net) |
| 02:41.16 | bkruse_home | DocHolliday: get 2 tdm400p cards |
| 02:41.30 | bkruse_home | and 8 fxs modules, just make sure that you got power in the case and that you have good air flow |
| 02:41.39 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 02:41.39 | *** mode/#asterisk [+o mog] by ChanServ |
| 02:41.49 | *** join/#asterisk SwK (n=Silik0nJ@12-214-191-109.client.mchsi.com) |
| 02:42.28 | DocHolliday | bkruse, the phones will be all IP-Based SIP |
| 02:44.08 | bkruse_home | DocHolliday: figured. |
| 02:44.11 | bkruse_home | still |
| 02:44.20 | bkruse_home | thats for PSTN connectivity |
| 02:44.38 | DocHolliday | i understand the FXO but why the FXS? |
| 02:44.44 | bkruse_home | ha |
| 02:44.45 | bkruse_home | my bad |
| 02:44.50 | bkruse_home | well |
| 02:44.51 | bkruse_home | actually |
| 02:44.59 | *** join/#asterisk stkn (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 02:45.02 | bkruse_home | well.... |
| 02:45.02 | DocHolliday | :) |
| 02:45.05 | bkruse_home | lets see |
| 02:45.21 | bkruse_home | you could do 8 fxo for pstn connectivity........or..... |
| 02:45.31 | bkruse_home | a single span T1 with a breakout box + channel bank |
| 02:45.38 | bkruse_home | but thats getting pricey |
| 02:46.01 | DocHolliday | well may need some FXS for door intercoms or something, you may be right.. (just not for the right reasons) *grin* |
| 02:47.00 | bkruse_home | DocHolliday: i know, trust me, just got mixed up ;] |
| 02:47.12 | bkruse_home | sup to you how you want to do pstn term though |
| 02:47.36 | DocHolliday | bkruse, im probably going to need 1 server per location and then 2 centrally managed interconnect boxes |
| 02:48.08 | DocHolliday | no point using redundant boxes at each location since the PSTN break out is directly to the machine |
| 02:48.13 | *** join/#asterisk inv_arp[work] (i=junya@c-71-206-88-100.hsd1.fl.comcast.net) |
| 02:50.00 | [TK]D-Fender | bkruse : You would take a nice digital card and waste it on a channelbank?! EW |
| 02:50.10 | bkruse_home | [TK]D-Fender: agreed!! |
| 02:50.13 | *** join/#asterisk olsen_ (n=olsen@200.61.236.49) |
| 02:50.20 | [TK]D-Fender | ICk |
| 02:50.20 | tzanger | [TK]D-Fender: what's wrong with a channel bank? |
| 02:50.22 | bkruse_home | DocHolliday: iax. |
| 02:50.43 | olsen_ | hi. my friend says that asterisk is illegal in our country. i mean. for private use |
| 02:50.44 | olsen_ | wtf? |
| 02:50.46 | bkruse_home | let the t1 company do your terminations :] |
| 02:50.52 | *** join/#asterisk oQPa (n=roque@198.Red-83-41-117.dynamicIP.rima-tde.net) |
| 02:51.00 | bkruse_home | olsen_: thats the craziest thing ive ever heard, |
| 02:51.04 | bkruse_home | slap your friend in the face |
| 02:51.10 | [TK]D-Fender | tzanger : Why lose digital signaling just to take in analog lines? expensive and wasteful |
| 02:51.12 | bkruse_home | and get new friends |
| 02:51.14 | tzanger | 10x better than any other FXS/FXO termination I've seen |
| 02:51.21 | DocHolliday | bkruse, you mean switch the POTS lines to a T1 PRI / SIP ? |
| 02:51.30 | tzanger | [TK]D-Fender: ahh for incoming... well if you can't get fractional T1/PRI it's a decent alternative |
| 02:51.41 | [TK]D-Fender | olsen_ : Your friend is on crack. Try to get a bulk deal with his supplier |
| 02:51.49 | DocHolliday | haha. |
| 02:52.06 | DocHolliday | bkruse, my other question would be how does asterisk react to 200-300 MS of latency? |
| 02:52.14 | tzanger | olsen_: in some countries VOIP is illegal |
| 02:52.18 | [TK]D-Fender | DocHolliday : Slowly :) |
| 02:52.19 | bkruse_home | between boxes? |
| 02:52.22 | bkruse_home | your iax trunks |
| 02:52.23 | tzanger | India's been in the news a lot for that actually |
| 02:52.28 | DocHolliday | bkruse, correct. |
| 02:52.41 | bkruse_home | i would not try to do that, how many calls are going between boxes? |
| 02:52.53 | DocHolliday | well interoffice calling.. the locations are *far* apart |
| 02:53.01 | DocHolliday | i.e. different continents |
| 02:53.03 | bkruse_home | gotcha |
| 02:53.09 | bkruse_home | that would be a fine solution then |
| 02:53.13 | bkruse_home | i dont think you have another choice |
| 02:53.25 | bkruse_home | DocHolliday: i meant get a t1 from a company and let them worry about your termination |
| 02:53.28 | tzanger | DocHolliday: asterisk doesn't give too much probelm with latency. the humans on either side though might take offense |
| 02:53.41 | olsen_ | no, he said "commercial" use |
| 02:53.49 | olsen_ | hi. my friend says that asterisk is illegal in our country. i mean. for commercial use |
| 02:54.00 | tzanger | olsen_: in some countries it is |
| 02:54.03 | tzanger | where are you from? |
| 02:54.11 | DocHolliday | bkruse, that sounds nice. |
| 02:54.11 | olsen_ | paraguay |
| 02:54.19 | bkruse_home | olsen_: thats still dumb, slap your friend in the face |
| 02:54.21 | bkruse_home | oh |
| 02:54.23 | DocHolliday | if possible.. it would be nice,, woould make for a dirty dirty transition though |
| 02:54.29 | bkruse_home | paraguay? well, it wouldnt be an asterisk problem |
| 02:54.34 | tzanger | olsen_: I'd check with your government then |
| 02:54.36 | bkruse_home | DocHolliday: the T1? nah |
| 02:54.49 | tzanger | olsen_: it's not *asterisk* that is illegal, but voice over ip |
| 02:55.00 | DocHolliday | bkruse, ya because there are existing POTS / PBX in each place.. cant just pull the cables if you do it that way |
| 02:55.03 | tzanger | whether it's specifically illegal in paraguay... I'm not usre. I'm in Canada. :-) |
| 02:55.05 | [TK]D-Fender | olsen_ : Nowhere is * illegal. Your specific ACTIONS with it may be in some retarded places. |
| 02:55.06 | bkruse_home | DocHolliday: that would be an ideal solution for terminiations |
| 02:55.17 | olsen_ | i know is not an asterisk issue |
| 02:55.21 | olsen_ | is a corruption issue in my countyr |
| 02:55.25 | DocHolliday | yes for me it would make life easy, but i'd have to time the transition between the POTS and the PRIs |
| 02:55.26 | olsen_ | my country is really corrupted |
| 02:55.28 | bkruse_home | DocHolliday: if you already have the lines, then use the some analog cards deffintely |
| 02:55.40 | [TK]D-Fender | olsen_ : Time to get the hell outta Dodge |
| 02:55.41 | bkruse_home | DocHolliday: i see now, i would go for some analog cards then |
| 02:55.42 | DocHolliday | i can probably get them to transition the lines but it would be unpredictable |
| 02:55.56 | tzanger | [TK]D-Fender: I'm not sure that Dodge is in Paraguay. :-) |
| 02:56.12 | DocHolliday | bkruse, my problem lies since i want interoffice calling and there is 200-300MS latency between some locations.. will quality be survivable? |
| 02:56.28 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 02:56.40 | bkruse_home | DocHolliday: depends, thats 200-300ms of audio difference |
| 02:56.44 | tzanger | bkruse_home: 300ms is getting to the point where you're going to start talking over each other, but if it's constant, it can be worked around |
| 02:56.46 | [TK]D-Fender | tzanger : It's like "Springfield" Every territory must have one! |
| 02:56.52 | tzanger | (by the humans) :-) |
| 02:56.55 | tzanger | [TK]D-Fender: indeed |
| 02:56.56 | bkruse_home | tzanger: thats a rule of thumb, yes |
| 02:56.57 | olsen_ | tzafrir: wait, voice over ip is illegal? wtf? |
| 02:57.10 | bkruse_home | olsen_: ask your government |
| 02:57.26 | DocHolliday | tzanger, would a higher quality codec help? like G729? |
| 02:57.35 | olsen_ | bkruse_home: he probably doesn't even know what voice over ip is, he is a native moron |
| 02:57.48 | tzanger | olsen_: yes, some nations telecommunications giants have pushed their governments to make it illegal to sidestep them to save long distance tarriffs, or even local connections |
| 02:58.23 | tzanger | DocHolliday: not relaly, if it takes 300ms to get from point A to point B, you're going ot take 300ms no matter what. if it's an 80kbps stream or an 8kbps stream is irrelavent |
| 02:58.24 | danp | [TK]D-Fender: was it you who thought earlier that me getting a bit of echo at the start of a call with a sangoma card was weird? |
| 02:58.31 | danp | or you said you don't experience that |
| 02:58.43 | *** join/#asterisk inv_arp[work] (i=junya@c-71-206-88-100.hsd1.fl.comcast.net) |
| 02:58.44 | tzanger | DocHolliday: where codec comes into play is when you're fighting bandwidth contention or if you're dealing with packet loss |
| 02:58.57 | DocHolliday | tzanger, there is nothinig i can do about it.. maybe put a server halfway between the two points (worst possible case) |
| 02:59.06 | tzanger | DocHolliday: how will that help? |
| 02:59.14 | [TK]D-Fender | danp : Just saying I've heard it maybe only one or twice ever. |
| 02:59.15 | bkruse_home | tzanger: exactly lol |
| 02:59.21 | tzanger | DocHolliday: the only way to fix it is to solve the connectivity problem (different provider) |
| 02:59.25 | bkruse_home | DocHolliday: its your only solution. |
| 02:59.29 | bkruse_home | between continents, 200-300 ms isnt THAT bad |
| 02:59.33 | bkruse_home | olsen_: that is quite dumb..... |
| 02:59.34 | danp | [TK]D-Fender: do you use echotraining in zaptel.conf? |
| 02:59.36 | tzanger | no it's not |
| 02:59.39 | DocHolliday | tzanger, there is no problem its just a distance problem |
| 02:59.40 | danp | sorry, zapata.conf |
| 02:59.41 | *** join/#asterisk horsesgofaster (n=dcantera@pool-72-82-183-46.cmdnnj.east.verizon.net) |
| 02:59.43 | [TK]D-Fender | danp : No. Nor should you. |
| 02:59.45 | tzanger | transoceanic TDM calls are in that range are they not? |
| 02:59.53 | danp | hmm, that's what i was thinking. i'm not using it |
| 02:59.57 | tzanger | I know that the last few times I can recollect calling .de it's been significantly delayed |
| 03:00.23 | DocHolliday | tzanger, talking AU here |
| 03:00.34 | tzanger | DocHolliday: ok that's one point, where's the other? |
| 03:00.49 | DocHolliday | US / Canada |
| 03:00.56 | danp | [TK]D-Fender: do you have anything special in zapata.conf besides echocancel=yes and echocancelwhenbridged=yes? |
| 03:01.11 | danp | i pretty much took my zapata.conf from the sangoma website |
| 03:01.13 | tzanger | DocHolliday: yu'd have to do some testing, including testing of normal PSTN calls for delay |
| 03:01.40 | DocHolliday | tzanger, traceroutes dont give you a perfect indicating but i have seen variances from 215 MS to 350 |
| 03:01.50 | DocHolliday | *indication |
| 03:01.55 | tzanger | I said voice calls over the PSTN :-) |
| 03:02.01 | tzanger | traceroute doesn't work with those |
| 03:02.02 | DocHolliday | yeah i know :) |
| 03:02.06 | [TK]D-Fender | danp : nope. You DID enable it in your wanpipe conf files right? |
| 03:02.14 | DocHolliday | worst case we just wont do interoffice for that location |
| 03:02.19 | danp | sure, yeah...it definitely goes away after the first couple seconds |
| 03:02.33 | DocHolliday | tzanger, for data 300 MS just means a slight delay.. nothing earth shattering |
| 03:02.38 | bkruse_home | danp: its probably just "learning" |
| 03:02.41 | danp | maybe quicker than that |
| 03:02.44 | tzanger | I'm sure you've got significant delay there iwth PSTN even... how much I'm not sure |
| 03:02.56 | DocHolliday | yeah |
| 03:03.03 | DocHolliday | i just hope to god there isnt a problem with the VPN over ~200-300MS |
| 03:03.16 | tzanger | oh fuck |
| 03:03.20 | tzanger | VPNs don't care |
| 03:03.21 | danp | that was my guess but i just wanted to make sure it was the best i could get it |
| 03:03.22 | tzanger | it's just a slow network |
| 03:03.27 | tzanger | or rather a laggy one |
| 03:03.33 | DocHolliday | yeah |
| 03:03.43 | danp | it sounds great otherwise |
| 03:03.49 | DocHolliday | its not a capacity problem just a distance limitation |
| 03:04.08 | olsen_ | tzanger: so we cant use asterisk for commercial purposes? |
| 03:04.26 | tzanger | olsen_: I do not know the Paraguay laws |
| 03:04.27 | olsen_ | tzanger: voip |
| 03:04.36 | olsen_ | tzafrir: what about in your country? |
| 03:04.44 | tzanger | contact a lowyer familliar with your nation's laws in this matter |
| 03:04.49 | tzanger | olsen_: in Canada we can use VOIP for any purpose |
| 03:04.55 | DocHolliday | indeed. |
| 03:05.13 | DocHolliday | tzanger, my asterisk box has been up 130+ days here |
| 03:05.46 | perd | do you do nightly restarts? |
| 03:05.49 | DocHolliday | correction ~135 |
| 03:05.56 | coppice | We can use VoIP for any illegal purpose :-) |
| 03:06.29 | perd | like a beastiality sex line? |
| 03:06.34 | WilliamK | like calling your mother from her own phone number? |
| 03:06.36 | WilliamK | =) |
| 03:07.05 | DocHolliday | perd, nope. |
| 03:07.08 | DocHolliday | Sex*CLI> dial 1900605555 |
| 03:07.32 | WilliamK | calling the bosses wife using his callerID and saying "I am bungholio then hanging up" |
| 03:07.33 | WilliamK | :) |
| 03:07.56 | *** join/#asterisk inv_arp[work] (i=junya@c-71-206-88-100.hsd1.fl.comcast.net) |
| 03:07.58 | DocHolliday | WilliamK, or just change the CID as requireed |
| 03:08.22 | WilliamK | Doc- exactly what I was talking about :) |
| 03:08.28 | perd | calling the military from gwbs number and telling them to fire nukes! |
| 03:08.31 | WilliamK | crazy ways to drive others around you insane |
| 03:08.48 | DocHolliday | perd, you have more guts then i do.. |
| 03:08.55 | WilliamK | perd, nah... they would just hunt you |
| 03:08.56 | undrdawg | anyone use asterisk on freebsd? |
| 03:08.56 | WilliamK | =) |
| 03:09.04 | coppice | perd: will Mr Nukes find it easy to get a new job? |
| 03:09.19 | perd | no, he's got a record |
| 03:09.19 | DocHolliday | coppice, dont need to worry.. not much will be left |
| 03:09.24 | Crashsys | Can you boot off an MD Volume? Or do you have to boot off of something else? |
| 03:09.40 | Cerlyn | CallerID may be easy to fake; but most firms are not ANI originators |
| 03:11.12 | tzanger | DocHolliday: :-) |
| 03:11.24 | coppice | perd: a criminal record? something like a Barry Manilow CD? |
| 03:12.58 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 03:13.01 | shmaltz | hi everyone |
| 03:13.12 | shmaltz | anybody here familiar with shoretel |
| 03:13.55 | *** join/#asterisk inv_arp[work] (n=junya@c-71-206-88-100.hsd1.fl.comcast.net) |
| 03:14.40 | *** join/#asterisk RoyK (n=roy@217-175-39.100710.adsl.tele2.no) |
| 03:18.57 | *** join/#asterisk SwK (n=Silik0nJ@12-214-191-109.client.mchsi.com) |
| 03:19.19 | *** join/#asterisk inv_arp[work] (i=junya@c-71-206-88-100.hsd1.fl.comcast.net) |
| 03:22.07 | *** join/#asterisk h0 (n=fakhir@unaffiliated/fakhir) |
| 03:23.52 | *** join/#asterisk docelmo (n=vircuser@c-68-32-143-73.hsd1.de.comcast.net) |
| 03:27.41 | *** join/#asterisk RoyK (n=roy@217-175-39.100710.adsl.tele2.no) |
| 03:28.21 | undrdawg | am i going to need a software phone to use with asterisk if i just want to make calls? |
| 03:28.46 | undrdawg | or can i use it for making and recieving calls too? |
| 03:29.01 | *** join/#asterisk k-man_ (n=jason@unaffiliated/k-man) |
| 03:29.03 | k-man_ | hi |
| 03:29.13 | k-man_ | anyone know where to get polycom voip phones in australia? |
| 03:29.33 | DocHolliday | k-man_, are you in australia? |
| 03:29.38 | k-man_ | yes |
| 03:29.45 | DocHolliday | mind if i ask you a question? |
| 03:29.58 | k-man_ | sure |
| 03:30.00 | k-man_ | ask away |
| 03:30.02 | [TK]D-Fender | undrdawg : Umm... not entirely clear there. Care to rephrase that.. |
| 03:30.05 | k-man_ | i may not know the answer though |
| 03:30.50 | DocHolliday | what have you found to be the best provider in australia when viewing north american content? |
| 03:31.18 | undrdawg | err i'll figure it out i guess im reading the docs now |
| 03:32.23 | *** join/#asterisk inv_arp[work] (i=junya@c-71-206-88-100.hsd1.fl.comcast.net) |
| 03:32.29 | *** join/#asterisk BSDTech (n=RNeese@69.239.85.71) |
| 03:32.37 | WilliamK | undrdawg, you need either a softphone (software+mic/speakers/headset), IP Phone, or ATA gateway to interface with asterisk and you will be able to make and recieve calls (unless the softphone software you use only allows one way which is not typical) |
| 03:33.00 | k-man_ | DocHolliday, regarding net access? |
| 03:33.05 | DocHolliday | yep |
| 03:33.24 | k-man_ | DocHolliday, well.... i think they are all about the same.. but i have not done any kind of real survey |
| 03:33.32 | k-man_ | internode who i use at the moment are good |
| 03:33.40 | k-man_ | but peopletelecom who i used to use, were fine too |
| 03:33.42 | DocHolliday | thanks ;) |
| 03:33.51 | k-man_ | i stopped using people telecom for other reasons |
| 03:33.56 | k-man_ | nw |
| 03:34.01 | DocHolliday | no problem, appreciated. |
| 03:34.26 | k-man_ | i think that there is only 1-2 wholesalers who provide a link to the US anyway |
| 03:34.51 | k-man_ | so no matter who you are with, your photons end up goind down the same bit of glass |
| 03:35.40 | shmaltz | anybody here familiar with shoretel |
| 03:35.43 | k-man_ | but i'm not expert so don't take that as gospel |
| 03:35.47 | undrdawg | i have a real fxo card |
| 03:36.06 | undrdawg | its just in my storage shed and i wanted to make a quick sip call |
| 03:36.36 | [TK]D-Fender | WilliamK : Left out the grand world of TDM cards there :) |
| 03:36.39 | undrdawg | are there any programs that are sorta all in one |
| 03:36.42 | undrdawg | ? |
| 03:37.37 | [TK]D-Fender | undrdawg : You technically don't even need *. Just get a soft-phone and setup with a SIp provider somewhere. |
| 03:38.36 | undrdawg | ok i have sip service |
| 03:38.49 | perd | x-lite works well |
| 03:38.51 | WilliamK | TK - I was trying to make it simple |
| 03:38.52 | WilliamK | =) |
| 03:38.54 | [TK]D-Fender | undrdawg : So got DL a soft-phone, set it up and you're done |
| 03:38.54 | undrdawg | i guess that setup would sorta be for providing service |
| 03:39.02 | [TK]D-Fender | WilliamK : FAILURE |
| 03:39.13 | WilliamK | TK - darn, I hate it when that happens :) |
| 03:39.40 | undrdawg | you think it would dial faster if i setup asterisk and then an softphone on another computer like my laptop? |
| 03:39.50 | [TK]D-Fender | no |
| 03:40.19 | WilliamK | TK - ever try using a softphone over an EDGE connection? |
| 03:40.19 | WilliamK | =) |
| 03:40.31 | BSDTech | oook anyword on the fix for g729 transcoding ? |
| 03:40.37 | [TK]D-Fender | WilliamK : Not really familiar with that term... |
| 03:40.48 | BSDTech | I have tested 4 systems with 1.4 and they all have the same issue |
| 03:40.50 | WilliamK | Edge is the next step up from GPRS |
| 03:41.07 | [TK]D-Fender | WilliamK : Sounds painful |
| 03:41.29 | WilliamK | kinda curious since I'm able to get 160-218k throughputs now |
| 03:43.41 | *** join/#asterisk inv_arp[work] (i=junya@c-71-206-88-100.hsd1.fl.comcast.net) |
| 03:51.03 | *** join/#asterisk phatboy (n=yada@152.83.10.193) |
| 03:52.51 | *** join/#asterisk bmg505 (n=leon@c1-201-13.rndf.isadsl.co.za) |
| 03:56.18 | *** join/#asterisk inv_arp[work] (i=junya@c-71-206-88-100.hsd1.fl.comcast.net) |
| 03:56.35 | *** join/#asterisk Lopal (n=sad@ppp115-164.static.internode.on.net) |
| 03:56.50 | Lopal | has anybody here got asterisk 1.4 working with a cisco 7961g-ge? |
| 04:07.28 | *** join/#asterisk inv_arp[work] (i=junya@c-71-206-88-100.hsd1.fl.comcast.net) |
| 04:11.27 | DocHolliday | Lopal, got the 7940 working at 1.2.. cant say anything about 1.4 |
| 04:16.08 | *** join/#asterisk inv_arp[work] (i=junya@c-71-206-88-100.hsd1.fl.comcast.net) |
| 04:19.23 | *** join/#asterisk sethwhit (n=SethWhit@70-56-234-164.clsp.qwest.net) |
| 04:19.28 | sethwhit | Hello |
| 04:19.41 | sethwhit | I need some help |
| 04:20.03 | sethwhit | when I try to start Asterisk i get a TTY (9) error |
| 04:20.08 | sethwhit | it says it can not find it |
| 04:21.38 | horsesgofaster | sethwhit: I think there is a linux config param which specifies the number of pseudo terms to configure on boot... |
| 04:21.59 | horsesgofaster | perhaps in /etc/init.d... maybe I can check... |
| 04:22.15 | sethwhit | To tell you the truth I dont even know what the error is |
| 04:22.42 | sethwhit | it worked on my local server but when I tried to insall on a remote one I get this error |
| 04:22.47 | horsesgofaster | it would seem that it is an open error |
| 04:23.14 | horsesgofaster | when asterisk tries to create it's console it uses tty #9 |
| 04:23.22 | sethwhit | STARTING ASTERISK |
| 04:23.22 | sethwhit | Cannot find your TTY (9) |
| 04:23.22 | sethwhit | safe_asterisk: no process killed |
| 04:23.22 | sethwhit | mpg123: no process killed |
| 04:23.50 | horsesgofaster | yep, that sounds like the problem... I'm checking the inittab now... |
| 04:25.00 | horsesgofaster | # Run gettys in standard runlevels |
| 04:25.00 | horsesgofaster | 1:2345:respawn:/sbin/mingetty tty1 |
| 04:25.00 | horsesgofaster | 2:2345:respawn:/sbin/mingetty tty2 |
| 04:25.00 | horsesgofaster | 3:2345:respawn:/sbin/mingetty tty3 |
| 04:25.00 | horsesgofaster | 4:2345:respawn:/sbin/mingetty tty4 |
| 04:25.00 | horsesgofaster | 5:2345:respawn:/sbin/mingetty tty5 |
| 04:25.02 | horsesgofaster | 6:2345:respawn:/sbin/mingetty tty6 |
| 04:25.22 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 04:25.23 | horsesgofaster | looks like my asterisk only specifies six ttys... that may not be the exact solution |
| 04:26.01 | *** join/#asterisk Globetrotter (n=eric@205.211.239.11) |
| 04:26.08 | sethwhit | hmm |
| 04:26.53 | horsesgofaster | there is a place to configure these, I haven't played with them extensively and it has been some time since I dealt with this... let me think on it some and do some additional research... |
| 04:27.05 | Globetrotter | hey Guys,, how do i retrive my recorded calls from my box.. I have asterisk Business edition. cani install thirdparty apps on it? |
| 04:27.45 | bkruse_home | Globetrotter: if you can build the modules then yes, but if you need to rebuild it with asterisk source then no, obviously |
| 04:28.06 | horsesgofaster | sethwhit: are there devices created in /dev/... like crw-rw---- 1 root tty 4, 7 Jan 5 17:54 tty7 |
| 04:28.06 | horsesgofaster | crw-rw---- 1 root tty 4, 8 Jan 5 17:54 tty8 |
| 04:28.06 | horsesgofaster | crw-rw---- 1 asterisk tty 4, 9 Jan 7 23:26 tty9 |
| 04:28.06 | horsesgofaster | crw-rw---- 1 root uucp 4, 64 Jan 5 17:54 ttyS0 |
| 04:28.06 | horsesgofaster | crw-rw---- 1 root uucp 4, 65 Jan 5 17:54 ttyS1 |
| 04:28.06 | bkruse_home | Globetrotter: what business edition version? |
| 04:28.52 | sethwhit | a whole bunch |
| 04:29.08 | horsesgofaster | can you see specifically tty9? |
| 04:29.12 | sethwhit | yes |
| 04:29.23 | horsesgofaster | assigned to asterisk as above? |
| 04:29.24 | sethwhit | but before when I was working on this i did a mkdir /dev/tty9 |
| 04:29.25 | [TK]D-Fender | Globetrotter : Its a *nix box, and I'm sure your imagination can come up with a few dozen ways to get a simple file off your server... just that none of them have anything to do with *. |
| 04:29.35 | horsesgofaster | ok, that is ithe problem |
| 04:29.44 | bkruse_home | [TK]D-Fender: agreed. |
| 04:29.46 | horsesgofaster | tty9 is a device not a directory |
| 04:29.52 | bkruse_home | file is always in the box |
| 04:29.54 | DocHolliday | arghhh darn elevator default music |
| 04:29.55 | bkruse_home | *gives file crackers* |
| 04:30.08 | file | yay |
| 04:30.17 | Qwell | boo file |
| 04:30.25 | DocHolliday | is it legal to use copyright songs that you have purchased as asterisk call hold music? |
| 04:30.29 | sethwhit | horsesgofaster: ok |
| 04:30.29 | perd | i just wrote a perl script to parse a custom Master.csv for CDR records and outputs the fields you select as an XLS |
| 04:30.33 | sethwhit | any ideas on how to fix |
| 04:30.37 | Qwell | DocHolliday: define "purchased" |
| 04:30.38 | file | Qwell, :( |
| 04:30.40 | [TK]D-Fender | You know I'm going to have to buy a building and play nothing but Iron Maiden in its elevator. Thats confuse the shit outta people,... |
| 04:30.43 | perd | you could do the same, then set it up to mail you with Mail::Sender |
| 04:30.46 | DocHolliday | Qwell, CD? |
| 04:30.57 | Qwell | no, all you own is the phsyical media |
| 04:31.10 | [TK]D-Fender | perd : Or you could just import it as its alcready CSV and pretty damn easy to import ;) |
| 04:31.13 | DocHolliday | thus you need a special licensing agreement? |
| 04:31.14 | bkruse_home | *gives file ritz to cheer him up* |
| 04:31.21 | perd | [TK]D-Fender: haha good point :/ |
| 04:31.21 | Qwell | DocHolliday: yes |
| 04:31.22 | horsesgofaster | sethwhit: the crw-rw---- indicates it is a char device 'crw' c=char d=directory... you have to remove the file and re-run the makedev cmd... I'll look up the format/syntax.... hold on... |
| 04:31.32 | file | walk with me... like lovers do! |
| 04:31.50 | DocHolliday | Qwell, so my options are either Sound Input (via Radio FM) or Satellite Radio? |
| 04:31.53 | perd | but i'm putting this up as a simple web script.. 'check fields you want, check month you want, click submit' |
| 04:31.58 | Qwell | DocHolliday: no, neither are legal |
| 04:32.07 | perd | and it color codes stuff too, long distance is diff color than local, and so on |
| 04:32.08 | bkruse_home | define 'legal' |
| 04:32.08 | perd | hehe |
| 04:32.10 | bkruse_home | :P |
| 04:32.11 | perd | i got bored |
| 04:32.15 | DocHolliday | Qwell, what do you recommend? |
| 04:32.39 | Qwell | DocHolliday: get a license for tens of thousands of dollars a year, or use royalty free music |
| 04:33.08 | DocHolliday | Qwell, i have better uses for the funds in the budget :) |
| 04:33.27 | DocHolliday | i.e. installing the asterisk system to begin with :P |
| 04:33.33 | horsesgofaster | sethwhit: let's try the easiest first... reboot... if that doesn't work, then cut and past your /dev/tty6-9 entries in here and I'll try to give you the makenod cmd. |
| 04:33.51 | horsesgofaster | ok, I forgot one key item |
| 04:34.02 | DocHolliday | Qwell, in your opinion have you found any phones that work better than the Cisco 79xx? |
| 04:34.02 | horsesgofaster | you have to remove /dev/tty9 directory first |
| 04:34.15 | Qwell | polycom 650 |
| 04:34.38 | DocHolliday | how do they compare? |
| 04:34.41 | [TK]D-Fender | DocHolliday : Just about any Polycom. |
| 04:34.50 | Qwell | sound quality is the best there is |
| 04:35.02 | Qwell | [TK]D-Fender: you used a 650 yet? |
| 04:35.02 | Lopal | hey guys, does anyone have the latest SIP firmware for the 79x0 and 79x1g phones? |
| 04:35.10 | Qwell | even with ulaw, it's SERIOUSLY amazing |
| 04:35.23 | DocHolliday | Lopal, download it off the cisco site. |
| 04:35.38 | [TK]D-Fender | Qwell : Nope. No point for me to. |
| 04:35.42 | DocHolliday | Qwell, the 650 appears to be more costly than the 7940 (Cisco), but i guess its worth comparing |
| 04:35.48 | bkruse_home | DocHolliday: or get a synthesizer and some software, slap my butt and call me chuck |
| 04:35.53 | sethwhit | horseesgofaster: ok I did both a reboot and deleted the directory |
| 04:35.55 | DocHolliday | heh |
| 04:35.58 | Lopal | DocHolliday: Dont have access :( |
| 04:36.00 | bkruse_home | the 650 freaking rules, no doubt |
| 04:36.07 | DocHolliday | Lopal, dont need access... |
| 04:36.09 | Qwell | bkruse_home: you heard one on speaker yet? |
| 04:36.12 | sethwhit | one other thing I noticed is that their is like ttya-y but no tty1 tty2 |
| 04:36.19 | [TK]D-Fender | Qwell : its a 601 + backlight (I don't really care), USB expansion (just what can youdo with it NOW?), and HD (profitable WHERE?). So... no :) |
| 04:36.20 | bkruse_home | nope, only got to see one walking by the office |
| 04:36.21 | bkruse_home | :[ |
| 04:36.21 | horsesgofaster | sethwhit, removing the directory before the reboot, of course :) |
| 04:36.35 | DocHolliday | http://www.cisco.com/pcgi-bin/tablebuild.pl/sip-ip-phone7960 |
| 04:36.39 | bkruse_home | HD profitable where huh?? hehehe |
| 04:36.40 | Qwell | [TK]D-Fender: the sound quality with even 8khz codecs is like...5x better |
| 04:36.50 | [TK]D-Fender | Qwell : You wanna GIVE me one, I wouldn't say no however ;) |
| 04:36.51 | perd | cisco phones suck |
| 04:36.58 | perd | i know this because i have them |
| 04:36.58 | horsesgofaster | sethwhit, what distro of linux are you using? |
| 04:37.00 | Qwell | perd: no, they don't |
| 04:37.03 | Lopal | yep, got that one, updated my 7960 and its up and running fine |
| 04:37.06 | [TK]D-Fender | Qwell : I find that dispicably hard to believe |
| 04:37.08 | sethwhit | CentOS 4 |
| 04:37.12 | Lopal | however my 7961G-GE wont load that firmware |
| 04:37.12 | bkruse_home | cisco == the internet |
| 04:37.17 | Qwell | [TK]D-Fender: You'll see :) |
| 04:37.20 | bkruse_home | true |
| 04:37.23 | DocHolliday | Qwell, yeah they might be a bit budget for deploymen of 100+ |
| 04:37.28 | DocHolliday | over budget |
| 04:37.47 | perd | meh , juniper is better |
| 04:37.57 | *** join/#asterisk unsucht (n=dwayne@64-42-247-120.mb.skyweb.ca) |
| 04:37.58 | perd | i dont have a friggen cisco login |
| 04:38.01 | DocHolliday | juniper phones? |
| 04:38.07 | perd | so i cant get any sip shit for my cisco phones |
| 04:38.11 | perd | that's why sicso sucks balls |
| 04:38.16 | [TK]D-Fender | DocHolliday : You have a large deployment to prepare? |
| 04:38.27 | DocHolliday | [TK]D-Fender, tentatively |
| 04:38.32 | Lopal | apparently for the 7961G-GE SIP firmware its at www.cisco.com/pcgibin/tablebuild.pl/ip-7900ser |
| 04:38.38 | horsesgofaster | sethwhit, ok, I'm on RH... somewhat similar... |
| 04:38.41 | unsucht | does anyone know if asterisk has a feature that will call an extention when another one becomes available if a use requests it? |
| 04:38.41 | Lopal | however you need a cisco service agreement |
| 04:38.45 | horsesgofaster | sethwhit, did asterisk start? |
| 04:38.46 | [TK]D-Fender | DocHolliday : Ip 430 then. Cheap PoE deployment and plenty for 99% of users |
| 04:38.51 | DocHolliday | [TK]D-Fender, if i do get it i'll be taking plenty of pictures :) |
| 04:39.21 | sethwhit | horsesgofaster: I am wating for the server to come online |
| 04:39.48 | horsesgofaster | sethwhit, I'm moving to the other session... |
| 04:39.51 | [TK]D-Fender | DocHolliday : clarification : enough features for 99% of your users |
| 04:39.52 | DocHolliday | [TK]D-Fender, i was thinking Cisco 7940's they have proved stable.. not sure if its worth spending more for Polycom (mind you that model is a little less) |
| 04:40.00 | bkruse_home | Qwell: the speaker that amazing? |
| 04:40.25 | Globetrotter | ok thanks, guys.. i thought that asterisk had a build it way to retrive the files |
| 04:40.39 | DocHolliday | heh. |
| 04:40.43 | *** join/#asterisk KuJaX (n=kuj@71-219-82-109.slkc.qwest.net) |
| 04:40.59 | DocHolliday | what is the USB port for? |
| 04:41.28 | [TK]D-Fender | DocHolliday : News flash : Polycom costs a lot LESS than Cisco. IP430 = $150 USd |
| 04:41.46 | bkruse_home | Globetrotter: system(wget http:// ) :D |
| 04:41.48 | DocHolliday | [TK]D-Fender, yeah.. i said that model in particular was less. |
| 04:42.05 | bkruse_home | exten => 1,1,System(wget http://digium.com) |
| 04:42.18 | [TK]D-Fender | DocHolliday : You weren't seriously looking at the 650 were you? total waste for the cost different |
| 04:42.21 | [TK]D-Fender | difference |
| 04:42.35 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
| 04:42.35 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 04:42.40 | DocHolliday | [TK]D-Fender, i look at everything.. |
| 04:42.44 | bkruse_home | Qwell: i was wondering where you went! |
| 04:42.45 | Qwell | meh |
| 04:42.49 | bkruse_home | [TK]D-Fender: they wont be in the future |
| 04:42.49 | [TK]D-Fender | DocHolliday : 601 + Expansion modules for your resceptionist, IP 430's for the rest |
| 04:43.08 | DocHolliday | [TK]D-Fender, Asterisk doesnt support call attendant? |
| 04:43.10 | Qwell | but yeah, it really does sound that much better |
| 04:43.36 | [TK]D-Fender | DocHolliday : ... HUH?! |
| 04:43.45 | DocHolliday | [TK]D-Fender, it doesnt show the status of each extension? |
| 04:43.59 | [TK]D-Fender | DocHolliday : Sure it does. |
| 04:43.59 | bkruse_home | Qwell: does the microphone pickup everything in the room still? |
| 04:44.07 | Qwell | dunno, never heard the other end |
| 04:44.10 | bkruse_home | hmm |
| 04:44.13 | Lopal | DocHolliday: The firmware at http://www.cisco.com/pcgi-bin/tablebuild.pl/sip-ip-phone7960 is V8.2.0 i heard there was a version 8.5.0 is this true? |
| 04:44.19 | DocHolliday | [TK]D-Fender, you rock my world.. the expansion module actually shows if individual users are using their handsets? |
| 04:44.47 | DocHolliday | Lopal, i use an older version.. 7.x but there may be a newer one.. check voip-info.org |
| 04:45.14 | [TK]D-Fender | DocHolliday : yes, thats entirely the point of my suggesting a single 601 + expansion modules for your receptionist |
| 04:45.25 | Lopal | yeah thats where i got the info from, there is a v 8.5.0 for 7960 and the 7961G-GE FW is at http://www.cisco.com/pcgibin/tablebuild.pl/ip-7900ser but i dont have the service aggreement to download it.... does anyone here have access to get it? |
| 04:45.33 | DocHolliday | my god, how does asterisk handle that for Polycom yet not cisco? |
| 04:45.35 | Grnd-Wire | TKD: Which brand are you referring to? |
| 04:45.43 | Grnd-Wire | Polycom? |
| 04:45.53 | DocHolliday | ya |
| 04:46.10 | undrdawg | i should have got a fxs card |
| 04:46.18 | undrdawg | i think i got the wrong thing |
| 04:46.40 | undrdawg | can i setup asterisk to give phone service to a POTS line? |
| 04:46.48 | [TK]D-Fender | DocHolliday : Cisco does not focus on better SIP support. Its against their business model to FREE you from their propriety gear |
| 04:46.49 | DocHolliday | i didnt think asterisk supported hardware-based receptionist features |
| 04:47.11 | DocHolliday | [TK]D-Fender, even if i go ahead with cisco its worth it to use that for the receptionists anyway |
| 04:47.15 | [TK]D-Fender | undrdawg : Clarify that.... given the terms you through around any answer would be potty right now... |
| 04:47.45 | undrdawg | i want to use a real phone |
| 04:47.51 | [TK]D-Fender | DocHolliday : Polycom Is still cheaper for the rest of your setup anyways. |
| 04:47.51 | DocHolliday | can you benefit from call attendant with the polycom when using cisco phones entirely? |
| 04:48.05 | [TK]D-Fender | undrdawg : with your SIP provider account? |
| 04:48.09 | undrdawg | yea |
| 04:48.10 | DocHolliday | [TK]D-Fender, sure it is.. but i have to give the company choice. |
| 04:48.14 | perd | fxs will give you dialtone to a telephone |
| 04:48.24 | perd | fxo will let you connect a phone line from the pstn to asterisk |
| 04:48.27 | file | a more beautiful vision... I have never seen, if you don't mind me saying... a lifelong ambition to fulfill my dream... what have you done to me? |
| 04:48.28 | *** part/#asterisk BSDTech (n=RNeese@69.239.85.71) |
| 04:48.30 | perd | fxo == no dialtone |
| 04:48.32 | [TK]D-Fender | undrdawg : get an ATA like the Linksys SPA-2002 then |
| 04:48.40 | KuJaX | Anyone know if IAX doesnt go through routers sometimes? I am not able to connect to my provider (Voicepulse) |
| 04:48.50 | KuJaX | After I switched routers |
| 04:49.19 | undrdawg | ya i accidently ordered a fxo card a while back |
| 04:49.26 | Grnd-Wire | KuJax: Make sure you have 4569/UDP forwarded to your Asterisk machine. |
| 04:49.28 | [TK]D-Fender | DocHolliday : Sure, Cisco should be runner up if it doesn't strain the budget. Mind you another great choise is the Aastra 480i. Thats a GREAT phone for * usage. my runner up from Polycom. |
| 04:49.30 | undrdawg | do they sell fxs in any stores usually ? |
| 04:49.39 | bkruse_home | perd: fxs fxSENDS dialtone |
| 04:49.43 | bkruse_home | thats how i always remember it |
| 04:49.55 | Qwell | ~fxofxs |
| 04:51.18 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 04:51.18 | *** topic/#asterisk is Asterisk: The Open Source PBX -=- Check out the new http://www.asterisk.org !!! -=- Asterisk, Asterisk-addons, Zaptel, and Libpri 1.4.0 released!!! (December 23, 2006) -=- Join #freepbx for freepbx/trixbox support. -=- Join #asterisk-gui to learn about the new Asterisk GUI framework |
| 04:51.22 | Qwell | ~fxofxs |
| 04:51.23 | jbot | i guess fxofxs is An FXO port expects to receive dialtone and receive ring voltage, you plug a phone LINE into this. An FXS port expects to provide dialtone and provide ring voltage, you plug a PHONE into this. |
| 04:51.29 | Lopal | has anyone had much to do with setting up * + skinny |
| 04:51.31 | Qwell | w00t |
| 04:51.39 | DocHolliday | sure for executive offices but not everywhere :O |
| 04:51.41 | file | jbot: uptime |
| 04:52.01 | [TK]D-Fender | Grnd-Wire : http://www.telephonydepot.com/Polycom_s/25.htm |
| 04:52.04 | KuJaX | What does "Verbosity is at least 1" mean? |
| 04:52.08 | file | jbot: status |
| 04:52.09 | jbot | Since Mon Jan 8 04:50:05 2007, there have been 0 modifications, 1 question, 0 dunnos, 0 morons and 1 command. I have been awake for 2m 3s this session, and currently reference 112319 factoids. I'm using about 15148 kB of memory. With 0 active forks. Process time user/system 17.4/2.03 child 0.04/0.01 |
| 04:52.27 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 04:52.29 | DocHolliday | [TK]D-Fender, the Cisco is about 20 bucks more per phone.. but that translates to a few thousand in savings.. |
| 04:52.30 | bkruse_home | zomgz |
| 04:52.38 | bkruse_home | Qwell: hide! |
| 04:52.40 | bkruse_home | Qwell: try it again |
| 04:52.47 | [TK]D-Fender | DocHolliday : $170 per phone? |
| 04:52.54 | DocHolliday | yup |
| 04:52.55 | Grnd-Wire | (*&^% I just looked at that particular site, and didn't see the sidecar.. then you paste a link, and THERE it is.. :D |
| 04:53.09 | bkruse_home | Qwell: i bet it was all the regular expression stuff from yesterday :P |
| 04:53.12 | [TK]D-Fender | DocHolliday : licensed any everything? What about PoE for them? |
| 04:53.12 | *** join/#asterisk mistermocha (n=espresso@adsl-75-40-52-112.dsl.irvnca.sbcglobal.net) |
| 04:53.16 | Qwell | heh |
| 04:53.25 | DocHolliday | [TK]D-Fender, with SIP image of course. no licensing? |
| 04:53.30 | Qwell | What's $170? |
| 04:53.46 | Qwell | 7940? |
| 04:53.59 | [TK]D-Fender | DocHolliday : And to power them? |
| 04:54.05 | mistermocha | how much bandwidth does a G711 u-law connection take up? |
| 04:54.09 | bkruse_home | 80 kbps! |
| 04:54.12 | bkruse_home | with ip overhead |
| 04:54.19 | bkruse_home | 64kbps to be EXACT, but 80 with overhead. |
| 04:54.27 | DocHolliday | [TK]D-Fender, we are planning on using Cisco 3560 so that doesnt matter. |
| 04:54.27 | bkruse_home | ~ulaw |
| 04:54.29 | jbot | ulaw is, like, pronounce "mu"-law and consumes 64 Kb/s in each direction. It is considered a loss-less CODEC with a sampling rate of 8,000 hz and is 8 bit. It delivers quality equivalent to that of a POTS line. |
| 04:54.29 | mistermocha | bkruse_home: that was quick! |
| 04:54.34 | bkruse_home | mistermocha: :] |
| 04:54.50 | file | it was drilled into his head in basic Asterisk training I betz |
| 04:54.57 | DocHolliday | Cisco 3560 supports PoE |
| 04:54.59 | mistermocha | well while I'm at it... |
| 04:55.01 | [TK]D-Fender | DocHolliday : How much does that run? You can power Polycom's off a $400 standard PoE switch for 24 ports.... |
| 04:55.01 | bkruse_home | file: it was a pri training |
| 04:55.02 | mistermocha | ~alaw |
| 04:55.10 | bkruse_home | :P |
| 04:55.23 | bkruse_home | kenny answered before me and got a piece of candy though ;[ |
| 04:55.34 | bkruse_home | [TK]D-Fender: POE is so cool. |
| 04:55.36 | DocHolliday | [TK]D-Fender, they are costly.. but it doent matter, need them for data, network cameras, wifi, rfid, security etc |
| 04:55.40 | file | bkruse_home, lame |
| 04:55.45 | bkruse_home | file: indeed |
| 04:56.02 | bkruse_home | but you got me some crackers so.....that was better than dumb ole candy |
| 04:56.16 | [TK]D-Fender | DocHolliday : Ok, if thats the only kind of switching scenario you can picture then hey, whatever right? |
| 04:56.19 | KuJaX | how can I ensure that I am connecting to voicepulse alright? I literally didn't touch a thing but the router and now I can't get outbound |
| 04:56.31 | [TK]D-Fender | DocHolliday : Big $ in there |
| 04:56.49 | DocHolliday | [TK]D-Fender, its the right choice.. i can picture using Netgear 48 port switches :P |
| 04:57.03 | bkruse_home | mistermocha: i guess jbot doesnt know alaw, but its close to the same as far as bandwith, if not exactly the same |
| 04:57.04 | DocHolliday | but its better to use the right hardware that will last (since there is no onsite staff to service the damn things) |
| 04:58.36 | [TK]D-Fender | DocHolliday : We all do what we've gotta do I guess. My D-Link DES-1526's are runninggreat with my Polycom deployment, and I use Netgear for small area switches. |
| 04:59.09 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 05:01.42 | DocHolliday | [TK]D-Fender, pm if your interested. |
| 05:07.06 | *** join/#asterisk BugKhaM (n=LAMER@ppp-58.8.2.224.revip2.asianet.co.th) |
| 05:10.19 | BugKhaM | anyone upgraded from 1.2.13 to 1.2.14 on FC5 and has problems where SIP clients is disconnected from * server frequently? |
| 05:10.25 | Qwell | my clock is wonky...can somebody give me a time check? |
| 05:10.47 | file | 01:10AM my time, 11:10PM HSV time |
| 05:10.57 | Qwell | 10 minutes...sounds about right |
| 05:11.21 | Qwell | I seem to have lost like 9 minutes |
| 05:11.37 | BugKhaM | 23:01:21 doesn't seem right |
| 05:11.48 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 05:11.49 | sbingner | 19:11 now |
| 05:14.11 | greendisease | is the asterisknow registration site broken, i havent received an email with a password |
| 05:14.15 | greendisease | yes, i check my spam filter |
| 05:15.06 | greendisease | there is also a problem with the site, it cant connect to the mysql db |
| 05:15.14 | bkruse_home | greendisease: oh noes!!11!! |
| 05:15.26 | bkruse_home | let me check it out....if it cant access the mysql database, your not going to get an email. |
| 05:15.39 | greendisease | check out the asterisk.org site |
| 05:15.43 | greendisease | thats where the sql error is |
| 05:15.45 | bkruse_home | we have had a ton of hits in the last week |
| 05:16.07 | greendisease | i can imagine |
| 05:16.10 | bkruse_home | Qwell file we should probably do something about this...... |
| 05:16.18 | bkruse_home | brookshire: you here? |
| 05:17.15 | file | I sent him a message from my cell |
| 05:17.36 | bkruse_home | file: thank you, he should be able to take care of it |
| 05:18.26 | bkruse_home | greendisease: thanks, asterisknow.org and asterisk.org are tied together |
| 05:18.33 | greendisease | yeah i figured |
| 05:18.45 | greendisease | its good that you guys were in here, i would have called mark otherwise |
| 05:19.43 | bkruse_home | greendisease: mark is half way across the world currently. |
| 05:19.53 | bkruse_home | but he does know some people at digium :P |
| 05:19.59 | file | he knows a few. |
| 05:20.13 | bkruse_home | hehe |
| 05:20.19 | greendisease | bkruse: i know, thats why i figured he would be awake |
| 05:20.20 | bkruse_home | file: any word from brooks? |
| 05:20.22 | Qwell | I wish I knew Digium people :( |
| 05:20.29 | file | no - I suspect he is unconscious |
| 05:20.30 | Qwell | They seem like uber cool guys! |
| 05:20.30 | bkruse_home | Qwell: one day man....one day |
| 05:20.35 | file | his sleeping habits are weird presently |
| 05:20.41 | bkruse_home | file: agreed. |
| 05:20.51 | bkruse_home | file: i came in at 8 one day and he was just leaving |
| 05:20.56 | greendisease | ah yes, good ol jetlag |
| 05:21.03 | greendisease | gets em every time |
| 05:21.07 | Qwell | ITlag |
| 05:21.11 | greendisease | hahaha |
| 05:21.14 | greendisease | that too |
| 05:21.20 | Juggie | file, every geeks sleeping scheduled is broken |
| 05:21.32 | bkruse_home | Juggie: b0rked* |
| 05:21.35 | bkruse_home | :P |
| 05:21.37 | Juggie | i mean its 1:20am on a sunday, and you are up, hence file = broken :) |
| 05:21.45 | file | I've been staying up rather late lately... |
| 05:21.52 | bkruse_home | Qwell: just hang out at the front door and have a smoke, youll meet alot of people |
| 05:21.56 | bkruse_home | thats how i met everything :P |
| 05:22.03 | greendisease | its 12:20am |
| 05:22.07 | bkruse_home | file: no you have not! |
| 05:22.08 | file | I'm actually pondering cleaning off my desk, there's about 7 bottles here... |
| 05:22.14 | greendisease | its 1:20 am somewhere in the middle of the atlantic |
| 05:22.17 | Juggie | file, i got up to 75 water bottles @ work |
| 05:22.26 | file | Juggie, that's just out of control d00d |
| 05:22.27 | Juggie | before one of the cleaners caved and took them to the recycle bin for me |
| 05:22.30 | bkruse_home | Juggie: same here, but doctor pepp0r |
| 05:22.38 | bkruse_home | Juggie: at least its water, thats good |
| 05:22.42 | file | bkruse_home, so are going to chillax d00d? |
| 05:22.48 | Juggie | we have a blue recycle bin @ work |
| 05:22.52 | Juggie | and i put bottles in it |
| 05:22.56 | Juggie | but it seems its only for paper |
| 05:23.07 | Juggie | so i just keep putting bottles in and eventually, they crack and take them. |
| 05:23.16 | Qwell | Juggie: he probably made like $20 off those bottles :P |
| 05:23.19 | bkruse_home | file: d00d i will be up lat0r that you!1!1!! |
| 05:23.23 | file | lame |
| 05:23.26 | Juggie | hah, they are plastic |
| 05:23.33 | Juggie | probally like 50cents worth. |
| 05:23.38 | Juggie | he can have it. |
| 05:23.43 | Juggie | a 30 pack of water is 2.99 here. |
| 05:23.49 | Qwell | heh |
| 05:23.58 | Qwell | that's pretty cheap |
| 05:24.01 | Juggie | 30x500ml |
| 05:24.05 | file | yup - cleaning off desk |
| 05:24.08 | Juggie | yah, it rants from 2.99-4.99 |
| 05:24.10 | *** part/#asterisk jeremyBSD (n=bitchass@exit-collective.com) |
| 05:24.10 | bkruse_home | Juggie: nice, i have to pay 50 cents for a dr pepper, which half the time the machine owns my dollar bill |
| 05:24.12 | *** join/#asterisk Bazy (n=bazy@86.125.49.55) |
| 05:24.14 | Juggie | er, rangers |
| 05:24.16 | Juggie | *ranges |
| 05:24.23 | file | and perhaps beside my desk |
| 05:24.27 | bkruse_home | file: will you grab me some crackers and bring them into the living room paweaze?? |
| 05:24.32 | Juggie | file, i just finished cleaning my whole house :) |
| 05:24.34 | Globetrotter | hi Guys,, i cant seem to get MOH to work, in musiconhold.conf i have mode=quietmp3 |
| 05:24.34 | Globetrotter | directory=/var/lib/asterisk/mohmp3 |
| 05:24.39 | Juggie | vacumed the entire thing :) |
| 05:24.42 | bkruse_home | Globetrotter: error? |
| 05:25.08 | Globetrotter | <PROTECTED> |
| 05:25.08 | Globetrotter | <PROTECTED> |
| 05:25.36 | Juggie | file, have you put a HD source into your new tv yet? |
| 05:25.48 | bkruse_home | NEVER! |
| 05:26.28 | file | Juggie, no |
| 05:28.42 | Juggie | file, you disapoint me |
| 05:28.47 | Juggie | at least hook up your computer to it. |
| 05:28.53 | bkruse_home | NEVER! |
| 05:29.02 | file | I have a PVR computer |
| 05:29.03 | file | :P |
| 05:29.19 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 05:29.27 | file | yay money |
| 05:29.51 | Juggie | rogers keeps adding new HD channels |
| 05:29.56 | Juggie | none of them perticularly intreasting |
| 05:30.00 | Juggie | but none the less we have like 30 now |
| 05:30.13 | Globetrotter | hey guys, can anyone point me in the right direction?? |
| 05:30.15 | file | ooh I found my music folders |
| 05:30.27 | Globetrotter | in getting my MOH to work?? |
| 05:31.51 | bkruse_home | Globetrotter: http://voip-info.org and search for music on hold |
| 05:32.05 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 05:32.13 | [TK]D-Fender | Globetrotter : Do you have mp123 installed? What version? |
| 05:32.16 | Globetrotter | thanks |
| 05:32.43 | Globetrotter | how do i check to see if its installed? |
| 05:33.10 | [TK]D-Fender | Globetrotter : "jsut type in mpg123 at your linux console |
| 05:33.55 | Globetrotter | command not found |
| 05:34.14 | Globetrotter | is it ok to install it on asterisk bussiness edition |
| 05:34.15 | Globetrotter | ? |
| 05:34.25 | [TK]D-Fender | Globetrotter : there's your problem. But forget about that for now and read up on Niative MoH. |
| 05:34.27 | [TK]D-Fender | ~book |
| 05:34.29 | jbot | methinks book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 05:34.30 | [TK]D-Fender | ~wikis |
| 05:34.31 | jbot | i heard wikis is http://www.voip-info.org |
| 05:35.17 | Globetrotter | thanks D-Fender |
| 05:36.04 | Juggie | the only benefit of mpg123 is one decode rather then one per caller on moh right? |
| 05:36.13 | Juggie | or does native moh deal w/ that now? |
| 05:39.22 | *** join/#asterisk xlnc2002 (n=root@203-97-110-166.cable.telstraclear.net) |
| 05:41.44 | *** join/#asterisk s1gny|wrk (n=s1gny@p5491623E.dip.t-dialin.net) |
| 05:41.58 | *** part/#asterisk s1gny|wrk (n=s1gny@p5491623E.dip.t-dialin.net) |
| 05:47.11 | *** part/#asterisk horsesgofaster (n=dcantera@pool-72-82-183-46.cmdnnj.east.verizon.net) |
| 05:48.42 | k-man_ | i'm thinking of getting an aastra voip phone |
| 05:48.48 | k-man_ | are they any good? |
| 05:50.50 | h0 | Has anyone noticed problems with losing connection to IRC servers (client xchat) when trying to make a call with X-Lite |
| 05:51.16 | h0 | i have had the problem on multiple systems |
| 05:52.31 | Qwell | h0: freenode doesn't allow sip calls while connected |
| 05:53.21 | h0 | ah ok probably other servers do the same thing because freenode is not the only one |
| 05:53.28 | h0 | do you know why they would do that |
| 05:55.03 | [TK]D-Fender | k-man_ : I'd only suggest the 280i |
| 05:55.22 | [TK]D-Fender | k-man_ : Though typically I'd suggest a Polycom IP 501 over it. |
| 05:55.29 | k-man_ | really? |
| 05:55.31 | k-man_ | interesting |
| 05:55.40 | k-man_ | you have used both? |
| 05:56.48 | [TK]D-Fender | k-man_ : yup |
| 05:57.08 | k-man_ | what is better about the polycom? |
| 05:58.05 | [TK]D-Fender | k-man_ : Polycom has a nicer screen for usability (pixel based instead of the 480i's char based), more solid body & weight, superior call handling & audio quality. |
| 05:58.21 | k-man_ | hmm... all good points |
| 05:58.23 | k-man_ | thanks |
| 05:59.00 | [TK]D-Fender | k-man_ : on the Aastra's side is their XML services (if you gfet around to coding them yourself) are functional, the screen is backlit, and the soft-keys are highly programmable. |
| 05:59.18 | k-man_ | hmmm |
| 05:59.49 | k-man_ | so the screen on the polycom is not backlit? |
| 05:59.49 | [TK]D-Fender | k-man_ : both are very good choices, but I prefer and suggest the Polycom |
| 05:59.50 | k-man_ | ok |
| 05:59.50 | k-man_ | thanks |
| 05:59.50 | [TK]D-Fender | k-man_ : nope |
| 05:59.59 | [TK]D-Fender | If backlight is a strong point for you the 480i isn't a bad choice. |
| 06:00.10 | k-man_ | no, its a minor point |
| 06:00.17 | k-man_ | our current phones don't have backlight |
| 06:00.25 | k-man_ | and its only 1 a year i wish it had |
| 06:00.41 | [TK]D-Fender | k-man_ : ? |
| 06:00.50 | k-man_ | sorry |
| 06:01.05 | k-man_ | i mean only once a year i wish it had backlight (rought estimate of frequency) |
| 06:02.03 | k-man_ | ever tried snom brand phones? |
| 06:02.22 | [TK]D-Fender | k-man_ : ICK. Craapy screen, flakey firmware. |
| 06:02.28 | k-man_ | ok |
| 06:02.32 | [TK]D-Fender | k-man_ : a nice idea poorly planned :) |
| 06:02.32 | k-man_ | just checking ;) |
| 06:03.57 | [TK]D-Fender | k-man_ : What kind of use & qty are tehse for? |
| 06:04.01 | [TK]D-Fender | these* |
| 06:04.29 | k-man_ | [TK]D-Fender, um... |
| 06:04.36 | k-man_ | at the moment i want one to experiment with |
| 06:04.44 | k-man_ | in the end we'll need about 10 i guess |
| 06:04.53 | k-man_ | but i want high end phones for me and my partner |
| 06:04.54 | [TK]D-Fender | k-man_ : Small office? |
| 06:04.57 | k-man_ | and lower ones for the users |
| 06:04.58 | k-man_ | yeah |
| 06:05.00 | k-man_ | small office |
| 06:05.11 | k-man_ | lots of intensive phone work though |
| 06:05.22 | k-man_ | order taking on the phone and sales calls |
| 06:05.25 | k-man_ | we need headsets |
| 06:05.25 | [TK]D-Fender | Ok, new plan : Polycom IP 430's for your low users, IP 601's for you and yourpartner. Get a PoE switch to run them all on. |
| 06:06.38 | k-man_ | 430's? |
| 06:06.39 | [TK]D-Fender | For headsets I'd recommend Plantronics M10 Aplifier w/ H261 Binaural Polaris headsets |
| 06:06.41 | k-man_ | still searching for that one |
| 06:07.00 | [TK]D-Fender | http://www.telephonydepot.com/Polycom_s/25.htm |
| 06:07.00 | k-man_ | i see you are experienced in this? |
| 06:07.18 | k-man_ | unfortunately polycom are poorly represented in australia |
| 06:07.25 | [TK]D-Fender | k-man_ : I'v got my call-center on those. I tried a lot of models before settling on those |
| 06:07.49 | robl^ | k-man_: take it from somone that loved Snom.. had several snom phones deployed and has recently begun a project to replace them ALL with Polycom. Polycom is the best I've seen. I've used Snom, Cisco, Grandstream, Aastra as well. I wish I had listened to people earlier |
| 06:08.05 | k-man_ | ok |
| 06:08.11 | k-man_ | thanks robl^ |
| 06:08.16 | k-man_ | hehe |
| 06:08.27 | k-man_ | so which should i get for initial testing and experimenting? |
| 06:08.33 | [TK]D-Fender | robl^ : perhaps some day they'll even things out and they'll be usable again... |
| 06:08.41 | [TK]D-Fender | k-man_ : the IP 430. |
| 06:08.55 | [TK]D-Fender | k-man_ : Start with the lower model you're planning on using the most |
| 06:09.03 | k-man_ | [TK]D-Fender, ah, good point |
| 06:09.06 | [TK]D-Fender | k-man_ : and leave room to decide on your high model. |
| 06:09.15 | [TK]D-Fender | k-man_ : keeps th initial budget low as well. |
| 06:09.26 | k-man_ | yeah |
| 06:09.28 | k-man_ | good idea |
| 06:09.31 | k-man_ | very good |
| 06:09.41 | k-man_ | what are its features that you selected that one for me? |
| 06:09.49 | [TK]D-Fender | k-man_ : Don't think that its a "cheap" phone. You can shuffle nearly a doze calls on it, has speakerphone, dual PoE & brink, the works. |
| 06:10.02 | [TK]D-Fender | dozen. |
| 06:10.36 | [TK]D-Fender | The IP430 is suggested because its 20$ cheaper than the 501, and has PoE capabilities built-in. Also has lighted line-keys which is kind nice. |
| 06:10.47 | [TK]D-Fender | Also a smaller frame which aids the space conscious. |
| 06:11.03 | Qwell | personally, I think the 501s are ugly |
| 06:11.07 | Qwell | the 430 isn't, so much |
| 06:11.22 | [TK]D-Fender | the 501 IS nicer for the screen, and offers another line key (unlikely to need though), and if youi WEREN'T planning on PoE I'd suggest that instead. |
| 06:11.37 | k-man_ | yeah |
| 06:11.39 | k-man_ | poe is a good idea |
| 06:11.44 | robl^ | the 430 and 601 are styled similarly. the 300/301s are being discontinued.. |
| 06:11.48 | k-man_ | i'd definately go that way |
| 06:12.01 | [TK]D-Fender | if your low end users have no need for speakerphone (they're on headsets so I'm thinkg speakerphone is a no-no anyways), and no PoE, you could save a bit more and get them IP 301's |
| 06:12.13 | [TK]D-Fender | robl^ : Annouced where? |
| 06:12.13 | *** join/#asterisk retrogradeorbit (n=crispin@203.63.120.237) |
| 06:12.32 | [TK]D-Fender | k-man_ : then IP 430 stays as your low model. |
| 06:12.37 | retrogradeorbit | Good day asterisk peeps |
| 06:12.40 | k-man_ | damn, the 430 is expensive here |
| 06:12.45 | bkruse_home | retrogradeorbit: what up |
| 06:13.08 | robl^ | [TK]D-Fender: I was old by my vendor that their supplies on the 301 has stopped and they had been told that Polycom may not be replinishing inventory |
| 06:13.09 | Qwell | k-man_: how much? |
| 06:13.12 | retrogradeorbit | I have a prob, thats similar to other stuff documented on the net but at the same time different! |
| 06:13.22 | retrogradeorbit | and this is it... |
| 06:13.39 | retrogradeorbit | starting "asterisk" everything runs fine. |
| 06:13.57 | retrogradeorbit | but there is no asterisk.ctl unix socket to run the console on |
| 06:14.01 | retrogradeorbit | but... |
| 06:14.19 | [TK]D-Fender | k-man_ : If budget is more of an issue than quality, Linksys is pretty decent down there. Just not sure about amps for your headsets. |
| 06:14.26 | retrogradeorbit | when I start asterisk -cvvvv, it again all starts fine, but now there *is* a asterisk.ctl file to connect to |
| 06:14.40 | [TK]D-Fender | k-man_ : More basic, and the LCD's are kinda sucky, but they'll do the job. |
| 06:15.11 | retrogradeorbit | my million dollar question is why does the -c make a .ctl file, but the normal mode not? |
| 06:15.27 | [TK]D-Fender | k-man_ : And believe me you WANT amplified headsets. having one direct plugged to aeither phone is pretty wimpy |
| 06:15.53 | retrogradeorbit | scouring the mailing lists the basic response is try the -c flag and see where is fails on startup. The problem is mine doesn't fail. it starts fine. |
| 06:17.00 | retrogradeorbit | if anyone has any ideas on where to look from here, fire em my way! |
| 06:17.16 | retrogradeorbit | Im presently building strace onto the platform so I can have a bit of a burrow in... :) |
| 06:18.05 | k-man_ | [TK]D-Fender, we currently have an NEC Xen system and don't have amps for the headsets |
| 06:18.29 | k-man_ | qwell, the 430 is around $355 |
| 06:18.35 | [TK]D-Fender | k-man_ : OUCH |
| 06:18.39 | Qwell | AUD? |
| 06:18.45 | Qwell | That's still ridiculous |
| 06:18.56 | `Sean | not that bad |
| 06:18.56 | `Sean | :P |
| 06:18.59 | [TK]D-Fender | Qwell : yeah, Polycom is only great in NA |
| 06:19.33 | [TK]D-Fender | k-man_ : Like I said, Linksys may be the choice for you, but try *1* and see. |
| 06:19.41 | k-man_ | yeah AUD |
| 06:20.00 | `Sean | Hrmp |
| 06:20.01 | k-man_ | i could get them from the US |
| 06:20.07 | k-man_ | how much is a 430 in the US? |
| 06:20.16 | `Sean | ive still not decided wich polycom phone i want |
| 06:20.18 | [TK]D-Fender | k-man_ : as linked, $150 USD |
| 06:20.19 | `Sean | im still thinking.. |
| 06:20.44 | k-man_ | hmmm |
| 06:21.45 | k-man_ | [TK]D-Fender, yeah, thats much cheaper than here |
| 06:22.16 | k-man_ | ~AU$191 |
| 06:22.23 | k-man_ | plus shipping... |
| 06:22.32 | k-man_ | still a lot less than AU$355 |
| 06:22.39 | [TK]D-Fender | k-man_ : import duty? |
| 06:22.48 | k-man_ | 10% GST |
| 06:22.59 | Qwell | plus probably $30 for shipping |
| 06:23.06 | Qwell | so, still < $250 |
| 06:23.21 | Qwell | k-man_: sounds like you need to become a reseller, heh |
| 06:23.24 | k-man_ | AU$355 ~= 277 us |
| 06:23.27 | Qwell | re-reseller, as it were |
| 06:23.54 | k-man_ | yeah |
| 06:24.02 | k-man_ | i wonder why polycom is so under represented here |
| 06:24.17 | Qwell | Do your phones ring in backward phase there? |
| 06:24.18 | k-man_ | strangely there is a lot of the polycom conferencing gear here |
| 06:24.24 | Qwell | that might be why - they just don't support it |
| 06:24.33 | k-man_ | qwell, nfi, sorry |
| 06:24.36 | retrogradeorbit | the linksys spa941 is quite nice |
| 06:24.44 | Qwell | k-man_: that was a southern hemisphere joke |
| 06:24.52 | k-man_ | qwell, i expect we have the same as the UK, but thats just a guess |
| 06:25.04 | k-man_ | qwell, oh right... i get it. |
| 06:25.09 | [TK]D-Fender | I'd ammend that the SPA941 is PASSABLE. , though far from "nice". |
| 06:25.24 | retrogradeorbit | whats people recomendation for a SIP phone |
| 06:25.34 | k-man_ | a friend has linksys at his work |
| 06:25.42 | k-man_ | and he bought a polycom off ebay, and says its heaps better |
| 06:25.45 | retrogradeorbit | not wanting to pay the sip firmware cost on cisco IP phones |
| 06:26.05 | retrogradeorbit | tried the polycom. it was ok. |
| 06:26.33 | retrogradeorbit | the sidecars were a nice feature |
| 06:26.54 | hads | The problem is you need to handle the warranty yourself if you import your own directly, that and most distributors won't sell outside their area so you have to buy off a reseller. |
| 06:27.31 | [TK]D-Fender | ok, way past my bed-time. Later all... |
| 06:29.33 | k-man_ | oh |
| 06:29.47 | k-man_ | thanks [tk] |
| 06:29.50 | k-man_ | night |
| 06:29.52 | k-man_ | bit slow |
| 06:30.18 | *** join/#asterisk dlynes_laptop (n=dlynes@S0106001346f7843f.vc.shawcable.net) |
| 06:30.38 | k-man_ | what is the status of sending faxes over voip? |
| 06:36.22 | *** join/#asterisk sumasuma (n=kurukko@61.14.86.23) |
| 06:42.15 | *** join/#asterisk intralanman (n=lanman@pool-71-253-204-107.nrflva.east.verizon.net) |
| 06:49.34 | *** join/#asterisk tsurko (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 06:52.11 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
| 06:55.46 | BugKhaM | where can I see the list of bugs or problems found in asterisk 1.2.14? |
| 06:57.00 | russellb | the changelog in the release is all that exists for that |
| 06:58.01 | *** join/#asterisk stephane_ (n=stephane@zafnib.cabale.net) |
| 07:01.06 | *** part/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 07:01.14 | retrogradeorbit | re: my problem. Turns out if I "asterisk -fp" it makes the .ctl unix domain socket |
| 07:01.32 | retrogradeorbit | but if I just run "asterisk" so it daemonises, I get no .ctl socket |
| 07:01.40 | retrogradeorbit | asterisk version 1.2.13 |
| 07:01.45 | retrogradeorbit | is this a known bug? |
| 07:02.31 | *** join/#asterisk alamantia (n=Anthony@65.4.24.231) |
| 07:03.46 | BugKhaM | russellb: what about after the release? |
| 07:04.26 | *** join/#asterisk Rhizome (n=Rhizome@81.191.147.145) |
| 07:05.20 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 07:05.27 | *** join/#asterisk rbrindley (n=ryan@user-69-73-26-211.knology.net) |
| 07:07.16 | hads | BugKhaM: use the subversion log |
| 07:07.30 | *** join/#asterisk zantafio (n=pwariche@c-67-188-69-103.hsd1.ca.comcast.net) |
| 07:08.42 | BugKhaM | hads: okay |
| 07:09.54 | russellb | there is a changelog in the tarball ... |
| 07:10.08 | russellb | http://svn.digium.com/view/asterisk/tags/1.2.14/ChangeLog |
| 07:17.34 | tzafrir | retrogradeorbit, but an Asterisk process was generated? |
| 07:17.53 | tzafrir | Are you running asterisk as root? |
| 07:18.34 | `Sean | i was reading earlier it says its recommended to run asterisk as a USER not as ROOT |
| 07:21.09 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 07:24.10 | retrogradeorbit | tzafrir, yes. running as root. process generated |
| 07:24.31 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 07:24.50 | retrogradeorbit | without -fp, asterisk runs 100% but no .ctl file generated, and thus future asterisk -r's fail |
| 07:25.14 | retrogradeorbit | with -fp, asterisk runs 100% and .ctl file IS generated, and future asterisk -r's succeed |
| 07:25.29 | bkruse_home | file: i am going to sleep! |
| 07:25.37 | *** part/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 07:25.39 | tzafrir | you can try running it with strace -f to follow its actions |
| 07:25.50 | retrogradeorbit | just finished that |
| 07:25.50 | tzafrir | good night |
| 07:25.56 | *** join/#asterisk atif_ (n=atif@202.92.17.40) |
| 07:26.03 | retrogradeorbit | the only reference to asterisk.ctl is as follows |
| 07:26.25 | atif_ | hello all, folks how I can check which asterisk module is consuming how much memory ??? |
| 07:26.33 | atif_ | any idea ? |
| 07:26.37 | retrogradeorbit | stat("/etc/asterisk/extconfig.conf", 0xbfdb7470) = -1 ENOENT (No such file or directory) |
| 07:26.37 | retrogradeorbit | socket(PF_FILE, SOCK_STREAM, 0) = 4 |
| 07:26.37 | retrogradeorbit | connect(4, {sa_family=AF_FILE, path="/var/run/asterisk.ctl"}, 110) = -1 ENOENT (No such file or directory) |
| 07:26.37 | retrogradeorbit | close(4) = 0 |
| 07:26.37 | retrogradeorbit | unlink("/var/run/asterisk.pid") = 0 |
| 07:27.19 | retrogradeorbit | and thats all. Sorry bout the miniflood. figured 5 lines would be safe ;) |
| 07:27.36 | tzafrir | atif_, look at the memory maps of the astrerisk process. But those can be misleading, as there are plenty ofover-allocations |
| 07:27.48 | retrogradeorbit | atif_, you may need something like valgrind |
| 07:27.57 | atif_ | yeah, but it will overload the system |
| 07:27.59 | tzafrir | retrogradeorbit, any chance it thinks it is asterisk -r ? |
| 07:28.06 | atif_ | its live system |
| 07:28.24 | atif_ | /proc/PID/status gives total memory being consumed by asterisk |
| 07:28.43 | tzafrir | what I mean is, that the value you see as the ammount of memory allocated by Asterisk may not all actually be allocated. |
| 07:28.48 | atif_ | but need to check which module is leaking or consuming the most,,,,as current mem is above 1 GB :) |
| 07:28.54 | atif_ | hmm, i c |
| 07:28.57 | tzafrir | Generally allocated memory is only actually allocated on first use |
| 07:29.35 | atif_ | but asterisk is requesting dynamically as well |
| 07:29.43 | tzafrir | retrogradeorbit, if you start asterisk -fp in one terminal, and then run 'asterisk' in another terminal, will it try to connect to the existing copy? |
| 07:29.51 | retrogradeorbit | yeh |
| 07:30.00 | atif_ | most, I think all applications use dynamic allocation model |
| 07:30.02 | tzafrir | Also: do you set Asterisk to run as a non-root user in asterisk.conf? |
| 07:30.37 | tzafrir | atif_, this is actually linux playing dirty tricks there |
| 07:31.00 | retrogradeorbit | asterisk is running as root |
| 07:31.23 | retrogradeorbit | asterisk -r connects to the "asterisk -fp" process |
| 07:31.43 | atif_ | hmmm |
| 07:32.46 | atif_ | but I have some other asterisk machines as well,,,they never went above 25-30 MB.....asterisk at this machine is growing unboundedly |
| 07:33.02 | atif_ | like as I said, current memory stats are 1.4 GB being using by asterisk |
| 07:33.17 | atif_ | *being used by asterisk |
| 07:33.43 | retrogradeorbit | wow. thats a lot |
| 07:33.57 | retrogradeorbit | what modules do you have loaded, atif_? |
| 07:34.43 | atif_ | hmmm, not much....instead running with very limited necessary modules |
| 07:34.46 | atif_ | shud I list them |
| 07:34.58 | retrogradeorbit | tzafrir, when I run "asterisk" in another term, it doesnt connect to the existing "asterisk -fp" it generates a new asterisk |
| 07:35.24 | retrogradeorbit | well I suppose it needs to be one of the ones loaded. As long as theres not too many |
| 07:35.35 | retrogradeorbit | you could try a process of elimination if its possible |
| 07:35.59 | tzafrir | does it create a pid file? |
| 07:36.19 | tzafrir | <PROTECTED> |
| 07:38.37 | retrogradeorbit | yes |
| 07:39.06 | retrogradeorbit | thats the wierd bit. Everything works 100%. Im making calls through it and all. just no .ctl |
| 07:39.29 | retrogradeorbit | unless I go "-fp". Then it all works, *and* there is a .ctl |
| 07:39.37 | retrogradeorbit | its quite strange... |
| 07:43.47 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 07:45.01 | tzafrir | can you post/pastebin the full strace somewhere? |
| 07:45.40 | retrogradeorbit | aye. I could. wheres the pastebin for this channel? |
| 07:45.54 | atif_ | ~pb |
| 07:45.56 | jbot | methinks pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 07:49.32 | atif_ | tzafrir & retro: here are loaded modules in my asterisk |
| 07:49.34 | atif_ | http://pastebin.ca/310008 |
| 07:51.42 | atif_ | shud I ask this question in -dev ??? |
| 07:52.28 | tzafrir | atif_, why exactly do you ask this? do you suspect a memory leak? |
| 07:53.09 | nitrus^ | is anyone having queue issues with 1.4? i setup queues and have member = Agent/200 and in my users.config i have [200] hasagent=yes but it isnt dialing the Zap channel it just sits there |
| 07:54.31 | tzafrir | nitrus^, users.conf? |
| 07:54.48 | nitrus^ | yeah asterisk-gui writes that crap out |
| 07:55.00 | nitrus^ | oops |
| 07:55.03 | nitrus^ | yeah .conf |
| 07:55.10 | nitrus^ | i didnt realize i typoed, sorry |
| 07:55.41 | atif_ | yeah, definitely its a memory leak....just want to know which module is causing this .... |
| 07:55.42 | tzafrir | in the [200] entry, do you have zapchan and/or channel? What values do you see there? |
| 07:56.06 | nitrus^ | zapchan = 13 |
| 07:56.46 | nitrus^ | i can dial the ext directly just fine |
| 07:56.51 | retrogradeorbit | asterisk starting but without .ctl strace: http://channels.debian.net/paste/5036 |
| 07:56.54 | tzafrir | and when you run 'zap show channels' do you see channel 13? |
| 07:57.13 | nitrus^ | yep |
| 07:58.08 | tzafrir | next: cli trace. set core verbose 3 |
| 07:58.15 | tzafrir | and see what happens when you dial |
| 07:59.17 | nitrus^ | it just sits in the queue |
| 07:59.20 | nitrus^ | no dialing occurs |
| 07:59.22 | nitrus^ | no errors |
| 07:59.29 | nitrus^ | when i close it says "almost made it" |
| 08:00.49 | nitrus^ | if i edit the queues.conf and add the agent = Zap/13 it works fine |
| 08:00.53 | *** part/#asterisk Deciphan (n=Erick@c-24-7-130-249.hsd1.ca.comcast.net) |
| 08:01.09 | nitrus^ | i may just have to forget this whole new config thing and go back to how i used to have it |
| 08:01.30 | *** join/#asterisk Nobbie (n=no@fwb003.fw.is.co.za) |
| 08:01.34 | tzafrir | nitrus^, are you sure that this is 1.4? |
| 08:01.42 | Nobbie | heya =) |
| 08:01.44 | nitrus^ | Asterisk 1.4.0, Copyright (C) 1999 - 2006 Digium, Inc. and others. |
| 08:01.50 | nitrus^ | after asterisk -r |
| 08:01.51 | tzafrir | retrogradeorbit, oops, that was for you, not nitrus^ |
| 08:02.31 | retrogradeorbit | strace asterisk -r, as well? |
| 08:02.49 | retrogradeorbit | or -fp? |
| 08:04.28 | tzafrir | retrogradeorbit, have you noticed it writes an incorrect PID to the pid file? |
| 08:04.33 | tzafrir | in the -fp trace |
| 08:04.39 | nitrus^ | anyone have a link for the users.conf format? |
| 08:06.17 | tzafrir | nitrus^, was the agent generated? The zaptel channel was indeed generated |
| 08:06.39 | nitrus^ | good question |
| 08:06.42 | nitrus^ | lemmie check |
| 08:06.45 | tzafrir | As I suggested, looking at the actual errors from the trace may be more useful |
| 08:07.02 | nitrus^ | yeah they are |
| 08:07.06 | nitrus^ | but they arent logged in |
| 08:07.08 | nitrus^ | that's why |
| 08:07.18 | nitrus^ | however theyre on zap channels, so there is no login |
| 08:07.23 | *** join/#asterisk lorinc (n=ang@213.178.120.186) |
| 08:08.47 | tzafrir | retrogradeorbit, hmmm... just to eliminate suspects, if you drop the option -p, nothing changes, right? |
| 08:08.49 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 08:09.28 | *** join/#asterisk Bhaal (i=bhaal@freenode/unconfirmed/bhaal) |
| 08:09.55 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 08:10.16 | tzafrir | retrogradeorbit, also: where do you run asterisk from? /usr/sbin/asterisk ? |
| 08:13.06 | tzafrir | Also: do you have nofork or alwaysfork set in the options section of asterisk.conf ? |
| 08:14.19 | retrogradeorbit | ok. run it from /usr/sbin/asterisk |
| 08:14.36 | retrogradeorbit | let me try the -p droppage |
| 08:15.22 | retrogradeorbit | aye! your suspicians are correct. running "asterisk -f" also creates the .ctl file |
| 08:17.03 | retrogradeorbit | checking the asterisk.pid file it is always correct, whether running -f, or not -f |
| 08:18.42 | *** join/#asterisk alphaque (n=alphaque@60.51.217.61) |
| 08:20.00 | retrogradeorbit | I have neither nofork, nor alwaysfork set. But I DO have this " load => app_forkcdr.so" in modules.conf |
| 08:21.30 | *** join/#asterisk UlbabraB (n=salama@88-149-155-155.f5.ngi.it) |
| 08:22.05 | *** join/#asterisk Nobbie (n=no@fwb003.fw.is.co.za) |
| 08:22.34 | Nobbie | has anyone tried the autofill=yes option in queues.conf in * 1.4.0 ? comments on the feature ? |
| 08:23.46 | nitrus^ | how do you auto login agents? |
| 08:24.03 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust337.leed.cable.ntl.com) |
| 08:27.24 | tzafrir | hmmm.... fun with main/asterisk.c |
| 08:28.13 | tzafrir | anybody here with asterisk 1.4? I suspect that the following would generate a segfault: ln -s /usr/sbn/asterisk a; ./a |
| 08:28.32 | tzafrir | because argv[0] has less than 10 characters. |
| 08:30.22 | retrogradeorbit | hard coded buffer? |
| 08:32.08 | *** part/#asterisk mav3rick (n=t0m@trankil.biz) |
| 08:32.08 | *** join/#asterisk Nobbie (n=no@fwb003.fw.is.co.za) |
| 08:32.33 | Nobbie | any takers on the autofill=yes queues option which i missed ? anyone tried it and found it useful ? |
| 08:34.30 | tzafrir | argv holds the program name and its parameters. |
| 08:34.58 | tzafrir | retrogradeorbit, What version do you have exactly? 1.4.0 from tarball? |
| 08:37.39 | Juggie | oh god how depressing |
| 08:37.48 | Juggie | my new mortgage just showed up in my online banking |
| 08:37.56 | Juggie | now shows me 185,000 in debt. |
| 08:37.58 | Juggie | awesome :) |
| 08:38.19 | tzafrir | retrogradeorbit, your strace does not seem to match the code. Any chance you have some local patches, specifically to main/asterisk.c ? |
| 08:38.23 | retrogradeorbit | from tarball 1.2.13 |
| 08:38.41 | tzafrir | hmm... why did I think it is 1.4? |
| 08:38.57 | retrogradeorbit | heh. I think thats nitrus |
| 08:39.18 | retrogradeorbit | its crosscompiled, Ill check the patch status |
| 08:43.56 | tzafrir | hmm... my mistake. It writes twice to the pid file |
| 08:45.36 | *** join/#asterisk uwe (n=uwe@dogbert.palnet.com) |
| 08:52.53 | *** join/#asterisk reber (i=reber@gateway/tor/x-c33dba66f987bc38) |
| 08:58.24 | *** join/#asterisk niZon (i=bleh@S0106beefd4cecc3d.wp.shawcable.net) |
| 08:58.53 | *** join/#asterisk inspired (n=mikael@85.221.7.59) |
| 08:59.40 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 09:00.51 | *** join/#asterisk Ahrimanes (n=ma@81.7.159.2) |
| 09:01.54 | *** join/#asterisk rajiv|wo1k (n=rajiv@gentoo/developer/rajiv) |
| 09:04.50 | *** join/#asterisk oQPa (n=roque@198.Red-83-41-117.dynamicIP.rima-tde.net) |
| 09:07.27 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 09:08.51 | nitrus^ | is there a variable that contains the extension being called from, say i call voicemailmain from my phone which has extension 200 is there a way to pass voicemailmain the exten i called from? |
| 09:09.06 | *** join/#asterisk santibiotico (n=santi@101.Red-83-58-114.dynamicIP.rima-tde.net) |
| 09:10.35 | *** join/#asterisk sasch (n=sasch@82.107.30.102) |
| 09:10.46 | Ahrimanes | nitrus^, CALLERID(num) ? |
| 09:10.50 | sasch | hi all |
| 09:11.36 | nitrus^ | that could work |
| 09:11.42 | nitrus^ | ill give it a try |
| 09:11.49 | sasch | i have a question for asterisk-gui |
| 09:11.59 | nitrus^ | my caller id numbers arent my extensions though |
| 09:12.05 | nitrus^ | they're my actually lines |
| 09:12.15 | sasch | hummm i look #asterisk-gui |
| 09:12.16 | sasch | excusme |
| 09:17.28 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 09:20.04 | *** join/#asterisk conver2 (n=chatzill@206-248-153-59.dsl.teksavvy.com) |
| 09:20.30 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 09:28.30 | *** join/#asterisk A-Tuin|work (n=A-Tuin@212.41.185.81) |
| 09:38.28 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 09:38.54 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 09:39.09 | bkruse_home | file: i cannot sleep ;[ |
| 09:39.25 | *** join/#asterisk Aurs (n=Aurs@81.191.123.189) |
| 09:40.13 | Makenshi | bkruse_home, i suffer from insomnia, and i take pills for it, but recently i have given hypnosis a try.. i was sceptical but it does seem to have a positive effect |
| 09:40.54 | bkruse_home | Makenshi: you telling me this at 3:30 in the morning does not make your story believable :P |
| 09:41.28 | Makenshi | bkruse_home, it is 0941 and i have been at work for 2 and a half hours already |
| 09:41.57 | bkruse_home | Makenshi: nice, i might have to try that |
| 09:43.18 | `Sean | !seen shellshakr |
| 09:43.26 | `Sean | !seen shellshark |
| 09:43.55 | `Sean | Hrmp, I'm bored |
| 09:44.30 | bkruse_home | `Sean: mess with dundi |
| 09:45.17 | Ahrimanes | or.. work on shared registration info in asterisk.. |
| 09:45.42 | bkruse_home | Ahrimanes: aka sip express router? |
| 09:46.26 | Ahrimanes | bkruse_home, hm havent looked that much at SER.. but would be nice if you could register to one Asterisk and then later on fail over to another asterisk without needing to re-register |
| 09:46.43 | bkruse_home | Ahrimanes: precisely what im working on now |
| 09:47.08 | Ahrimanes | bkruse_home, cool, hows it comming? talked to oej about it a while ago |
| 09:47.16 | bkruse_home | good good |
| 09:47.25 | Ahrimanes | ready for 1.4.1 ? ;) |
| 09:47.26 | bkruse_home | moreso documentation how to do so, problems I encounter, and how to setup a distributed call centers with failover and load balancing |
| 09:47.30 | bkruse_home | :X |
| 09:47.35 | Ahrimanes | ah ok |
| 09:47.50 | Ahrimanes | anyone sponsoring this work? |
| 09:47.55 | bkruse_home | uh |
| 09:47.57 | bkruse_home | digium! |
| 09:47.59 | Ahrimanes | ah |
| 09:48.01 | Ahrimanes | :) |
| 09:48.02 | bkruse_home | not really |
| 09:48.09 | bkruse_home | its not something i work on, on the clock |
| 09:48.18 | Ahrimanes | hm |
| 09:48.20 | bkruse_home | just a side project that i think will be super useful to the community |
| 09:48.24 | Ahrimanes | you need to ;) |
| 09:48.52 | bkruse_home | all of our brain power is going to working on core asterisk, which is great! but what about solutions for the enterprise type situation you know? |
| 09:49.13 | Ahrimanes | if you end up having a time estimate or something like it, do send, i might be able to fund some work |
| 09:49.26 | bkruse_home | that would be great |
| 09:49.38 | bkruse_home | nothing better to learn then to experiment and implement right? |
| 09:49.45 | Ahrimanes | my current employer needs this functionality |
| 09:49.54 | bkruse_home | hmm.....interesting :] |
| 09:50.03 | Ahrimanes | yup |
| 09:50.04 | *** join/#asterisk Newbie___ (n=me@211.24.146.11) |
| 09:50.08 | bkruse_home | i think i can do EVERYTHING but....sip session holdup |
| 09:50.19 | bkruse_home | as in, if an asterisk box goes down that your talking too...if you can keep up this call...... |
| 09:50.20 | Ahrimanes | holdup? |
| 09:50.23 | bkruse_home | sip session handling rahter* |
| 09:50.26 | bkruse_home | rather* |
| 09:50.27 | Ahrimanes | not loosing the call? |
| 09:50.50 | bkruse_home | right |
| 09:50.52 | Ahrimanes | ah |
| 09:50.55 | Newbie___ | hi all, i intend to use 2 network cards in 1 asterisk box connected to 2 different ISP and terminate calls using SIP, any idea ? |
| 09:51.03 | bkruse_home | at the very least...the functionality will be right back up instantly |
| 09:51.04 | Ahrimanes | well oej should be helpful on the sip parts |
| 09:51.19 | bkruse_home | Ahrimanes: ya i need to talk to ole oej |
| 09:51.28 | bkruse_home | hes so busy these days though |
| 09:51.39 | bkruse_home | but the solutions ive "blueprinted" and labbed up are pretty freaking cool |
| 09:51.40 | Ahrimanes | bkruse_home, i heard he knows a thing or two about sip :) |
| 09:51.47 | bkruse_home | Ahrimanes: indeed he does |
| 09:51.59 | Ahrimanes | yeah, he's way busy.. he owes me some quotes on some work |
| 09:52.10 | bkruse_home | ha, oej knows whats up though |
| 09:52.13 | bkruse_home | hes got alot of stuff on his plate |
| 09:52.34 | bkruse_home | his name is always poppin up into things i hear around the latest talk at digium |
| 09:52.39 | Ahrimanes | hehe |
| 09:52.47 | Ahrimanes | he's a smart fella |
| 09:52.52 | Ahrimanes | met him a few times |
| 09:52.52 | bkruse_home | but ive got some things drawn up using mysql for storing and replications and mass voicemail storage and all that good stuff |
| 09:53.03 | bkruse_home | ill meet him soon enough |
| 09:53.27 | Ahrimanes | hm, wont mysql be slow for shared registrations? |
| 09:53.32 | bkruse_home | NO! |
| 09:53.35 | bkruse_home | not at all |
| 09:53.39 | `Sean | bkruse_home why mess with dundi |
| 09:53.40 | bkruse_home | its MYSQL in a nbd cluster |
| 09:53.42 | `Sean | i have no use for it tho |
| 09:53.46 | bkruse_home | `Sean: its good to know |
| 09:53.54 | `Sean | hrmp yea youre right |
| 09:53.55 | Ahrimanes | bkruse_home, ew ndb... no foreign keys etc |
| 09:54.11 | bkruse_home | Ahrimanes: you can throw your registrations into a self replicating cluster |
| 09:54.23 | bkruse_home | and SER has mysql support built in |
| 09:54.27 | bkruse_home | kind of...... |
| 09:54.30 | Ahrimanes | yeah |
| 09:54.39 | Ahrimanes | but how often will it hit the database? |
| 09:54.42 | bkruse_home | but you can do some REALLY cool things with that though.... |
| 09:54.49 | bkruse_home | Ahrimanes: for registration lookups |
| 09:55.00 | Ahrimanes | yeah.. but how often is that? |
| 09:55.16 | bkruse_home | and its ALL loaded into ram (inside the cluster) if your not fimiliar with mysql clusters i can throw down a quick demonstration of what it looks like |
| 09:55.32 | bkruse_home | Ahrimanes: not sure, depends on how often SER looks things up, which can probably be tweaked |
| 09:55.49 | Ahrimanes | bkruse_home, well i'm thinking for asterisk.. |
| 09:55.53 | bkruse_home | I imagine most of the registrations can be stored into ram on the SER server all.. |
| 09:56.02 | Ahrimanes | database load is a big concern for me |
| 09:56.13 | bkruse_home | Ahrimanes: I dont think it will be if we throw it all into ram |
| 09:56.18 | bkruse_home | into that cluster rather. |
| 09:56.33 | Ahrimanes | well ndb is not an option for me unfortunatel |
| 09:56.34 | Ahrimanes | y |
| 09:56.37 | bkruse_home | Ahrimanes: some of the sun boxes can handle 500,000 queries per second :D |
| 09:56.45 | bkruse_home | Ahrimanes: then we could find something else, no big |
| 09:56.57 | Ahrimanes | ok |
| 09:57.20 | Ahrimanes | database without constraints is just not that useful, hehe |
| 09:57.25 | bkruse_home | Ahrimanes: well, lets see, as far as asterisk lookups go, I am pretty sure that SER does all the sip authentication, so asterisk should only have to register to SER |
| 09:57.34 | bkruse_home | Ahrimanes: agreed |
| 09:57.45 | Ahrimanes | bkruse_home, hm... well i'd rather see a pure asterisk setup |
| 09:58.22 | bkruse_home | Ahrimanes: there are SOME solutions out there |
| 09:58.33 | Ahrimanes | yeah |
| 09:58.40 | bkruse_home | i would like to see that sort of functionality built into a pure asterisk setup also, which i think will be in the future. |
| 09:58.45 | Ahrimanes | but i havent found any suitable for me yet |
| 09:59.50 | bkruse_home | hmm |
| 09:59.53 | bkruse_home | i can do some research |
| 10:00.13 | bkruse_home | so what are you interested in mainly? |
| 10:00.32 | bkruse_home | load balancing for a huge implementation, or failover for your current setup, or both |
| 10:00.40 | Ahrimanes | clustering basically.. we need failover for our customers |
| 10:00.54 | bkruse_home | theres some really cool things we could do |
| 10:01.02 | Ahrimanes | load balancing is not needed yet... maybe later this year |
| 10:01.13 | bkruse_home | even failover switches (hardware) for PRI's and PSTN terminations and such |
| 10:01.23 | bkruse_home | well when you say "clustering" what do you mean |
| 10:01.28 | bkruse_home | because asterisk "clustering' |
| 10:01.37 | bkruse_home | is essentially load balancing calls |
| 10:01.43 | Ahrimanes | for now failover |
| 10:02.06 | bkruse_home | Ahrimanes: I have some cool solutions for that :] |
| 10:02.19 | Ahrimanes | yup |
| 10:02.35 | *** join/#asterisk reber (i=reber@gateway/tor/x-f2875b66c9fd789c) |
| 10:02.39 | Ahrimanes | but "all" i need right now is registration information shared between asterisk's |
| 10:03.26 | bkruse_home | sharing your sip configs? |
| 10:03.35 | Ahrimanes | the rest can be done with round-robin dns and vserver |
| 10:03.48 | Ahrimanes | no, then the phones have to re-register when switching asterisk |
| 10:03.48 | bkruse_home | right |
| 10:03.51 | Ahrimanes | not good enough |
| 10:03.57 | bkruse_home | gotcha |
| 10:04.07 | bkruse_home | are you using some kind of heartbeat? |
| 10:04.18 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 10:04.22 | bkruse_home | to control your dns switch? |
| 10:04.27 | Ahrimanes | nope |
| 10:04.30 | bkruse_home | eww..... |
| 10:04.33 | Ahrimanes | we'd just switch the ip |
| 10:04.47 | bkruse_home | how do you know when a system goes down? |
| 10:04.58 | Ahrimanes | i have lots of monitoring |
| 10:05.18 | bkruse_home | i feel like were flooding #asterisk (even though theres no one here) can you PM me? |
| 10:05.19 | Ahrimanes | that's not a problem.. problem is phone re-register |
| 10:05.25 | Ahrimanes | heeh sure |
| 10:11.02 | *** join/#asterisk k-man__ (n=jason@ppp244-232.static.internode.on.net) |
| 10:13.48 | *** join/#asterisk mega (n=mega@217.201.144.180) |
| 10:14.58 | `Sean | bkruse you know anything about callerid unblocker? |
| 10:15.13 | bkruse_home | `Sean: unblocker? watcha mean |
| 10:15.19 | `Sean | pm |
| 10:27.54 | *** join/#asterisk Newbie___ (n=me@211.24.146.11) |
| 10:28.35 | *** join/#asterisk FaithX (n=FaithX@ns.linuxterminal.com) |
| 10:28.42 | Newbie___ | hi all, any ideas on if using 2 network card in 1 asterisk box to send calls using 2 different ISP with SIP. |
| 10:31.52 | DrCron | Newbie___, it would probably be easiest to set that up in your OS rather then in asterisk |
| 10:32.19 | DrCron | ie: all traffic to server 1 goes through interface a, all trafic to server 2 goes through interface b |
| 10:34.16 | Newbie___ | DrCron: how do i tell * which network card to go ? say 1. goes to eth0 and 011 goes to eht1 |
| 10:34.19 | Newbie___ | eth1 |
| 10:35.16 | DrCron | it connects to diffrent servers? |
| 10:35.31 | DrCron | what operating system are you using? |
| 10:36.03 | Newbie___ | whotebox EL4 |
| 10:36.06 | Newbie___ | whitebox |
| 10:36.44 | Newbie___ | is there something in sip.conf that says, for 1. go to eth0 and 011 goes to eth1 |
| 10:37.49 | Newbie___ | sorry, i mean extensions.conf |
| 10:37.55 | DrCron | not to my knowledge |
| 10:38.09 | DrCron | asterisk has no idea of networking cards |
| 10:38.19 | DrCron | you need to use your OS for that |
| 10:38.26 | `Sean | yea |
| 10:38.34 | `Sean | asterisk is not a routing program |
| 10:38.36 | `Sean | its a PBX |
| 10:38.39 | Newbie___ | DrCron: ok thanks, i guess is google time |
| 10:39.06 | bkruse_home | `Sean: agreed, rahter you should use operating system features |
| 10:39.11 | bkruse_home | s/rahter/rather/ |
| 10:45.05 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 10:45.08 | *** part/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 10:50.44 | *** join/#asterisk Nobbie (n=no@fwb003.fw.is.co.za) |
| 10:51.13 | Nobbie | has anyone had an issue with a SNOM360 operator phone where the asterisk queue shows the phone as ringing, but the actual phone doesn't ring ? |
| 10:57.31 | *** join/#asterisk phearless (n=phear@host81-138-68-106.in-addr.btopenworld.com) |
| 10:59.48 | *** join/#asterisk jgoo (n=e4b80e21@foodtecsolutions.com) |
| 11:00.09 | jgoo | hey people, who uses soundpoint 501's? just some stupid questions regarding power |
| 11:00.28 | jgoo | I want to make sure as like linksys, powercom love to use 23 different power supplies |
| 11:00.43 | jgoo | does the 501 use a 12V 400mA adapter? |
| 11:02.48 | Mavvie | Rotated Logs Per SIGXFSZ (Exceeded file size limit) |
| 11:02.54 | Mavvie | that's the first time I saw that message. |
| 11:04.24 | sasch | i have one problem with asterisk and squillo trunk |
| 11:04.32 | sasch | when I start asterisk say |
| 11:04.41 | sasch | NOTICE[6181]: chan_sip.c:7085 sip_reg_timeout: -- Registration for '63759@sip.squillo.it' timed out, trying again (Attempt #2) |
| 11:05.15 | jgoo | just to bump my request, if anyone here has a soundpoint 501, can you check the power rating on the adapter? Just want to be 100% |
| 11:11.53 | Ahrimanes | hm i might have on |
| 11:11.54 | Ahrimanes | e |
| 11:12.43 | Ahrimanes | oh we do Power over ethernet |
| 11:12.51 | jgoo | :p |
| 11:12.52 | ThoMe | Hi. |
| 11:12.56 | ThoMe | Kann hier wer deutsch? ;) |
| 11:14.59 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 11:15.14 | *** join/#asterisk reber (i=reber@gateway/tor/x-954842f0eb1f82dc) |
| 11:15.20 | *** join/#asterisk angryuser (n=Miranda@i03v-213-44-169-43.d4.club-internet.fr) |
| 11:16.02 | *** join/#asterisk emiquelito (n=evandro@200-155-185-1.static.spo.ifx.net.br) |
| 11:16.06 | tzafrir | ThoMe, try #asterisk-de , though it's not particularily crowded right now |
| 11:16.36 | ThoMe | tzafrir: Hui, 3 people;) |
| 11:16.49 | tzafrir | you are the third, right? |
| 11:17.14 | *** join/#asterisk Cae2 (n=none@gw.curitibaonline.com.br) |
| 11:17.18 | ThoMe | tzafrir: jep ;) |
| 11:18.36 | DrCron | have any of you run a sip or iax client on a pocket pc? |
| 11:18.55 | DrCron | i am trying to figure out the minimus recomended specs |
| 11:19.01 | DrCron | minimum even |
| 11:19.01 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 11:20.18 | *** join/#asterisk p0g0 (n=pogo@madwifi/support/p0g0) |
| 11:21.27 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 11:29.46 | *** join/#asterisk emiquelito (n=evandro@200-155-185-1.static.spo.ifx.net.br) |
| 11:34.11 | angryuser | well i have used sjphone over-wifi |
| 11:35.05 | angryuser | DrCron: 200 MHz is ok, but quality in nor really good |
| 11:36.46 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 11:41.54 | *** join/#asterisk creativx (n=creadure@226.62-97-205.bkkb.no) |
| 11:44.09 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
| 11:48.09 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 11:48.29 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 11:52.17 | *** join/#asterisk creativx (n=creadure@226.62-97-205.bkkb.no) |
| 11:54.20 | *** join/#asterisk emiquelito (n=evandro@200-155-185-1.static.spo.ifx.net.br) |
| 11:57.54 | *** join/#asterisk emiquelito (n=evandro@200-155-185-1.static.spo.ifx.net.br) |
| 11:58.45 | *** join/#asterisk lorinc (n=ang@213.178.120.186) |
| 12:01.06 | *** join/#asterisk apardo (n=apardo@87.217.144.255) |
| 12:02.43 | *** join/#asterisk lorinc (n=ang@213.178.120.186) |
| 12:03.36 | vgster | does anyone have any knowledge on call recording using asterisk but having the calls answered on a nortel system? is it even possble? |
| 12:04.15 | *** join/#asterisk oQPa (n=roque@198.Red-83-41-117.dynamicIP.rima-tde.net) |
| 12:06.07 | backblue | vgster: mitm? |
| 12:06.13 | *** join/#asterisk zeeesh (i=aadilism@9-237-154-202.wol.net.pk) |
| 12:06.16 | zeeesh | hi |
| 12:06.27 | vgster | i think they said it was a BM whatever that is |
| 12:06.38 | backblue | man in the midle? |
| 12:06.45 | backblue | it is what you want to do? |
| 12:07.45 | vgster | put assterisk in the middle between the pri and nortel |
| 12:10.20 | backblue | yes it's that |
| 12:10.29 | backblue | i have never done that |
| 12:11.56 | vgster | ok |
| 12:12.01 | vgster | i shall go away and look harder |
| 12:12.12 | backblue | but i have read something about it |
| 12:12.59 | backblue | vgster: Monitor will not do that? |
| 12:13.04 | backblue | monitor() |
| 12:13.04 | backblue | ? |
| 12:13.29 | vgster | ok ill have a look thanks |
| 12:15.48 | backblue | MixMonitor will be better i think |
| 12:15.52 | backblue | less cpu |
| 12:16.42 | Ahrimanes | hm more cpu i think actually.. but one file at the end |
| 12:19.37 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 12:23.12 | backblue | no, Mixmonitor less cpu and only 1 file |
| 12:23.31 | *** join/#asterisk jm|work (n=jamie@dilbert.jamiem.com) |
| 12:23.40 | backblue | monitor can have only 1 file, but actually it only merges in the end, it records 2 streams |
| 12:24.33 | backblue | mixmonitor record 2 streams in only 1 file |
| 12:24.58 | vgster | yes doesnt mixmonitor mixing on the fly and not at the end |
| 12:27.04 | DrCron | so should having a 400 mhz pocket pc increase sound quality over a 200mhz? |
| 12:30.54 | backblue | TDMoE anyone? |
| 12:33.56 | *** join/#asterisk Skarmeth (n=Skarmeth@201009061013.user.veloxzone.com.br) |
| 12:35.05 | tzafrir | backblue, ask anyway |
| 12:35.54 | backblue | i need to pass 2 channels for a card, over ip, to another asterisk machine, what it's the best way to do it? |
| 12:38.51 | zoa | iax2 |
| 12:38.56 | zoa | dont use tdmoe for that |
| 12:38.58 | zoa | its overkill |
| 12:42.37 | backblue | tdmoIP |
| 12:42.51 | backblue | iax2? but iax2 will not pass directly the channel to the other asterisk |
| 12:43.42 | backblue | if the iax2 trunk it's down, i will not be able to treat the return msg's |
| 12:50.15 | *** join/#asterisk inspired (n=mikael@85.221.7.59) |
| 12:52.28 | `Sean | Hey, anyone know how i can route a specific call to certain phone, lets say via a callerid |
| 12:54.57 | DrCron | put it in your dialplan |
| 12:55.57 | DrCron | there are the standard operators and conditional statements in *; if, then ,else |
| 12:56.41 | *** join/#asterisk rbw_ (n=rowi@slave.rixport80.se) |
| 13:00.23 | jgoo | RE: soundpoint 501, can anyone check the power rating on the adapter? Just want to be 100% (12VDC 400mA) |
| 13:01.15 | jgoo | polycom 501 that is |
| 13:07.18 | kigh_ | Hey there .. how usable / stable is the AEL module? i'd love to write my generic dialplan with that language |
| 13:08.31 | tzafrir | kigh_, it has been rewritten in 1.4. That's all I know |
| 13:09.53 | kigh_ | ok then i wont use it now |
| 13:16.48 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 13:22.21 | *** join/#asterisk unik-rados (n=rados@c-68-62-71-76.hsd1.mi.comcast.net) |
| 13:22.42 | unik-rados | mornin' everyone |
| 13:23.08 | DrukenHME | uhg.... morning.... |
| 13:23.13 | unik-rados | did anyone work with asterisk and SMS? |
| 13:25.23 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 13:25.58 | DrukenHME | zoa: what is tdmoe? |
| 13:26.27 | zoa | TDM over ethernet |
| 13:27.10 | DrukenHME | well i kinda figured that, i'm not complettly stupid... but i've never heard of it to date... |
| 13:28.15 | DrukenHME | how is it accomplished? |
| 13:30.28 | zoa | http://www.spidermux.org/faq/#TDMOE |
| 13:31.16 | sasch | for this thing |
| 13:31.16 | sasch | [Jan 8 14:22:45] NOTICE[6257]: chan_sip.c:7085 sip_reg_timeout: -- Registration for '63759@sip.squillo.it' timed out, trying again (Attempt #419) |
| 13:31.22 | sasch | what is the problem ?? |
| 13:36.21 | DrukenHME | the registration is timing out... |
| 13:36.39 | DrukenHME | maybe their server is down ? |
| 13:40.26 | *** join/#asterisk markit (n=konversa@host119-245-static.72-81-b.business.telecomitalia.it) |
| 13:43.34 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 13:45.56 | DrukenHME | [TK]D-Fender: top'o'the morning to ya |
| 13:46.50 | [TK]D-Fender | DrukenHME: Aye |
| 13:47.56 | *** join/#asterisk Dibbler_Lap (n=Dibbler@212.248.196.179) |
| 13:48.52 | *** join/#asterisk HarryR (n=Harry@host-83-146-53-46.bulldogdsl.com) |
| 13:51.13 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-71-241-163-59.buff.east.verizon.net) |
| 13:53.08 | *** join/#asterisk foRza (n=forza@firewall.hikt.no) |
| 13:55.33 | *** join/#asterisk johann8384 (n=jonathan@69.64.10.198) |
| 14:05.48 | ThoMe | how i can use cdr with mysql? |
| 14:06.19 | pif | read the wiki? |
| 14:06.23 | *** join/#asterisk tsurko (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 14:06.51 | ThoMe | pif: u mean http://www.voip-info.org/wiki-Asterisk+cdr+mysql ? |
| 14:06.57 | markit | is it me, or there is no a sound (in the standard set) for "all lines are busy, try later" or something like that? |
| 14:08.00 | inittab | markit use tt-monkeys |
| 14:08.05 | inittab | it will be good for a laugh |
| 14:08.06 | inittab | =P |
| 14:08.31 | markit | inittab: ehm, I think my business will soffer a lot for it ;P |
| 14:09.35 | [TK]D-Fender | markit: FreePBX has one for that, go rip it off of them or just make it yourself. |
| 14:10.00 | markit | [TK]D-Fender: all-circuits-are-momentary-busy or something like that |
| 14:10.12 | markit | but I'm surprised is not present in the standard set, no? |
| 14:10.26 | markit | I'm the one that produces the italian version of the sounds |
| 14:11.07 | markit | so wondering if I have to produce some "additional" ones, but really would like to know why this very common needed sound is not in the "base" set |
| 14:11.24 | inittab | they needed room for tt-monkeys :( |
| 14:11.58 | ThoMe | pif: ? |
| 14:13.05 | jgoo | RE: polycom 501, can anyone check the power rating on the adapter? Just want to be 100% (12VDC 400mA) |
| 14:13.16 | inittab | jgoo, sec let me look at mine |
| 14:13.21 | jgoo | \o/ yey |
| 14:13.53 | inittab | 12v 400mah |
| 14:13.56 | jgoo | also anyone using one of these : http://www.peplink.com/products_balance_study.php |
| 14:14.06 | jgoo | inittab, sweet, that is what I have here, but I wanted to be sure! thanks!!!! |
| 14:14.11 | inittab | np |
| 14:14.13 | jgoo | inittab++ |
| 14:14.39 | Katty | morning. |
| 14:16.10 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:17.20 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 14:18.36 | *** join/#asterisk yogurt2ungue (n=yogurt2u@65-49-231-201.fibertel.com.ar) |
| 14:18.46 | jgoo | ok inittab: stupid Q#2, does it matter, on the POE cable where the power connects - which end goes into the phone, and which to the switch? |
| 14:18.56 | jgoo | there are no idiot proof markings, and I, admittedly, and an idiot |
| 14:19.09 | monsted | jgoo: it's a standard cat5 cable, so no |
| 14:19.29 | jgoo | monsted: it has a bit that you plug the power into, so i'd say it isn't standard cat5 |
| 14:19.48 | Makenshi | jgoo is it an inline injector? |
| 14:19.51 | monsted | jgoo: hmm? the cable looks different than standard cat5? |
| 14:19.51 | jgoo | yes |
| 14:19.53 | Makenshi | if so, yes it does matter |
| 14:20.05 | jgoo | phew. ok, so shortest end to the phone? |
| 14:20.23 | Makenshi | no idea, you'd best check with the manufacturer |
| 14:20.34 | monsted | the injector itself must point the right way, but the cables should be just like any other |
| 14:20.36 | Makenshi | i have seen different styles of inline injectors |
| 14:21.12 | monsted | the whole point is that you can supply power through your existing cables |
| 14:21.23 | jgoo | anyone know about polycom? |
| 14:21.35 | jgoo | the 501 - hrm. they don't like labelling things do they |
| 14:21.44 | *** join/#asterisk Galeras (n=xchat@Dynamic-IP-cr20011833103.cable.net.co) |
| 14:21.58 | jgoo | the poe cable, short end to the phone, or long end? |
| 14:22.14 | [TK]D-Fender | <PROTECTED> |
| 14:22.24 | jgoo | *looks* |
| 14:22.56 | jgoo | oh yes - I guess I got lucky and tried that end first, so i didn't realise |
| 14:23.05 | jgoo | [Tk]D-Fender+++++++ |
| 14:23.54 | *** join/#asterisk undrdawg (n=urgay@unaffiliated/underdawg) |
| 14:24.03 | undrdawg | hi |
| 14:24.30 | *** join/#asterisk bhrobinson (n=brobinso@northtx1-static.telwestonline.com) |
| 14:24.44 | undrdawg | is there a worthwhile modem to look for that can be easily turned into a fxs setup? |
| 14:24.56 | bhrobinson | can anyone tell me how to force a sip channel to use a specific trunk? |
| 14:25.14 | undrdawg | or |
| 14:25.28 | undrdawg | those sound/modem combos |
| 14:25.34 | undrdawg | to just use phone in/out |
| 14:25.48 | *** join/#asterisk jesselang (n=jesse@h69-130-164-40.69-130.unk.tds.net) |
| 14:25.48 | undrdawg | i really dont feel like paying shipping and waiting for it |
| 14:26.23 | *** join/#asterisk [tasty]freeze (n=simba@204-181-50-10.skybest.com) |
| 14:26.36 | [tasty]freeze | anyone play with bluetooth and asterisk around these parts? |
| 14:28.23 | Katty | [tasty]freeze: only a keyboard sorta way. |
| 14:28.36 | Katty | [tasty]freeze: and headsets. |
| 14:30.14 | DrukenHME | undrdawg: no, don't be a cheap ass and pay the damn shipping and oh you'll wait... yes you will wait... :) |
| 14:30.37 | `Sean | Hey Katty |
| 14:30.40 | `Sean | hows youre day |
| 14:30.43 | undrdawg | anyone want to sell one ? |
| 14:30.56 | undrdawg | or trade for digium x100p |
| 14:31.09 | `Sean | lmao |
| 14:31.15 | `Sean | dont buy digium simple buy sangoma |
| 14:31.27 | DrukenHME | don't use FXS, get voip phones.... |
| 14:31.34 | DrukenHME | or use adaptors |
| 14:31.43 | `Sean | cards are ok |
| 14:31.51 | `Sean | if there from someone that has good industry leading experience like sangoma |
| 14:31.57 | `Sean | they been doing this for over 20 yrs... |
| 14:31.57 | undrdawg | i want to answer calls for all my girlfriends in different area codes though :P |
| 14:32.08 | `Sean | and accoridng to some people wich i heatrd mark worked at sangoma as a interm |
| 14:32.09 | [TK]D-Fender | undrdawg: You can't |
| 14:32.15 | Katty | `Sean: allo...my day is.....still needing caffeine before it answers you. |
| 14:32.29 | [TK]D-Fender | Katty: Mew. |
| 14:32.32 | `Sean | haha Katty, No hugs for me ? |
| 14:32.34 | `Sean | :() |
| 14:32.36 | Katty | [TK]D-Fender: you'll be happy to know my brain is no longer in freak out mode. |
| 14:32.40 | undrdawg | couldnt i answer from more than 1 sip service? |
| 14:32.46 | bhrobinson | on the FXS, has anyone any experience with the xorcom? |
| 14:32.54 | undrdawg | really i just wanted a few phone lines |
| 14:32.58 | [TK]D-Fender | Katty: Excellent. Just some clarity, or some progress as well? |
| 14:33.04 | `Sean | haha |
| 14:33.15 | Katty | [TK]D-Fender: a little bit of both. |
| 14:33.26 | undrdawg | from one running asterisk if possible |
| 14:33.30 | [TK]D-Fender | Katty: Good, take your time on it all. |
| 14:35.05 | `Sean | hey fender, tell me one thing how do i tell asterisk in a dail plan to call the sip phone.... |
| 14:35.16 | `Sean | if the phone is connected via ethernet cable to my asterisk server on a seperate nic |
| 14:35.20 | Katty | [TK]D-Fender: yeah, i just needed to calm my tail down and put things in perspective. |
| 14:35.28 | `Sean | i know for zap its (Zap/1,30) |
| 14:35.34 | `Sean | but what would it be for a IP phone? |
| 14:36.03 | *** join/#asterisk lorinc (n=ang@caracas-1300.adsl.interware.hu) |
| 14:37.01 | DrukenHME | (SIP/username) |
| 14:39.30 | [TK]D-Fender | ~book |
| 14:39.32 | 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 |
| 14:39.46 | *** join/#asterisk oQPa (n=uawename@198.Red-83-41-117.dynamicIP.rima-tde.net) |
| 14:48.08 | unik-rados | can I receive SMS responses on Asterisk? |
| 14:48.50 | RoyK | unik-rados: depends how you want to do it, where you are, and what sort of equipment you use |
| 14:49.07 | RoyK | but yes, generally there are several ways to interface sms/asterisk |
| 14:49.41 | *** join/#asterisk karmatronic (n=boumkar@84.77.150.147) |
| 14:49.43 | unik-rados | RoyK: well, let's say I wanted to send an sms to a client of mine with an option for him to respond, if they want to be contacted right now, for example... |
| 14:49.56 | karmatronic | hi there |
| 14:50.43 | karmatronic | i get no sound when using bluetooth with asterisk ,although connections out and in are launched ..... |
| 14:51.59 | unik-rados | RoyK: can you point me to some resources where I can read up on it? |
| 14:53.05 | [TK]D-Fender | unik-rados: Look up SMS on the WIKI, and on the mailing lists |
| 14:53.08 | [TK]D-Fender | ~wikis |
| 14:53.10 | jbot | extra, extra, read all about it, wikis is http://www.voip-info.org |
| 14:53.10 | [TK]D-Fender | ~mailinglist |
| 14:53.11 | jbot | Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.htmm |
| 14:53.23 | RoyK | unik-rados: it depends what you have. do the sms come through a server to your box on the net? are you going to receive it on a cell phone connected to a serial port? SMS on PSTN works somewhere, but not everywhere |
| 14:53.24 | *** join/#asterisk bkw_ (n=brian@adsl-64-149-38-152.dsl.tul2ok.sbcglobal.net) |
| 14:53.38 | RoyK | bkw_: damn. haven't they banned you from here yet? |
| 14:53.48 | unik-rados | [TK]D-Fender: Thanks!! |
| 14:54.40 | *** join/#asterisk djflux (n=djflux@mm.shermfin.com) |
| 14:54.48 | Gido-E | the noodle inventor died :-( (quick instant noodle) |
| 14:54.56 | unik-rados | RoyK: well, I have lots of flexibility as to how I can set this up. I'm just looking at different options/ways that I can do it. |
| 14:55.29 | [TK]D-Fender | Ramen King no more! |
| 14:55.59 | *** join/#asterisk basty (n=basty@212.218.65.254) |
| 14:56.01 | *** join/#asterisk Lokiji (n=Lokiji@ip-89-102-178-195.karneval.cz) |
| 14:56.02 | basty | Hi |
| 14:56.07 | RoyK | unik-rados: we just use an sms provider. that works well until the internet link goes down, but for this usage, i'd just use AGI and script it up |
| 14:56.23 | *** join/#asterisk Druken (n=jdumais@CPE000854ddcdb1-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 14:56.36 | Lokiji | hi |
| 14:57.12 | unik-rados | RoyK: so you get an sms provider and that will allow me to send and receive sms messages via Asterisk, right? |
| 14:57.14 | basty | anyone knows how to setup the dialplan in asterisk to display the called number? I mean when I try to dial an internal number it should convert the number to the name. Is that possible? |
| 14:58.09 | tzafrir | unik-rados, from which country are you? |
| 14:58.21 | unik-rados | tzafrir: US |
| 14:58.30 | SuPrSluG | basty:set or get? |
| 14:58.32 | RoyK | unik-rados: no, I just use a simple perl script, calling it from asterisk with AGI or System to do an HTTP POST to another site |
| 14:58.44 | RoyK | and then a daemon to receive sms |
| 14:58.45 | tzafrir | basty, something like ${EXTEN} ? |
| 14:58.46 | Lokiji | hello could anyone tell me how to configure a jitter buffer on asterisk 1.4 on sip channel |
| 14:58.59 | RoyK | Lokiji: jb-enable=yes |
| 14:59.10 | Lokiji | royk: is that all? |
| 14:59.20 | RoyK | Lokiji: yes, but it's global, not per peer/channel |
| 14:59.36 | unik-rados | RoyK: ok, I'll try to read up on it and hopefully I won't have to ask any more about this... thanks for your help! |
| 15:00.25 | Lokiji | royk, so if i want to enable a jitter buffer on all sip channel i would write "jb-enable=yes" to global section in sip.conf? |
| 15:00.50 | RoyK | jb-enable will enable jitterbuffer for sip/zap bridges |
| 15:01.00 | RoyK | the only place you want it, really |
| 15:01.37 | RoyK | except you might want it for misdn or capi or whatever, but afaik asterisk doesn't support that. the 'generic' jitterbuffer is only generic by design, not by implementation |
| 15:01.43 | basty | suprslug/tzafrir: I want to "get" the name. Means the first and lastname should be displayed on the called party (instead of the number that I have called). |
| 15:02.02 | basty | s/called/calling |
| 15:02.07 | tzafrir | ${CALLERID(name)} ? |
| 15:02.17 | RoyK | Lokiji: remember that the jb is on the bridge, so stuff like meetme won't get dejittered (although meetme has it's own little jb) |
| 15:02.46 | tzafrir | http://voip-info.org/wiki/view/Asterisk+func+callerid |
| 15:02.47 | basty | tzafrir: that would be a way to set the callerid on the "other side" |
| 15:02.50 | tzafrir | If I got the URL right |
| 15:02.52 | SuPrSluG | or ${CALLERID(all)} for both, but some providers don't pass name |
| 15:03.06 | basty | suprslug: its only for internal |
| 15:03.06 | undrdawg | im not exactly sure what the point of ipkall is |
| 15:03.07 | zoa | it takes only a few hours to add it to misdn or capi i think |
| 15:03.18 | tzafrir | You can set whatever it is you send. However your provider may override it |
| 15:03.25 | undrdawg | it wants me to get SIP service so i can get a free phone number |
| 15:03.28 | Lokiji | royk: could jb help when i have one sip phone conected by wifi? |
| 15:03.36 | undrdawg | i thought they sorta went together |
| 15:03.44 | RoyK | zoa: sure, but that's features, so we'll have to wait till 1.6 :D |
| 15:03.49 | zoa | hehe |
| 15:03.52 | RoyK | Lokiji: indeed |
| 15:04.01 | zoa | nobody seems to care about the jb now that its in there |
| 15:04.06 | zoa | i rarely see questions about it |
| 15:04.25 | zoa | looks like it only had to be on the feature list in the technical details :) |
| 15:04.29 | RoyK | zoa: we have reversed customer loss after installing it. thanks, mate |
| 15:04.29 | Juggie | zoa! |
| 15:04.50 | zoa | cool, thats good news |
| 15:04.51 | basty | tzafrir: Mmmm - well I dont want to "set" it. Lets say, right now I am dialing "18" from my phone. During the call my display keeps showing "18". I dont wanna see "18" - I want to get the Name on MY PHONE from "18" like Paul Harbourg or something..you know ? :-) |
| 15:05.02 | zoa | we actually stopped the project we needed it for |
| 15:05.12 | zoa | so we developped something for a year without using it in the end |
| 15:05.28 | JerJer | zoa: sounds familiar |
| 15:05.31 | zoa | hey ho juggie |
| 15:05.31 | RoyK | zoa: the code we paid you to write? |
| 15:05.41 | zoa | well you paid only a tiny part of it :) |
| 15:05.45 | RoyK | i know |
| 15:05.46 | zoa | but thanks anyway! |
| 15:05.47 | zoa | yeah |
| 15:05.48 | [TK]D-Fender | basty: What you're looking for isn't a standard feature though there is a patch out there for support for it with Polycom phones. |
| 15:05.55 | zoa | we dont use it now |
| 15:06.02 | Lokiji | royk:one last question:-)do u know how to set a size of jb?:-) |
| 15:06.11 | zoa | we would if we would do termination |
| 15:06.12 | RoyK | Lokiji: sip.conf |
| 15:06.15 | zoa | but not now |
| 15:06.18 | *** part/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 15:06.38 | RoyK | Lokiji: but don't use the adptive one. use the static. it adds a little latency, but it's stable, and it works. adaptive sucks |
| 15:06.41 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 15:06.47 | [TK]D-Fender | oops |
| 15:07.03 | JerJer | for whatever reasons I have never had to bother with Jb |
| 15:07.06 | basty | TKD-Fender: Any workarounds for doing that on Snom phones ? |
| 15:07.17 | *** join/#asterisk in-pt (n=lokesh@estrela.nortenet.pt) |
| 15:07.32 | Lokiji | royk:-)could u tell me where can i find more info about ib?:-) |
| 15:07.47 | [TK]D-Fender | basty: Check your phone's manual, it'd have top be a special service offered by them |
| 15:07.55 | RoyK | Lokiji: just see the sample sip.conf |
| 15:08.02 | Aurs | will copying voicemail inboxes (when using realtime voicemail) have any negative impact on asterisk? like this: cp -R user1 user2 (pwd = /var/spool/asterisk/voicemail/default/) |
| 15:08.05 | [TK]D-Fender | basty: Some phones can be told to look in the local directory, or on a server. |
| 15:08.18 | [TK]D-Fender | basty: But I wouldn't bet on it if I were you. |
| 15:08.19 | Lokiji | royk:ok, thanks a lot:-) |
| 15:09.03 | basty | [TK]D-Fender: okay - snom supports it with the local directory...but my boss doesnt want to have local numbers in the global address directory... |
| 15:09.15 | [TK]D-Fender | basty: TFB |
| 15:09.33 | basty | TFB ? |
| 15:09.38 | [TK]D-Fender | ~tfb |
| 15:09.53 | basty | heheh |
| 15:09.58 | [TK]D-Fender | Too #%@ing bad :) |
| 15:10.04 | basty | oh okay :-) |
| 15:10.29 | basty | Mmm...maybe I could realise it with SendText() ? |
| 15:10.52 | basty | I mean there must be a way to do that...its actually a normal feature for PBX :-\ |
| 15:11.03 | [TK]D-Fender | basty: Highly doubt it. |
| 15:11.32 | [TK]D-Fender | basty: WAke up and smell the toast burning. You aren't going to get everything you get on key systems. |
| 15:11.59 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 15:13.25 | tzafrir | Hmmm... just got a spam from atcom |
| 15:13.32 | `Sean | atcom? |
| 15:14.03 | *** join/#asterisk adker_ (n=chatzill@74-33-198-79.br1.glv.ny.frontiernet.net) |
| 15:15.43 | SuPrSluG | does sip jitterbuffer work w/ 1.2.13? |
| 15:16.20 | zoa | no |
| 15:16.25 | *** part/#asterisk karmatronic (n=boumkar@84.77.150.147) |
| 15:16.27 | basty | [TK]D-Fender: hehe |
| 15:16.34 | SuPrSluG | only 1.4 |
| 15:16.54 | basty | such a pitty :-\ so well...thanks for your support, [TK]D-Fender. |
| 15:17.17 | [TK]D-Fender | basty: Try #drphil |
| 15:17.20 | *** join/#asterisk greekman (n=alex@63.118.244.59) |
| 15:17.27 | basty | pft! |
| 15:17.28 | [TK]D-Fender | NEXT!!@!@ (c) BKW |
| 15:17.28 | greekman | hi all |
| 15:17.36 | basty | heheh bye for now ! ;-) |
| 15:17.57 | greekman | does anyone have any experience with removing * from the media path? |
| 15:18.22 | undrdawg | yea |
| 15:18.30 | undrdawg | try this |
| 15:18.45 | greekman | i'm using an svn head from last may |
| 15:18.46 | undrdawg | perl -le "unlink '\*'" |
| 15:19.12 | greekman | what? |
| 15:19.27 | greekman | undrdawg u talkin to me? |
| 15:19.27 | [TK]D-Fender | greekman: "canreinvite=yes" for all applicable entries. |
| 15:19.39 | [TK]D-Fender | greekman: Ignore him, he's being an ass. |
| 15:20.36 | undrdawg | ya but wow that doesnt work |
| 15:20.47 | undrdawg | i thought you were talking about a file called * |
| 15:21.21 | undrdawg | perl -le "unlink '*'" |
| 15:21.22 | undrdawg | i mean |
| 15:21.35 | [TK]D-Fender | undrdawg: * = ASTERISK |
| 15:22.10 | undrdawg | oh hehe |
| 15:22.12 | undrdawg | sorry |
| 15:24.04 | b11d | morning lads |
| 15:24.07 | suma | how can i configure jingle with the asterisk 1.4 ? |
| 15:24.21 | suma | could not see file jingle.conf |
| 15:24.40 | b11d | has it been depreciated in favor of an alternative? |
| 15:25.15 | suma | b11d: how can i talk to googletalk using asterisk ? |
| 15:25.39 | [TK]D-Fender | suma : Maybe you have to make it yourself. Did you TRY? |
| 15:26.07 | b11d | haha |
| 15:26.09 | suma | i checked in voip-info.org it has instructions for the previous versions |
| 15:26.25 | sasch | for italian jingles for asterisk |
| 15:26.34 | sasch | exist a site that sell it or no ?? |
| 15:26.38 | *** join/#asterisk anthm (n=anthmct@CPE-72-131-113-50.wi.res.rr.com) |
| 15:26.38 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:26.43 | greekman | i am having problems getting asterisk out of media path with canreinvite=yes already set... can anyone help? |
| 15:27.05 | Ahrimanes | greekman, do you use the t or T option in Dial() ? |
| 15:27.11 | suma | i found it |
| 15:27.15 | suma | gtalk.conf |
| 15:27.20 | b11d | :) |
| 15:28.02 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 15:28.11 | b11d | ugh |
| 15:28.16 | b11d | why didnt I think this all the way through.. |
| 15:28.25 | b11d | damn my initiate mind |
| 15:28.47 | *** join/#asterisk Blok (n=sa@c-b323e353.1257-1-64736c12.cust.bredbandsbolaget.se) |
| 15:29.38 | Blok | Hi, I looking for a good tutorial for the extension file |
| 15:29.44 | b11d | voip-info.org |
| 15:29.55 | b11d | extensionc.conf.sample too |
| 15:29.56 | b11d | :) |
| 15:29.58 | b11d | doh |
| 15:30.01 | b11d | extensions.conf.sample |
| 15:30.39 | SuPrSluG | if I want the caller to be able to get back to the menu from whence they came, is that explained in the options for voicemail app? |
| 15:30.48 | b11d | yeah |
| 15:30.59 | b11d | it usually says "pres # for main menu" |
| 15:31.01 | b11d | or something like that |
| 15:32.16 | *** join/#asterisk infernix (i=nix@spirit.infernix.net) |
| 15:32.30 | Blok | I got a pretty nice file right now, but I want to go further. What are the acl files? |
| 15:33.38 | [TK]D-Fender | Blok: acl files? |
| 15:34.05 | Blok | I want to be able to use a more C-like syntax if possible with loops and selection syntax |
| 15:35.14 | Ahrimanes | Blok, look at AEL2 ? |
| 15:35.36 | Blok | yeah, I just found it, not acl, but AEL1/2 |
| 15:35.37 | [TK]D-Fender | Blok: That would be "AEL", not ac. It also is typically used only by luke-jr here.... so support is spares. |
| 15:35.53 | [TK]D-Fender | Blok: Nobody cares much about it, and with good reason. |
| 15:36.31 | Blok | checking it right now, looks like thing I as looking for |
| 15:37.08 | [TK]D-Fender | sparse* |
| 15:37.26 | Blok | dude, it looks so sweet ;) |
| 15:38.34 | pif | my extensions.conf is so nice I somtimes jerk off to it |
| 15:38.35 | b11d | yeah it does |
| 15:38.36 | b11d | AEL2 is nice |
| 15:38.43 | [TK]D-Fender | Blok: How complex is your dialplan to need that? |
| 15:38.50 | Blok | Why does nobody like it? |
| 15:38.52 | [TK]D-Fender | b11d: Will it work with RealTime? :) |
| 15:39.02 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 15:39.18 | Blok | pif: Then I guess AEL would be like heaven or better ;) |
| 15:39.19 | *** join/#asterisk flot (n=flot@ppp85-141-82-125.pppoe.mtu-net.ru) |
| 15:39.23 | *** part/#asterisk flot (n=flot@ppp85-141-82-125.pppoe.mtu-net.ru) |
| 15:39.32 | [TK]D-Fender | Blok: Because it allows you to write even MORE fucked up code thats next to impossible to debug and with nobody else using it your odds decreae exponentially. |
| 15:39.37 | pif | AEL would be like a bj :) |
| 15:39.55 | [TK]D-Fender | pif: You mean more like a "Lewinski" ;) |
| 15:40.00 | Blok | hehe |
| 15:40.03 | pif | yeah |
| 15:40.14 | [TK]D-Fender | pif: Blow you now.... fuck you later :D |
| 15:40.20 | b11d | I dont know.. I dont use realtime. |
| 15:40.49 | pif | "hillary put the free world in jeapardy by not sucking bill" |
| 15:40.57 | b11d | watch some trek |
| 15:42.02 | Blok | [TK]D-Fender: The thing is that in my eyes the "usual" extension.conf syntax looks like a huge inconsistent mess. Though my eye is untrained. :) |
| 15:42.14 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 15:42.34 | [TK]D-Fender | b11d: I think most offices would protest their employees watching TV on the clock.... unless its the Neilsons's ;) |
| 15:43.04 | [TK]D-Fender | Blok: Well jump on the AEL bandwagon if you want, but the wheels are loose and there is no insurance :) |
| 15:43.12 | b11d | "most" |
| 15:43.25 | pif | Katty : ChanSpy() you users when you're bored |
| 15:43.46 | Katty | heh |
| 15:43.47 | pif | lotsa fun how inane the convos are |
| 15:43.53 | b11d | thats a lie |
| 15:44.02 | b11d | i remember thinking how "cool" it would be to listen via a scanner back in the day |
| 15:44.07 | b11d | people talk about such boring shit.. |
| 15:44.11 | pif | did you buy such and such vegetable for tonight honey? |
| 15:44.18 | b11d | lol |
| 15:44.51 | pif | I told you that bitch was crazy, etc.. etc.. |
| 15:45.22 | Ahrimanes | much more fun with business users |
| 15:45.29 | Blok | And when bandwagon takes off to hell, only then, will I find true happiness. :) |
| 15:45.46 | Katty | oh! |
| 15:45.50 | Katty | i know something i could do! |
| 15:46.08 | Katty | and it has everything to do with vegetables ^_^ |
| 15:46.08 | pif | ? |
| 15:46.15 | pif | cucumber? |
| 15:46.23 | Katty | don't much care for those. |
| 15:46.29 | b11d | zucchini |
| 15:46.30 | Katty | cept on baby rye. |
| 15:46.30 | b11d | ? |
| 15:46.36 | Katty | hmm, no |
| 15:46.41 | b11d | sqash? |
| 15:46.43 | b11d | egplant? |
| 15:46.45 | b11d | doh |
| 15:46.47 | Katty | never had it. |
| 15:46.56 | Katty | favorite all time recipes. |
| 15:47.11 | Katty | b11d: what's your favorite thing? |
| 15:47.19 | b11d | in what category? or all around? |
| 15:47.38 | *** join/#asterisk bhrobinson (n=brobinso@northtx1-static.telwestonline.com) |
| 15:47.45 | Katty | Ahrimanes: i can make mean garlic mashed potatos ;) |
| 15:47.50 | *** join/#asterisk karmatronic (n=boumkar@84.77.150.147) |
| 15:48.02 | Ahrimanes | Katty, ah even better.. just make my dinner then ;) |
| 15:48.10 | [TK]D-Fender | :F |
| 15:48.16 | Ahrimanes | big words.. |
| 15:48.17 | Katty | Ahrimanes: i only make dinner for those who appreciate it. |
| 15:48.32 | Katty | [TK]D-Fender: what's your favorite thing? |
| 15:48.32 | Ahrimanes | Katty, believe me, i would appreciate it :) |
| 15:48.33 | pif | and what happens after diner? |
| 15:48.41 | [TK]D-Fender | Katty: This is a family channel! |
| 15:48.43 | [TK]D-Fender | :O |
| 15:48.48 | Katty | [TK]D-Fender: come now, you perv. |
| 15:49.04 | Ahrimanes | haha |
| 15:49.09 | pif | all this vegetable talk is arousing |
| 15:49.19 | [TK]D-Fender | Katty: lol.... Depends I guess. I have a lot of favourite things. In terms of hobbies, its music & pool. |
| 15:49.20 | Ahrimanes | [TK]D-Fender, hm i'm no expert, but i do believe i need my blood |
| 15:49.22 | Katty | last thing i made was chimichanga. |
| 15:49.31 | Katty | [TK]D-Fender: we're talking dinner, you silly rabbit. |
| 15:49.41 | Ahrimanes | Katty, no fair.. mexican is my fav.. |
| 15:49.54 | [TK]D-Fender | Ahrimanes: Torpor! All the good Anti-Deluvians are doing it! Catch the wave and await Gehenna! |
| 15:50.06 | Katty | Ahrimanes: well the chimichanga is all gone. hence me needing to make something tonight. |
| 15:50.30 | santibiotico | hi |
| 15:50.35 | Ahrimanes | [TK]D-Fender, play a lot of vampire, do we? |
| 15:50.41 | [TK]D-Fender | Katty: Food? Oh... Can't pick a single favourite. I love everything from middle-east to aisin most. |
| 15:50.42 | b11d | I love REAL homemade pasta.. in rigatoni form. |
| 15:50.56 | [TK]D-Fender | Ahrimanes: We spoke on this earlier, just thought you'd pick up faster :) |
| 15:51.03 | Ahrimanes | Katty, hehe.. ah.. actually i have others cooking dinner for me tonight now i think about it |
| 15:51.19 | [TK]D-Fender | Ahrimanes: And not for over 5 years, but I have a pornographic memory.... |
| 15:51.24 | Ahrimanes | [TK]D-Fender, hehe i'm never fast to pick up on things |
| 15:51.31 | Ahrimanes | [TK]D-Fender, hehe i havent played for about 10 years.. |
| 15:51.42 | santibiotico | i'm having a problem...whenever i receive a call i make asterisk to answer the call and then to transfer it to an extension...well...on that extension the caller id i see is "asterisk"...is there any way to show the number which has called me?? |
| 15:51.58 | *** part/#asterisk karmatronic (n=boumkar@84.77.150.147) |
| 15:52.12 | Ahrimanes | santibiotico, why do you answer() the call before the transfer? |
| 15:52.19 | [TK]D-Fender | santibiotico: That happens usually on shitty X100P cards where CALLERID fialed. In this case there is nothing to see. |
| 15:52.25 | [TK]D-Fender | santibiotico: Or you ust didn' |
| 15:52.29 | [TK]D-Fender | t set it up at all. |
| 15:52.31 | *** join/#asterisk infernix (i=nix@spirit.infernix.net) |
| 15:52.43 | Katty | [TK]D-Fender: i've not made any asian stuff yet. |
| 15:52.46 | santibiotico | i'm not using any x100p card |
| 15:52.55 | Katty | [TK]D-Fender: i shall require an asian recommendation. |
| 15:53.00 | santibiotico | i'm using a junghanns quadbri card |
| 15:53.01 | [TK]D-Fender | Katty: I make sushi regularly, and am due for another batch this week. |
| 15:53.09 | pif | Katty : still in romania? |
| 15:53.21 | Ahrimanes | mm sushi |
| 15:53.31 | *** join/#asterisk wunderkin (i=kev@ip72-208-3-221.ph.ph.cox.net) |
| 15:53.32 | santibiotico | but i know it's not a matter of the card, as i'm using another one in another server where i have no problems |
| 15:53.35 | Katty | i'm actually not big on sushi |
| 15:54.24 | *** join/#asterisk infernix (i=nix@spirit.infernix.net) |
| 15:54.28 | santibiotico | Ahrimanes: i'm not using answer() |
| 15:54.35 | santibiotico | Ahrimanes: i use a dial statement |
| 15:54.38 | [TK]D-Fender | pif: Careful, bkw_ is lurking.... |
| 15:54.38 | DonX | Maybe someone can help me with this. I'm trying to only use the last 4 digits of an extension that I receive. I know ${EXTEN:*} drops however many digits are in * from the beginning but I can do this as the number I get is different each time |
| 15:54.57 | DonX | How can I tell a dial sequence to only dial the last four digits of an extension? |
| 15:55.04 | pif | is he gay too? |
| 15:55.05 | robl^ | pif: careful! you never know how people will intepret that comment around here. you might get an offer for the wrong type of sausage |
| 15:55.23 | Katty | pif: sausage? |
| 15:55.30 | Katty | pif: just...sausage? |
| 15:55.37 | [TK]D-Fender | Katty: Considering your vegen tendencies, you should start.... |
| 15:55.37 | Katty | you guys are boring. |
| 15:55.44 | Katty | [TK]D-Fender: i'm not vegan anymore. |
| 15:55.48 | pif | with mustard |
| 15:55.54 | Katty | pif: italian sausage? |
| 15:56.41 | pif | don't trust italian sausages |
| 15:56.51 | [TK]D-Fender | Katty: Then let me suggest the filet mignon, blue & seared :D My fav.... make sure it MOO's when you stab it. |
| 15:56.53 | Ahrimanes | DonX, hm not sure, but maybe you can do ${EXTEN:-4} |
| 15:57.02 | santibiotico | is there a way to rewrite the caller id? |
| 15:57.13 | DonX | I'll try that |
| 15:57.26 | [TK]D-Fender | santibiotico: Pastebin your zapata.conf |
| 15:57.26 | [TK]D-Fender | ~pb |
| 15:57.27 | jbot | well, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 15:57.33 | file | Katty: ! |
| 15:57.38 | [TK]D-Fender | santibiotico: and for practicalitity'ssake : NO |
| 15:57.40 | *** join/#asterisk gr0mit_home (n=Tim@extrt.txrx.org.uk) |
| 15:57.48 | [TK]D-Fender | file: ! ! ! |
| 15:57.55 | zoa | katty!!! |
| 15:57.59 | zoa | file!!!! |
| 15:58.04 | zoa | [TK]D-Fender!!!! |
| 15:58.06 | file | !?! |
| 15:58.10 | zoa | pif!!!!! |
| 15:58.12 | [TK]D-Fender | zoa: MY inside joke, get your own |
| 15:58.20 | pif | zao! |
| 15:58.23 | pif | phile! |
| 15:58.23 | *** join/#asterisk brain- (n=ruy@116-144-142-200.mcmtelecom.com.br) |
| 15:58.34 | file | [TK]D-Fender: I just want |
| 15:58.54 | file | [TK]D-Fender: it's snowing here :( |
| 15:58.58 | *** join/#asterisk PupenoR (n=pupeno@2002:c87b:b75a:0:240:f4ff:fe6b:7650) |
| 15:59.57 | *** part/#asterisk brain- (n=ruy@116-144-142-200.mcmtelecom.com.br) |
| 15:59.58 | Katty | file: have you had anything yummy lately? i need a recipe so i can go get groceries. |
| 16:00.00 | [TK]D-Fender | file : 2cm here, and rain coming to erase it at 6oC today :) |
| 16:00.12 | file | Katty: no'm |
| 16:00.18 | Katty | :< |
| 16:00.40 | *** join/#asterisk nesys (n=nesys@81-174-12-111.f5.ngi.it) |
| 16:01.18 | *** join/#asterisk brain- (n=ruy@116-144-142-200.mcmtelecom.com.br) |
| 16:01.26 | pif | a chopper crashed at take-off yesterday in south of france (passengers and pilot fine) 3 onlookers dead with severed head, one women with servered leg |
| 16:02.28 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.149) |
| 16:02.32 | Ahrimanes | DonX, success? |
| 16:02.44 | santibiotico | [TK]D-Fender: http://pastebin.ca/310244 |
| 16:02.53 | santibiotico | [TK]D-Fender: that's my zapata.conf |
| 16:02.58 | nesys | hello folks ... someone using asterisk, or trixbox, with ccm through a sip trunk? |
| 16:05.49 | Katty | what goes with fried chicken? |
| 16:06.08 | Ahrimanes | more fried chicken or maybe rice and curry sauce |
| 16:06.17 | DonX | Ahrimanes: Yes, it worked. Thanks :) |
| 16:06.29 | Ahrimanes | DonX, no problem :) |
| 16:06.54 | Katty | oooh. |
| 16:06.55 | DonX | Ya know, my wife sent me to the store the other day with a list and put 'file' on there |
| 16:06.57 | Katty | cranberry rice! |
| 16:07.02 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 16:07.03 | DonX | I had no clue what she was talking about |
| 16:07.06 | Ahrimanes | haha |
| 16:07.14 | DonX | Finally I went and asked someone what file was |
| 16:07.38 | DonX | and after about 5 minutes of embarrassing confusion, I learned there was a spice called file` |
| 16:07.46 | nesys | I'm trying to connect asterisk with ccm 4.1(3), the sip trunk goes up, but when I try to call, the phone rings, but no voice ... maybe an rtp problem, but where? could you help me my troubleshooting? |
| 16:08.27 | Juggie | 'rtp debug' |
| 16:08.36 | DonX | nesys: what codec are you using? |
| 16:09.06 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 16:09.09 | [TK]D-Fender | asjkldhlkajshdlkashjd;lsfkljgdf |
| 16:09.22 | nesys | g711u |
| 16:09.54 | nesys | the rtp debug says something like that: |
| 16:09.54 | *** join/#asterisk apardo (n=apardo@87.217.144.255) |
| 16:09.55 | nesys | Got RTP packet from 192.168.17.21:8000 (type 0, seq 660, ts 105600, len 160) |
| 16:10.11 | DonX | sounds like an RTP problem maybe |
| 16:10.32 | nesys | 17.21 is an x-lite registered on my asterisk |
| 16:10.47 | nesys | I try to call from a 7960 registered on cisco callmanager |
| 16:11.41 | nesys | would you see my sip debug ip 192.168.17.21 ? |
| 16:12.23 | *** join/#asterisk s1gny|wrk (n=s1gny@p5491623E.dip.t-dialin.net) |
| 16:13.21 | undrdawg | is there a free service that lets you call POTS lines? |
| 16:13.32 | undrdawg | i have a broadvoice account i just have to call them to get my pw |
| 16:13.36 | *** part/#asterisk s1gny|wrk (n=s1gny@p5491623E.dip.t-dialin.net) |
| 16:13.38 | undrdawg | and no phone |
| 16:13.41 | *** join/#asterisk hoobastooba (n=ckwall@63.149.122.93) |
| 16:14.00 | *** join/#asterisk infernix (i=nix@spirit.infernix.net) |
| 16:14.26 | nesys | DonX http://pastebin.ca/310255 ... 192.168.17.21 is my x-lite on *, 192.168.17.20 is *, and 192.168.50.28 is ccm |
| 16:15.32 | nesys | the problem is maybe that: |
| 16:15.34 | nesys | om: "601" <sip:601@192.168.17.20>; |
| 16:16.08 | nesys | 601 is the ccm registered phone ... but 17.20 is *, and not ccm ... why? |
| 16:17.08 | nesys | asterisk says to x-lite "talk with me, 601 is mine" ... that's normal? |
| 16:18.08 | *** join/#asterisk IPmonger (n=ipmonger@c-68-84-208-206.hsd1.pa.comcast.net) |
| 16:20.54 | nesys | have you got an advice for me? |
| 16:21.16 | hoobastooba | anyone here well versed in phpagi? |
| 16:21.53 | *** join/#asterisk Mad|Cow (n=thirt@74.92.109.205) |
| 16:24.13 | *** join/#asterisk mercestes (n=merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 16:25.17 | Mad|Cow | I am thinking about allowing external users to use a soft phone for when they work remote. It seems kind of insecure. The softphones use a rawtext password to connect to the PBX which means a potential hacker could sniffer the username & password and connect to my PBX and make phone calls. Am I missing something? |
| 16:26.03 | tset | use a vpn? |
| 16:26.03 | mercestes | Mad|Cow: I potential hacker could man in the middle attack you and intercept a Sip authentication and authenticate themselves in your Softphone's stead, with or without it's password. |
| 16:26.36 | Mad|Cow | I dont want to use a VPN b/c of the overhead... |
| 16:26.36 | *** join/#asterisk jmls (n=asterisk@62.49.235.130) |
| 16:26.46 | jmls | hey, is this a good thing or bad thing ? |
| 16:26.47 | jmls | NOTICE[1184]: sched.c:283 ast_sched_del: Attempted to delete nonexistent schedule entry 24260 |
| 16:26.50 | jmls | (lots of em) |
| 16:26.51 | mercestes | Mad|Cow: There is also the host of buffer overflows you may be vunerable to. |
| 16:26.56 | DonX | nesys: Can you setup an extension that just does a playback on welcome or something? |
| 16:27.06 | DonX | and then dial from a phone behind your cm |
| 16:27.12 | DonX | isolate xlite |
| 16:27.13 | tset | Mad|Cow its a choice you need to decide on :) |
| 16:27.28 | Mad|Cow | If I dont want to use a VPN, how secure is this if I allow the use of a soft phone? |
| 16:27.28 | mercestes | Mad|Cow: Have you considered permit/deny statements? |
| 16:27.55 | mercestes | Mad|Cow: About as secure as windows |
| 16:27.55 | Mad|Cow | mercestes: How so, by IP address? Sounds like a admin nightmare |
| 16:27.58 | tset | Mad|Cow do a test case and analyze your network packets |
| 16:28.13 | Mad|Cow | lol |
| 16:28.28 | mercestes | Mad|Cow: anyone who knows what they are doing who is comitted to stealing your phone service is likely going to be able to |
| 16:28.30 | Katty | [TK]D-Fender: i think i decided on fried chicken. |
| 16:28.41 | hoobastooba | I have some scripts that I think are causing a memory leak. Is there any chance that someone here has enough experience with this that they could look it over and help me out? |
| 16:28.53 | Mad|Cow | mercestes: Good point. |
| 16:29.22 | nesys | DonX when I try to call from 601 to x-lite, I check with the moh on my ccm |
| 16:30.15 | tset | Mad|Cow at my work we use VPN's to allow for a little extra security, we only use SIP communication for remote employee's, and they are required to VPN (the phone server is non routable otherwise) |
| 16:32.08 | *** join/#asterisk s1gny|wrk (n=s1gny@p5491623E.dip.t-dialin.net) |
| 16:33.18 | mercestes | Mad|Cow: Now hte question is...how likely is someone gonig to be dedicated on stealing your service? |
| 16:33.26 | *** part/#asterisk s1gny|wrk (n=s1gny@p5491623E.dip.t-dialin.net) |
| 16:33.57 | De_Mon | mercestes you cant MTM unless you work at the hotel they are working remotely from |
| 16:34.11 | De_Mon | MITM |
| 16:34.11 | nesys | DonX there's an extension for test purpose, is it? |
| 16:35.39 | nesys | DonX I'm trying with an extension with playback(demo-echotest) |
| 16:35.43 | *** join/#asterisk codefreeze (n=steve_mu@216.166.159.235) |
| 16:36.14 | mercestes | De_Mon: Yea yo ucan. |
| 16:36.40 | mercestes | De_Mon: But the Sip Hack doesn't even require a MITM so it's a moot point. |
| 16:36.45 | *** join/#asterisk jmls (n=asterisk@62.49.235.130) |
| 16:43.00 | De_Mon | how do you mitm someone without being at a location the traffic travels through? |
| 16:44.09 | Rhizome | Call them and trick them into using your proxy? |
| 16:44.17 | Rhizome | :P |
| 16:44.52 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 16:44.52 | *** mode/#asterisk [+o mog] by ChanServ |
| 16:44.56 | *** part/#asterisk jmls (n=asterisk@62.49.235.130) |
| 16:45.17 | *** part/#asterisk hoobastooba (n=ckwall@63.149.122.93) |
| 16:45.24 | greekman | can anyone help with SDP? |
| 16:45.55 | greekman | i'm having issues with re-invite |
| 16:47.09 | *** join/#asterisk buzzydex (n=psp-man@194.164.236.34) |
| 16:47.27 | buzzydex | hi guys |
| 16:48.12 | buzzydex | Is there anyway to increase the volume of a call or cancel echo when calling Sip to Sip ?? |
| 16:48.57 | buzzydex | I see lots to do with zap interfaces but nothing on sip to sip either that I'm not looking properly |
| 16:48.57 | backblue | you only have echo from sip to sip with stupid phones |
| 16:49.09 | buzzydex | snom 190 to Spa 941 it keeps happening |
| 16:49.29 | backblue | you teorically dont have echo problems from sip to sip, unless you speak with someone that it's right in front of you, and you speak lought. :) |
| 16:50.12 | buzzydex | thats what I thought seeing as they both digital |
| 16:51.07 | buzzydex | the SPA941 keeps having issues with low TX when dialing out over the PSTN via e1 Cisco gateway something my snoms have no issues with |
| 16:51.25 | backblue | buzzydex: http://www.spiritdsp.com/pdf/article_4.pdf |
| 16:51.25 | buzzydex | its all very odd |
| 16:51.28 | zoa | backblue: well unless you have speaker to headset things |
| 16:52.10 | backblue | zoa: that's what i have said, stupid phones. |
| 16:52.30 | buzzydex | speak to headset things? |
| 16:52.39 | Katty | mercestes: think you might have time to help me with cdr/mysql stuff this afternoon? |
| 16:52.56 | buzzydex | not sure what you getting at on that but they using the handsets |
| 16:53.09 | buzzydex | cheers for link will have a read |
| 16:53.12 | *** join/#asterisk anthonyl (i=Anthony@nat/digium/x-b35b3acef406d9a3) |
| 16:54.55 | mercestes | Katty: Only for you..:) |
| 16:59.03 | buzzydex | out of interest whats the best phones to use with asterisk these days? |
| 17:00.08 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:00.12 | blitzrage | morning all |
| 17:00.20 | blitzrage | guess its technically afternoon here now |
| 17:01.01 | blitzrage | I have a g.729 license from digium that I'm using on one server, however that box is being reallocated to something other than asterisk. If I re-register the g729 driver on the other server, will it update? (i.e. work?) |
| 17:01.26 | Qwell[] | blitzrage: You nub, call support :P |
| 17:01.37 | Qwell[] | blitzrage: you have like 1-2 hostid changes allowed |
| 17:01.39 | blitzrage | Qwell[]: no way :) |
| 17:01.48 | blitzrage | Qwell[]: how annoying -- this is just for dev |
| 17:01.51 | Qwell[] | so, it...may work |
| 17:01.56 | blitzrage | guess I'll just switch my phone to use ulaw |
| 17:02.02 | Qwell[] | otherwise, support may need to "fix" it |
| 17:02.31 | blitzrage | ok cool. Might not have a choice as the provider I'm terminating to might only accept g729 (which would be annoying) |
| 17:02.52 | blitzrage | just waiting for asterisk to compile on this machine, which is why I didn't just test it myself. Had to kill some time ;) |
| 17:04.14 | *** join/#asterisk zmef420 (n=zmef420@metarb3-pool4-182.mtco.com) |
| 17:07.38 | *** join/#asterisk QbY_ (n=Kelvin@66.236.241.67.ptr.us.xo.net) |
| 17:08.10 | QbY_ | does anyone have any suggestions for the clipping of the first second of audio in an ivr? |
| 17:08.22 | Qwell[] | Answer() |
| 17:08.23 | Qwell[] | Wait(1) |
| 17:08.45 | QbY_ | k. |
| 17:09.41 | docelmo | Do a rain dance |
| 17:10.38 | b11d | rain_dance($locale) |
| 17:11.09 | docelmo | hehe |
| 17:11.22 | *** join/#asterisk rycan (i=ryan@nat/digium/x-3ba3fc57ae44fc08) |
| 17:14.00 | *** join/#asterisk h0 (n=fakhir@unaffiliated/fakhir) |
| 17:19.09 | *** join/#asterisk SwK[Work] (n=SwK@24.214.206.254) |
| 17:19.42 | *** join/#asterisk olinux (n=olinux@starbucks.wellspublishing.net) |
| 17:28.31 | b11d | TK... you're a source of logical statements.. |
| 17:28.37 | b11d | im getting rid of my |
| 17:28.42 | b11d | "9" requirement for outside lines.. |
| 17:28.48 | b11d | and now im thinking of getting rid of the "1" |
| 17:28.58 | b11d | do you think thats a mistake? |
| 17:30.26 | b11d | sounds good.. |
| 17:30.28 | b11d | i'll dot hat |
| 17:30.34 | blitzrage | exten => _NXXNXXXXXX,1,Do_Something |
| 17:30.35 | b11d | s/dot hat/do that |
| 17:30.45 | blitzrage | exten => _1NXXNXXXXXX,1,Goto(${EXTEN:1},1) |
| 17:30.56 | b11d | i've already got it all figured.. |
| 17:31.09 | b11d | i just dont always see into the future so clearly.. |
| 17:31.53 | blitzrage | hrmmm... nope -- ignore list is empty :) |
| 17:31.58 | b11d | nah he's around |
| 17:32.14 | blitzrage | just they way were talking it sounded like he was in a conversation with you |
| 17:32.22 | blitzrage | I didn't say he wasn't around |
| 17:32.29 | b11d | hehe.. i just responded as though he had replied to me |
| 17:32.34 | b11d | as though I had the conver in my head |
| 17:32.47 | blitzrage | that teaches me for paying attention in here |
| 17:33.01 | danp | hmm, i have a phantom call i can't get rid of. soft hangup didn't kill it |
| 17:33.17 | *** join/#asterisk bkw__ (n=brian@24.249.5.125) |
| 17:34.11 | *** join/#asterisk intralanman (n=lanman@pool-71-253-204-107.nrflva.east.verizon.net) |
| 17:38.08 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:39.34 | danp | is there something more forceful than soft hangup? |
| 17:39.39 | b11d | yeah |
| 17:39.41 | b11d | reboot the fucker |
| 17:40.07 | danp | hmm |
| 17:40.23 | CunningPike | :D |
| 17:40.39 | *** join/#asterisk LeadCrayon (n=leadcray@flpn-sentry.natcotech.com) |
| 17:41.03 | mercestes | danp: halt now :D |
| 17:41.33 | [TK]D-Fender | Occasionaly... this is "WORK" afterall :) |
| 17:44.28 | b11d | im not going to stop CunningPike.. |
| 17:44.30 | b11d | although i am trying.. |
| 17:44.33 | b11d | that last one was a slip :) |
| 17:44.42 | CunningPike | :) |
| 17:44.58 | *** join/#asterisk qdk (n=qdk@0x535eae17.boanxx9.adsl-dhcp.tele.dk) |
| 17:46.16 | *** join/#asterisk bkw_ (n=brian@adsl-70-143-49-53.dsl.tul2ok.sbcglobal.net) |
| 17:46.30 | b11d | i'll get it under control... |
| 17:47.50 | `Sean | i want a phone sigh |
| 17:47.51 | `Sean | :S |
| 17:48.00 | danp | i wish there was a way to send updated hints out to all the phones |
| 17:48.16 | CunningPike | danp: Explain? |
| 17:48.27 | danp | i restarted asterisk but my phones still think people are offhook from the phantom call i was trying to get rid of |
| 17:48.49 | Corydon-w | The phones are no longer subscribed, then |
| 17:49.15 | Corydon-w | Lower your re-registration interval to 10-20 minutes |
| 17:49.50 | *** join/#asterisk marv[work] (n=timr@24.214.206.254) |
| 17:50.51 | danp | i have them down to 10 minutes |
| 17:51.03 | b11d | thats weird eh |
| 17:51.17 | b11d | can you remotely restart the phones in question? |
| 17:52.02 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 17:52.37 | danp | not at the moment. it's not a huge deal but i hope it doesn't happen again :P |
| 17:53.58 | b11d | yeah strange though eh |
| 17:54.14 | danp | i wonder if i'm not hanging up where i should be |
| 17:54.59 | b11d | I thought asterisk had "autofallthrough" and hung up no matter what.. |
| 17:55.01 | b11d | after 1.2 |
| 17:56.56 | *** join/#asterisk bkw__ (n=brian@adsl-70-143-49-53.dsl.tul2ok.sbcglobal.net) |
| 17:57.25 | [TK]D-Fender | bkw_ :*boing* |
| 17:58.10 | *** join/#asterisk Skarmeth (n=Skarmeth@201009061013.user.veloxzone.com.br) |
| 17:58.10 | b11d | ftw! |
| 17:59.23 | *** join/#asterisk hoobastooba (n=ckwall@63.149.122.93) |
| 18:00.25 | hoobastooba | has anyone seen or heard of phpagi causing issues with memory on asterisk server? I was told once there were references to this on the mailing list, but I am not finding anything. |
| 18:04.07 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 18:06.24 | *** join/#asterisk Ebola (n=Ebola@host81-152-204-157.range81-152.btcentralplus.com) |
| 18:07.28 | *** join/#asterisk rpm (n=russell@66.119.170.34) |
| 18:08.56 | *** join/#asterisk benno2 (n=benno2@host23-110-dynamic.4-87-r.retail.telecomitalia.it) |
| 18:08.59 | *** join/#asterisk bkw_ (n=brian@adsl-70-143-49-53.dsl.tul2ok.sbcglobal.net) |
| 18:09.07 | zoa | hey ho brian |
| 18:11.30 | rpm | what do people in here recommend for a conferencing bridge.. should i use meetme or is there anything else which is open that works better? |
| 18:11.41 | hoobastooba | i use meetme and it works very well |
| 18:11.52 | b11d | meetme works great |
| 18:11.56 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-71-241-163-59.buff.east.verizon.net) |
| 18:12.48 | zoa | there is also app_conference |
| 18:12.50 | *** join/#asterisk Netgeeks_ (n=root@pbx5.netgeeks.net) |
| 18:12.52 | rpm | how about conferences with 100+ people? will meetme still work.. such as company wide announcements or will it have timing issues? |
| 18:12.55 | zoa | app_Conference uses more cpu |
| 18:13.16 | hoobastooba | rpm: so long as your server can support thatmany connections |
| 18:13.40 | zoa | hoobastooba: not really |
| 18:13.47 | zoa | he might run into timing problems earlier |
| 18:13.55 | zoa | i'd say try it |
| 18:14.03 | zoa | make some call files |
| 18:14.07 | zoa | put the calls in meetme |
| 18:14.11 | zoa | see how it sounds |
| 18:14.40 | benno2 | Hi, I have an asterisk box behind a linux NAT, browsing works well and standalone VOIP clients work well too (for example registering with an external SIP provider) but when I want asterisk registering to the same external SIP provider it does not work. tcpdumping on the linux NAT box shows me that UDP packets go towards the sipgate.de SIP provider but nothing comes back. see here my sip.conf. any idea ? thanks ! |
| 18:14.41 | benno2 | http://pastebin.ca/310366 |
| 18:14.48 | Netgeeks_ | Anyone know of any incompatabilites between 1.2 and 1.4 with respect to a re-invite? Or perhaps any problems with 1.4 and reinvited in general? |
| 18:17.12 | *** join/#asterisk kink0 (n=k@161.pool62-37-205.static.orange.es) |
| 18:17.50 | b11d | read the upgrading file, or the changelog maybe? |
| 18:18.03 | b11d | search 1.4 bugs for 'reinvite' as well? |
| 18:18.47 | benno2 | anyone having nat issues when registering asterisk with external sip providers ? |
| 18:19.30 | bcnl | b11d: sorry for a stupid question, but where can I go to see a summary of all open bugs in 1.4 |
| 18:19.42 | [TK]D-Fender | benno2: You are missing "externip", "localnet", and "can-reinvite=no" in [general] |
| 18:19.45 | *** join/#asterisk Assid (n=assid@202.88.132.238) |
| 18:19.53 | b11d | bugs.digium.org I think.. |
| 18:20.04 | wunderkin | canreinvite |
| 18:20.05 | [TK]D-Fender | benno2: And while you're at it remove that defined IP for bindaddr and put 0.0.0.0 |
| 18:20.15 | benno2 | [TK]D-Fender: thanks trying now |
| 18:20.27 | [TK]D-Fender | (no dash as mentioned, my typo) |
| 18:21.59 | benno2 | [TK]D-Fender: and as local net what should I use ? the net where my IP phones sit (eth0) is 192.168.0.1 (asterisk) , netmask 255.255.255 while I access the internet through eth1 local IP 192.168.2.3 gateway 192.168.2.1 ? so should I use 192.168.2.0 or 192.168.0.0 as localnet ? |
| 18:22.22 | [TK]D-Fender | benno2: Defin each local network on a seperate line |
| 18:22.42 | [TK]D-Fender | benno2: Routes will work themselves out after that. |
| 18:23.03 | benno2 | [TK]D-Fender: ah basically I have to add 2 localnetwork lines ? |
| 18:23.14 | [TK]D-Fender | benno2: Any SIP device on a local network will talk back to the closest IP for your box, and everything else will use your externip value |
| 18:23.19 | [TK]D-Fender | benno2: correct |
| 18:25.44 | benno2 | [TK]D-Fender: thanks, just did what you said, but tcpdump still shows me only outgoing packets, for example: 19:25:24.516278 192.168.0.1.5060 > ser02.de.sipgate.net.5060: udp 418 (DF) |
| 18:25.50 | benno2 | nothing coming back on the NAT gateway |
| 18:26.05 | *** join/#asterisk A-Tuin (n=a-tuin@gw.ip.v4.me.uk) |
| 18:26.16 | benno2 | but |
| 18:26.22 | benno2 | it could be due to my a bit silly setup |
| 18:26.30 | [TK]D-Fender | benno2: Did you reload sip? |
| 18:26.43 | [TK]D-Fender | benno2: and please pastebin it again so we can see what you did |
| 18:27.09 | Netgeeks_ | exit |
| 18:27.13 | Netgeeks_ | gah |
| 18:27.49 | benno2 | yes I did issue a reload command. (full reloade) for example : NATed asterisk box: eth0: 192.168.0.1, eth1: 192.168.2.3 gateway 192.168.2.1 -> NAT box: eth0: 192.168.0.1 eth0:0 192.168.2.1 ppp0 runs over eth0, it's the ADSL interface |
| 18:27.56 | *** join/#asterisk BSDTech (n=RNeese@69.239.85.71) |
| 18:28.28 | benno2 | I just added this: bindaddr=0.0.0.0 |
| 18:28.28 | benno2 | externip=87.4.110.23 |
| 18:28.28 | benno2 | localnet = 192.168.0.0/255.255.255.0 |
| 18:28.28 | benno2 | localnet = 192.168.2.0/255.255.255.0 |
| 18:28.53 | benno2 | I'm not sure if this eth0:0 aliasing makes problems |
| 18:29.47 | [TK]D-Fender | benno2: pastebin it please. |
| 18:30.16 | benno2 | ok |
| 18:31.29 | *** join/#asterisk nortex (n=breeves@snapper.titanspecialties.com) |
| 18:33.51 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqia.cable.mindspring.com) |
| 18:34.37 | *** join/#asterisk champster (n=asterisk@AH.tescogroup.com) |
| 18:35.08 | benno2 | [TK]D-Fender: http://pastebin.ca/310379 |
| 18:35.23 | benno2 | now I try to fire up ethereal on the nat router and see what's in the SIP packets |
| 18:35.44 | benno2 | I think asterisk does not write the correct IP into the sip packets |
| 18:36.06 | hardwire | why wouldn't it? |
| 18:36.17 | hardwire | and do you have funky nat support turned on? |
| 18:36.25 | [TK]D-Fender | benno2: it doesn't change the PACKET'S IP, just the rport header in the SIP info. |
| 18:36.27 | nortex | Got a question regarding parked calls, When a call is picked up from being parked, in my case they dial #700, should it be able to be parked again by dialing #700? I was sure it could and that I had even done it, but recently I have had people asking why it does not work. |
| 18:37.14 | nortex | To clarify, #700 should park the call in both cases. |
| 18:38.40 | *** join/#asterisk _Sam-- (n=sam@fresco.kneedraggers.com) |
| 18:40.00 | _Sam-- | hey how do you use X-lite's built in SIP instant mesasging with asterisk? i have 2 xlite clients, but when they try to send message to each other, " from "Sam L"<sip:sam@192.168.1.7>;tag=2377b241, dropped it... |
| 18:40.53 | *** join/#asterisk h0 (n=fakhir@unaffiliated/fakhir) |
| 18:41.20 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 18:42.32 | benno2 | [TK]D-Fender: using ethereal I see it sends an SIP OPTION packet: from: "astersik" <sip:asterisk@87.4.110.23>;tag=as76336e84 to: <sip:217.10.79.9> Contact: <sip:asterisk@87.4.110.23> but I don't see lSIP-id number |
| 18:46.04 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 18:46.56 | *** join/#asterisk shinux__ (n=shinux@196.220.27.111) |
| 18:48.16 | perd | anyone know where i can get cisco sip/skinny images? i cant seem to find them on ciscos page... |
| 18:48.19 | shinux__ | hey guys... is it possible for SIP clients on an asterisk service to send SMS to each other? is there any implementation for that? |
| 18:48.21 | sasch | for send message what is the function ?? |
| 18:48.39 | shinux__ | to send messages to each other |
| 18:48.57 | perd | i think you want email, not voip |
| 18:49.10 | shinux__ | im thinking of wifi VOIP clients |
| 18:49.18 | _Sam-- | xlite sends instant messages over SIP |
| 18:49.32 | _Sam-- | but i cant make it work with my older asterisk, yet |
| 18:49.38 | shinux__ | hmm |
| 18:49.45 | shinux__ | xlite in linux too? |
| 18:49.50 | sasch | yez |
| 18:49.56 | sasch | mac windows and linux |
| 18:50.08 | sasch | http://www.xten.com/index.php?menu=download |
| 18:50.22 | _Sam-- | do you know what ver. of asterisk is needed to use the SIP messaging? |
| 18:50.34 | _Sam-- | im on an old 1.2.5 that has been working perfectly, and am reluctant to upgarde. |
| 18:50.48 | _Sam-- | phone*CLI> show uptime |
| 18:50.48 | _Sam-- | System uptime: 24 weeks, 13 hours, 48 minutes, 22 seconds |
| 18:50.48 | _Sam-- | Last reload: 2 weeks, 2 days, 16 hours, 30 minutes, 24 seconds |
| 18:51.01 | shinux__ | no i dont |
| 18:51.04 | shinux__ | im using |
| 18:51.09 | sasch | i found it |
| 18:51.10 | sasch | http://www.voip-info.org/wiki/view/Asterisk+SIP+Messaging |
| 18:51.12 | file | _Sam--: it's not supported. |
| 18:51.22 | _Sam-- | file: hello! and thank you. |
| 18:51.45 | nortex | file, Any plans for it? |
| 18:52.10 | shinux__ | whats this link then, sasch |
| 18:52.11 | _Sam-- | file: you been up to anything good since last i IM'd about server related hassles? |
| 18:52.24 | hoobastooba | anyone here who uses phpagi to dial? I have a webpage that uses a php script to dial a lead... I am wondering if it is causing my memory issues... I am hoping that someone could look at my scripts or give me some helpful advice. it was suggested that posibly the script is not recognizing the hangup. |
| 18:52.27 | file | nortex: people want it, so it may appear |
| 18:52.58 | file | _Sam--: can't say I have |
| 18:53.19 | [TK]D-Fender | _Sam--: * does not support SIP messaging |
| 18:53.32 | *** join/#asterisk justinc123 (n=jcamp@host-64-179-18-177.spr.choiceone.net) |
| 18:53.38 | _Sam-- | thanks to both -- but the correct answer isnt that it doesnt support it -- its that it doesnt support it natively. |
| 18:53.52 | _Sam-- | the link sasch gave does give info how to make it work, and even a link to some patch for my version |
| 18:53.52 | *** join/#asterisk syzygyBSD (n=chatzill@poplar.matraex.com) |
| 18:54.04 | file | it's not supported in the SVN tree. |
| 18:54.20 | sasch | what is the URL of op panel |
| 18:54.35 | _Sam-- | wherever you copied the files to? |
| 18:54.37 | nortex | www.asternic.org |
| 18:55.11 | justinc123 | greetings all |
| 18:55.12 | sasch | <nortex> thanks |
| 18:55.29 | zoa | hey sam! |
| 18:55.34 | nortex | Got a question regarding parked calls, When a call is picked up from being parked, in my case they dialed #700 to park, should it be able to be parked again by dialing #700? I was sure it could and that I had even done it, but recently I have had people asking why it does not work. |
| 18:55.51 | _Sam-- | zoa! heya bud! how have you been??? |
| 18:55.55 | zoa | doing fine |
| 18:56.04 | _Sam-- | have you made the first 5 million dollars yet? |
| 18:56.11 | zoa | hehe no not really |
| 18:56.24 | _Sam-- | are you still developing / working on fisk? |
| 18:56.30 | zoa | of course :) |
| 18:56.33 | SheriF_SpacE | nortex: i didn't use parked calls before but i think u need to take a look at the features.conf file and some voip-info might help ? |
| 18:56.40 | Qwell[] | zoa: I tried running fisk the other day, and it wouldn't run on ubuntu :p |
| 18:56.46 | Qwell[] | idefisk rather |
| 18:57.03 | zoa | Qwell: it should run on ubuntu |
| 18:57.07 | sasch | there is a person that use ubuntu ?? |
| 18:57.08 | SheriF_SpacE | Qwell[]: i'm using it ;-) |
| 18:57.15 | zoa | my collegue uses it on ubuntu |
| 18:57.25 | nortex | SheriF_SpacE, Thanks, but already tried that. |
| 18:57.27 | sasch | I have ubuntu server with asterisk |
| 18:57.31 | sasch | and work all ok |
| 18:57.32 | _Sam-- | zoa: does your client do SIP instant messaging? |
| 18:57.41 | zoa | not yet |
| 18:57.53 | zoa | we have a version that does it |
| 18:57.53 | sasch | only with debian i get a problem ... |
| 18:57.58 | _Sam-- | the site you messaged, is a different product than the idefisk? |
| 18:57.59 | zoa | but its not public yet |
| 18:58.01 | zoa | yes |
| 18:58.05 | SheriF_SpacE | zoa: ur main devoloper of idefisk ? |
| 18:58.16 | zoa | no im the guy who pays the bills for idefisk :) |
| 18:58.23 | SheriF_SpacE | lol |
| 18:58.26 | SheriF_SpacE | cool:-) |
| 18:58.42 | _Sam-- | what is going to be the difference between fisk, and the site you showed me? |
| 18:59.49 | zoa | different company selling it, different licensing, features and look |
| 19:00.06 | _Sam-- | how come you wanted to compete against yourself with another client? |
| 19:00.08 | SheriF_SpacE | will not be open source? |
| 19:00.31 | _Sam-- | or, they licensed fisk, and customized it for their service? |
| 19:00.34 | SheriF_SpacE | _Sam--: he will get both money anyway ;-) |
| 19:00.39 | DonX | Is there any way to run a Macro when any extension in a context is dialed without having to add a macro line for each extension? |
| 19:00.44 | _Sam-- | i typically dont like to compete against myself. |
| 19:01.16 | justinc123 | use pattern matching? |
| 19:01.43 | DonX | well, say I have exten => 600 and exten => 610 |
| 19:01.52 | zoa | idefisk is not open source |
| 19:01.55 | zoa | and probably will never be |
| 19:02.04 | DonX | all within one context |
| 19:02.15 | SheriF_SpacE | zoa: fisk will ? |
| 19:02.20 | zoa | no |
| 19:02.23 | justinc123 | just do exten => _6xx,1,Macro(${EXTEN}) |
| 19:02.31 | SheriF_SpacE | Katty: voip-info has a very good howto |
| 19:02.46 | SheriF_SpacE | Katty: read it and if u have any problems tell me i'll show u my configuraions |
| 19:03.03 | DonX | Can I add one line into the context that will always run the macro before each extension is dialed? |
| 19:03.22 | justinc123 | sort of |
| 19:03.26 | justinc123 | pattern matching ... |
| 19:03.40 | DonX | hrmm |
| 19:03.42 | justinc123 | use a pattern that matches everything and then dial the exact pattern after |
| 19:03.46 | DonX | can you give me an example |
| 19:03.50 | justinc123 | er exact exten |
| 19:03.53 | DonX | ahhh |
| 19:03.55 | DonX | I see |
| 19:03.58 | SheriF_SpacE | Katty: cdr.conf almost has enable=yes only cdr_mysql,.conf has the connection strings and cdr_custom.conf if u want to have special cdr recrods |
| 19:04.01 | justinc123 | um exten => _xxx |
| 19:04.05 | justinc123 | something like that |
| 19:04.14 | justinc123 | check out voip-info.org for more on pattern matching |
| 19:04.22 | SheriF_SpacE | zoa: hmmm okay :-) |
| 19:04.28 | greekman | does anyone have detailed knowledge about re-invite? |
| 19:04.28 | justinc123 | or even exten => _. for everything :) |
| 19:04.44 | greekman | i have a very unique scenario |
| 19:05.02 | Katty | SheriF_SpacE: i found a doc at voip-info and i'm reading it. |
| 19:05.24 | Katty | SheriF_SpacE: it talks about asterisk 1.2, but when i do show version at the CLI it doesn't tell me which version of asterisk i'm running. |
| 19:05.32 | SheriF_SpacE | Katty: good best luck :-) need anything just hola |
| 19:05.35 | Katty | SheriF_SpacE: know of a quick way to get the actual version so i know where to check it out? |
| 19:05.40 | justinc123 | show version Katty |
| 19:05.41 | SheriF_SpacE | Katty: hmmmm ur kidding right :D? |
| 19:05.45 | justinc123 | CLI> show version |
| 19:05.47 | Katty | SheriF_SpacE: 'fraid not |
| 19:05.54 | [TK]D-Fender | justinc123>just do exten => _6xx,1,Macro(${EXTEN}) <- thats just broken..... |
| 19:06.06 | Katty | SheriF_SpacE: Asterisk CVS-HEAD built by root@BES.COPI-RITEINC on a i686 running Linux on 2005-06-08 18:35:10 <- the exact output. |
| 19:06.14 | justinc123 | sort of .. you should use ${MACRO_EXTEN} |
| 19:06.15 | SheriF_SpacE | Katty: when u connect to asterisk -vvr u'll find hte version writen |
| 19:06.19 | Katty | k |
| 19:06.22 | justinc123 | I was just giving a nutshell |
| 19:06.32 | SheriF_SpacE | Katty: oh ! why too old ? |
| 19:07.05 | Katty | well it didn't show a version there either |
| 19:07.28 | justinc123 | Katty: can you do a CLI> show version |
| 19:07.29 | justinc123 | ? |
| 19:07.29 | [TK]D-Fender | ${MACRO_EXTEN} <- another wasted value... just pass it as a parameter. |
| 19:07.32 | SheriF_SpacE | Katty: try the how to it will work anyway :-) i used it for 1.2 and 1.4 |
| 19:07.40 | Katty | k |
| 19:07.55 | justinc123 | huh fender? MACRO_EXTEN is a set variable once you're in the macro |
| 19:08.01 | SheriF_SpacE | Katty: http://www.voip-info.org/wiki/view/Asterisk+cdr+mysql |
| 19:08.01 | *** join/#asterisk s1gny|wrk (n=s1gny@p54915F16.dip.t-dialin.net) |
| 19:08.13 | *** part/#asterisk s1gny|wrk (n=s1gny@p54915F16.dip.t-dialin.net) |
| 19:09.26 | justinc123 | <PROTECTED> |
| 19:09.42 | Katty | [TK]D-Fender: what's 'svn'? |
| 19:09.51 | justinc123 | it's the new CVS :) |
| 19:10.00 | [TK]D-Fender | Katty: The replacement for CVS. Same idea, different app |
| 19:10.19 | Aurs | ~svn |
| 19:10.23 | jbot | extra, extra, read all about it, subversion is version control software. see http://subversion.tigris.org/ it aims to be a better CVS than CVS. |
| 19:10.23 | Katty | [TK]D-Fender: i did an apt-cache search for svn...is it listed under another name? |
| 19:10.38 | [TK]D-Fender | Katty: I'm unsure of its package title. |
| 19:10.42 | Aurs | Katty: subversion |
| 19:10.59 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-c4bcfeee8bef66f6) |
| 19:11.56 | justinc123 | does anyone know what digium plans to do with the agent channel in 1.4.x now that agentcallbacklogin is deprecated? |
| 19:12.01 | Katty | Aurs: thanks (= |
| 19:12.10 | Aurs | Katty: np ;) |
| 19:12.12 | justinc123 | it seems now you can't get agent status or events now :( |
| 19:12.31 | *** join/#asterisk asdfkjaslkdfjaks (n=Elive_us@eyesopen.org) |
| 19:13.26 | justinc123 | I'll take that elongated silence as a 'no' :( |
| 19:13.29 | asdfkjaslkdfjaks | does anyone know how I can route calls based on the ringtone that I get from my provider on a tdm400p card |
| 19:13.33 | file | justinc123: chan_agent wasn't originally made for callback agents |
| 19:13.43 | file | justinc123: so it's not going away, just that functionality most likely |
| 19:13.54 | justinc123 | what was it made for ? |
| 19:14.13 | Corydon-w | AddQueueMember and RemoveQueueMember |
| 19:14.18 | Katty | SheriF_SpacE: the wiki says i need mysql and the development packages installed. I have mysql running on a second box. do i still need the packages on the same box as asterisk? |
| 19:14.39 | justinc123 | I know how to replicate the agentcallbacklogin functionality within the dial plan .. the problem is we can no longer TRACK agent status |
| 19:14.39 | file | it was made for an individual to dial to log in, listen to hold music, and wait for someone to call in and get connected to them |
| 19:15.00 | justinc123 | so in other words it was designed for agentLogin |
| 19:15.27 | file | yes, callback agents were added afterwards - and were not kept in mind when it was originally written, which is part of the issue |
| 19:15.29 | justinc123 | is there any way to replicate the events and status that agentcallbacklogin USED to provide in the new system? |
| 19:15.51 | justinc123 | it's quite problematic for our setup not to have that functionality |
| 19:16.01 | syzygyBSD | Is there a way to get the correct number if someone sets their callerid to a fake one? |
| 19:16.05 | asdfkjaslkdfjaks | anyone know how I can route calls on an analog line if there are 2 numbers attached to it |
| 19:16.14 | asdfkjaslkdfjaks | syzygyBSD, yes |
| 19:16.21 | syzygyBSD | oh.. how? |
| 19:16.34 | asdfkjaslkdfjaks | syzygyBSD, usually requires having access to one of the switches |
| 19:16.46 | SheriF_SpacE | Katty: let me see |
| 19:16.50 | Corydon-w | syzygyBSD: based upon a lookup on their channel name |
| 19:16.58 | asdfkjaslkdfjaks | syzygyBSD, you cannot fool the switches and your telecom can give you access to those if they see it fit |
| 19:17.21 | syzygyBSD | oh... well, I don't have that, i was just wondering cuz some 800 numbers only send out the extension |
| 19:17.38 | Corydon-w | syzygyBSD: you're SOL |
| 19:18.00 | syzygyBSD | ya, it isn't that critical, just wondering. Once I get on the SS7 network it will be eaiser |
| 19:18.09 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 19:18.49 | SheriF_SpacE | Katty: i think u need to install the libmysql " headers " and recoming asterisk-addon |
| 19:18.54 | Corydon-w | You might check the ANI to see if the telco set it correctly |
| 19:19.07 | SheriF_SpacE | Katty: not sure never had asterisk on a box and database on the other box but give it a try |
| 19:19.38 | Corydon-w | That's ${CALLERID(ani)} |
| 19:19.49 | syzygyBSD | thanks Corydon-w |
| 19:19.50 | SheriF_SpacE | zoa: where i can get fisk ? or it's not public yet ? |
| 19:20.56 | justinc123 | is there a place to get better info about asterisk 1.4? voip-info seems a little out of date |
| 19:21.04 | asdfkjaslkdfjaks | nobody does routing based on ring tones anymore ? |
| 19:22.27 | hads | asdfkjaslkdfjaks: Use the distinctive ring features in zapata.conf |
| 19:22.52 | asdfkjaslkdfjaks | hads, ok and what about extensions.conf how do I specify the ring? |
| 19:23.12 | asdfkjaslkdfjaks | hads, I think I said it wrong not ring tones but dtmf tones .. right ?? the double ring etc... |
| 19:23.34 | justinc123 | file: thanks for the info bro ... back to work for me :) |
| 19:24.56 | Katty | SheriF_SpacE: the headers? |
| 19:25.09 | Katty | SheriF_SpacE: you mean the client library or the development files? |
| 19:26.37 | *** join/#asterisk Chris-NB (n=chris@argos.campus-sbg.at) |
| 19:27.39 | *** join/#asterisk FaithX (n=FaithX@ns.linuxterminal.com) |
| 19:27.40 | SheriF_SpacE | Katty: i'll tell u something it's stupied but safe :-) u can install mysql on the asterisk machine with the devlopment packages for it u compile the addson pacakges and don't use the mysql server on teh asterisk machine |
| 19:27.47 | *** join/#asterisk [tasty]freeze (n=yamahabr@204-181-50-10.skybest.com) |
| 19:28.17 | hads | asdfkjaslkdfjaks: Yes that is "ring tone" |
| 19:28.47 | hads | Why do you want to specify the ring? I thought you wanted to route based on it. |
| 19:28.49 | [tasty]freeze | Using the GXP-2000, I am trying to setup a common voicemail box for everyone to use. I have the message indicator light working, but when I press the msg button it sends me to voicemail instead of voicemail main control panel... anyone helpe? |
| 19:29.34 | asdfkjaslkdfjaks | hads, I found it in the voip-info.org article it is clear and simple.. thanks and yes I wanted to specify the ring so that I can direct the call through a different context |
| 19:30.38 | asdfkjaslkdfjaks | hads, thanks a lot |
| 19:31.06 | Katty | SheriF_SpacE: k |
| 19:31.32 | Katty | SheriF_SpacE: i'll give it a shot, thanks (= |
| 19:32.22 | *** join/#asterisk Strom_M (n=strom@dsl092-221-162.lax1.dsl.speakeasy.net) |
| 19:32.32 | SheriF_SpacE | Katty: ur welcome |
| 19:32.48 | SheriF_SpacE | Katty: i hope this will help but i'm so tired to think about something else :-) |
| 19:34.17 | Aurs | SheriF_SpacE, Katty: yes, mysql-devel is needed |
| 19:34.25 | Aurs | dont know package name in debian |
| 19:34.32 | Aurs | (mysql-devel in centos) |
| 19:34.36 | SheriF_SpacE | Aurs: libmysql --version |
| 19:34.45 | Katty | i've got a lot of libmysql package thingies. |
| 19:34.51 | Katty | i got the 14. |
| 19:34.58 | Katty | mysql-server is installing the 12 |
| 19:35.18 | Katty | no clue what the difference is. |
| 19:35.43 | Katty | i'm getting some errors though. |
| 19:36.32 | Katty | http://pastebin.com/854495 |
| 19:37.29 | Katty | i'm not sure what it's trying to tell me :< |
| 19:38.12 | [TK]D-Fender | Katty: Is that a brand new version of asterisk-addons? |
| 19:38.29 | *** join/#asterisk nays85 (i=nays85@shell.thehostbusters.com) |
| 19:39.30 | SheriF_SpacE | [TK]D-Fender: Katty: i think Katty using an old CVS version of the addons on too not sure it's supports mysql4.1 or not |
| 19:40.03 | *** part/#asterisk bcnl (n=mike@24.86.64.144) |
| 19:40.09 | *** join/#asterisk bcnl (n=mike@24.86.64.144) |
| 19:40.28 | SheriF_SpacE | [TK]D-Fender: u mean she might been usng a new adds-on pkg ? |
| 19:40.32 | Katty | [TK]D-Fender: that's 1.2, accordding to what the wiki told me to use. |
| 19:40.48 | [TK]D-Fender | Katty: You need to find the version that matches your release of * IIRC. |
| 19:40.59 | Katty | [TK]D-Fender: show version didn't show anything. know of another way to get it? |
| 19:41.42 | SheriF_SpacE | Katty: why u don't upgrade the asterisl server 1st and port ur configrautions ? |
| 19:41.52 | SheriF_SpacE | Katty: or it's production server ? |
| 19:41.57 | PupenoR | How do you unpause a memer of a queue ? |
| 19:41.57 | Katty | it's a production server. |
| 19:42.09 | Katty | i think i should just up to 1.4 |
| 19:42.15 | [TK]D-Fender | Katty: You may be forced to upgrade.... I;m not sure how to match with the CVS version you're running.... |
| 19:42.15 | Katty | this is silly. |
| 19:42.21 | Katty | k |
| 19:43.11 | mercestes | Katty: What version of * are you running? |
| 19:43.42 | SheriF_SpacE | mercestes: a CVS-head back to 2005 |
| 19:44.07 | mercestes | oh, so it's not even a version. |
| 19:44.10 | Katty | mercestes: show version doesn't tell me. |
| 19:44.12 | mercestes | 1.09? |
| 19:44.13 | mercestes | lol |
| 19:44.15 | Katty | no clue |
| 19:44.19 | mercestes | nice. |
| 19:44.27 | Katty | i'm just going to setup a new server and put 1.4 on it |
| 19:44.32 | mercestes | it may be time to upgrade babe. |
| 19:44.34 | Katty | they can bite me if they don't like it. |
| 19:44.39 | mercestes | ....can I? |
| 19:44.41 | Katty | no |
| 19:44.46 | mercestes | ^;.;^ |
| 19:44.51 | mercestes | aw. |
| 19:45.20 | SheriF_SpacE | lol |
| 19:45.24 | SheriF_SpacE | Katty: they will like it :-) the gtalk channel is cool ;-) |
| 19:45.43 | SheriF_SpacE | but i hate that why my connection gose down the japper dosn't connect again i have to restart asterisk :( |
| 19:45.51 | *** join/#asterisk jm|laptop (n=jamie@dilbert.jamiem.com) |
| 19:47.10 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 19:47.51 | Katty | i shouldn't have to work on production boxes in the first place. |
| 19:48.11 | mercestes | Katty: I try to avoid that |
| 19:48.25 | Katty | k, all better. |
| 19:48.29 | mercestes | :) |
| 19:48.48 | Katty | i'm so unmovitated. |
| 19:49.36 | mercestes | me too |
| 19:49.45 | mercestes | today has just been a really shiesy day. |
| 19:49.46 | Katty | i should have worked at a zoo |
| 19:49.48 | [TK]D-Fender | Katty: I usually find cheese or elcto-shock highly functional, and at the very least, entertaining :) |
| 19:50.06 | mercestes | [TK]D-Fender: oooo...I could use some of that right now |
| 19:50.09 | cpm | is that like a merdey day? |
| 19:50.10 | [TK]D-Fender | Katty: "If I'm not at the circus, then why am I surrounded by clowns?" |
| 19:50.26 | mercestes | cpm: Yea, it's kinda like a mercy day. Except worse |
| 19:50.42 | Katty | [TK]D-Fender: i'd rather be one of those people that make lunch for all the animals. |
| 19:51.19 | mercestes | I'd rather be the guy that gets shot out of the cannon |
| 19:51.26 | SheriF_SpacE | few days i thought i could be a taxi driver :D |
| 19:51.34 | SheriF_SpacE | Katty: what u have done to the production server? |
| 19:51.41 | Katty | SheriF_SpacE: nothing. |
| 19:52.28 | SheriF_SpacE | Katty: ah okay :-) |
| 19:52.44 | SheriF_SpacE | i'll make a new server with 1.4 so the megration will be smoth :-) |
| 19:55.45 | killfill_ | hi |
| 19:55.51 | Katty | allo. |
| 19:56.00 | killfill_ | anyones uses asteirtsk with google talk? |
| 19:56.09 | killfill_ | im not sure how this works.. |
| 19:56.12 | killfill_ | ive set this up |
| 19:56.13 | Katty | i used it with yahoo for awhile |
| 19:56.22 | Katty | but it was too sloooooooowwww |
| 19:56.38 | Katty | s/too/to |
| 19:56.51 | Corydon-w | Katty: it was right the first time |
| 19:56.59 | Katty | Corydon-w: mew? |
| 19:57.09 | Katty | Corydon-w: you do not parse. |
| 19:57.17 | Katty | Corydon-w: or my memory is really really bad. |
| 19:57.23 | killfill_ | http://www.voip-info.org/wiki/view/Asterisk+Google+Talk |
| 19:57.51 | Corydon-w | "to" assigns. "too" indicates excess |
| 19:57.56 | killfill_ | how do i debug this?..my pbx user is not shoing up in the gtalk network.. :-S |
| 19:59.07 | Katty | i thought too indicated also. |
| 19:59.35 | mercestes | Katty: It does. |
| 19:59.54 | Katty | i could understand /too/ much slowness. |
| 20:00.29 | PupenoR | Katty: it does when it is at the end. "I installed Asterisk too." |
| 20:00.50 | mercestes | English is stupid. |
| 20:01.12 | Katty | this is why i wasn't an english major. |
| 20:02.20 | Katty | however! |
| 20:02.29 | Katty | i think i'm going to blog how to put asterisk onto a dell poweredge 2600 |
| 20:03.51 | SheriF_SpacE | killfill_: yes i do |
| 20:04.24 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 20:04.36 | SheriF_SpacE | Katty: one of my backup servers got kernel panic :-s |
| 20:04.41 | SheriF_SpacE | hardidks crash :( |
| 20:05.10 | killfill_ | SheriF_SpacE: should i see the user logged in into gtalk network? |
| 20:05.17 | SheriF_SpacE | killfill_: follow the howto :-) it will show u the light i did it and it works perfect i think leave debug=yes as option |
| 20:05.19 | Katty | great. |
| 20:05.22 | SheriF_SpacE | killfill_: sure u'll |
| 20:05.24 | Katty | i sure called that one. it did kernel panic. |
| 20:05.29 | Katty | sigh. |
| 20:05.35 | SheriF_SpacE | Katty: lol |
| 20:05.35 | killfill_ | hm.. |
| 20:05.39 | killfill_ | its not doing it.. :-S |
| 20:05.48 | DonX | Can I make it where Goto doesn't hangup if it can't find the context? |
| 20:06.01 | SheriF_SpacE | killfill_: pastebi ur confitrations for japper.conf and gtalk.conf and hash ur password please |
| 20:06.02 | mercestes | Katty: You should. It's mean. |
| 20:06.40 | Katty | i'm hungry. |
| 20:06.49 | mercestes | Katty: blog * on the PE2600 I mean.... not kernel panic. |
| 20:07.05 | Katty | why's a PE2600 mean? |
| 20:07.10 | killfill_ | ok letme put it how it was |
| 20:07.11 | DonX | There is a priority under one of my Goto strings but when the target context doesn't exist, it exits without getting to that next priority |
| 20:07.15 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 20:09.56 | mercestes | Katty: Power Edge 2600 |
| 20:10.05 | Katty | mercestes: that's not what i asked. |
| 20:10.06 | killfill_ | SheriF_SpacE: http://pastebin.ca/310476 |
| 20:10.09 | Katty | mercestes: i asked /why/ |
| 20:10.15 | mercestes | Katty: Oh. |
| 20:10.24 | Katty | mercestes: whywhywhy. pretend i'm 2 |
| 20:10.41 | mercestes | Katty: See, I read that and didn't understand and it made more sense to me with "what" so I changed it in my head lol |
| 20:11.04 | Katty | mercestes: you said a PE2600 was mean. i asked why. |
| 20:11.36 | mercestes | Katty: Because I have a PE with an Intel Xeon 64bit processor, which has the EMT64 architecture, which uses the AMD64 package of Gentoo and not the X86_64 package like I would expect it to use. |
| 20:11.53 | Katty | that sounds horribly complicated. |
| 20:11.53 | mercestes | Katty: Unless it's the 32bit Intel Xeon then it uses the X86 package. |
| 20:12.12 | mercestes | Katty: Yea, it is. |
| 20:12.16 | Katty | mercestes: mine has a xeon. |
| 20:12.18 | killfill_ | im not sure how it works.. |
| 20:12.22 | mercestes | It's not really the PE that's mean, it's the Intel Xeon |
| 20:12.27 | killfill_ | SheriF_SpacE: does that look too bad?.. :-p |
| 20:12.59 | [TK]D-Fender | DonX: Pastebin the CLI output of your failed call as well as your dialplan please. |
| 20:13.20 | DonX | I found a solution [TK]D-Fender... Thanks :) |
| 20:13.36 | DonX | the context exists but the extension doesn't |
| 20:13.41 | *** join/#asterisk waverly360 (n=waverly@209.12.249.243) |
| 20:14.02 | SheriF_SpacE | killfill_: sounds very good |
| 20:14.26 | SheriF_SpacE | killfill_: ok restart asterisk and go to CLI u should see lots of XML requests done by asterisk to register to gtalk network |
| 20:14.39 | SheriF_SpacE | anyone did RSS + festival + asterisk ? |
| 20:14.45 | mercestes | Use gentoo |
| 20:14.45 | SheriF_SpacE | Katty: oh why !? |
| 20:14.56 | SheriF_SpacE | mercestes: ah so she wait 14 day :P? |
| 20:14.57 | Katty | cause i want to, clearly. |
| 20:14.59 | SheriF_SpacE | +s |
| 20:15.18 | Katty | it gives me time to browse old navy's website for shiny new shirts. |
| 20:16.56 | killfill_ | SheriF_SpacE: i see nothing.. :-S maybe im missingamodule? |
| 20:17.05 | killfill_ | how wouldi check this out? |
| 20:17.46 | SheriF_SpacE | killfill_: try to module reload chan_gtalk.so |
| 20:18.34 | jm|laptop | Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. ZT_CHANCONFIG failed on channel 1: No such device or address (6) |
| 20:18.35 | SheriF_SpacE | killfill_: and also check /usr/lib/asterisk/modules for chan_gtalk.so and res_japper |
| 20:18.54 | killfill_ | hm.. they are not there.. :-S |
| 20:19.02 | SheriF_SpacE | jm|laptop: try from terminal ztcfg -vvvvvvvv as root |
| 20:19.03 | jm|laptop | I have tried this card (which was working perfectly for months) in two different machines using two different OSes - I take it the card is fubarred :( |
| 20:19.24 | jm|laptop | SheriF_SpacE: ok two ticks just restarting the box :) |
| 20:19.31 | perd | anyone knwo where i can get sip/skinny images for cisco 7902/7912/7960 ? |
| 20:19.39 | SheriF_SpacE | killfill_: there was a lib u compile then it's better u reamove the old asterisk and recomipile again .. it did give me some problems. |
| 20:19.52 | perd | i guess 7902 has no sip.. so just skinny on that... i cant find the damn things on ciscos site and i'm only a guest |
| 20:19.58 | perd | sucks to be me |
| 20:20.18 | killfill_ | hm.. |
| 20:20.37 | [TK]D-Fender | perd: Should have checked the compatability charts..... |
| 20:20.48 | perd | these phones were bought way before i came along |
| 20:21.06 | [TK]D-Fender | perd: Goto(suckstobeyou,s,1) |
| 20:21.13 | jm|laptop | ffs! now the box won't boot!! |
| 20:21.25 | killfill_ | SheriF_SpacE: i just noticed, in the tutorial url, tthere are differen @gmail users.. whats buddyusername@gmail.com and whats username@gmail.com? |
| 20:22.14 | SheriF_SpacE | killfill_: not sure but it works even without buddy=balbal@gmail.com |
| 20:22.26 | killfill_ | hehe |
| 20:22.27 | SheriF_SpacE | anyone did RSS + festival + asterisk ? |
| 20:24.25 | DonX | I'm trying to match a network in my sip.conf so that I can make it dial through a certain context. It works if I use host but I can't figure out how to do it for the entire network. Does anyone know how I could accomplish this? |
| 20:24.43 | *** join/#asterisk Aces1Up (n=rich@ip68-96-234-180.lv.lv.cox.net) |
| 20:24.55 | killfill_ | hm.. |
| 20:25.14 | killfill_ | ive selected chan_gtalk and red_jabber in make menuconfig |
| 20:25.26 | killfill_ | but make clean&&make does compile the .o's.. :-S |
| 20:26.36 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 20:28.43 | Katty | i need a snack. |
| 20:30.25 | *** join/#asterisk tsurko (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 20:31.13 | *** join/#asterisk BlepsoaF (n=pbaker@nnat-gw.adeptra.com) |
| 20:31.40 | syzygyBSD | oh.. snack... |
| 20:31.50 | BlepsoaF | hello all, this is sort of an off topic asterisk question - but does anyone use polycom telephones here? If so, does anyone know of a way to have the phone display dialed digits after the call has been connected? |
| 20:32.29 | [TK]D-Fender | BlepsoaF: Yes & no, respectively. |
| 20:32.30 | syzygyBSD | wow, if polycom phones are off topic 50% of the discussion here is off topic |
| 20:32.32 | *** join/#asterisk h0 (n=fakhir@unaffiliated/fakhir) |
| 20:32.38 | greekman | lol |
| 20:32.54 | `Sean | omg Jan 9 03:38:06 WARNING[5754]: chan_zap.c:1584 zt_set_hook: zt hook failed: Device or resource busy |
| 20:32.55 | syzygyBSD | my polycom tells me who I dialed |
| 20:32.55 | `Sean | fuck fuck |
| 20:32.55 | BlepsoaF | [TK]D-Fender: ok thanks, I cant find anything in the Admin manual :S |
| 20:33.30 | BlepsoaF | syzygyBSD: but I assume, once the call is connected it doesnt show dialed digits, IE if you were to go through an IVR ( our QA team needs this ) |
| 20:33.39 | *** join/#asterisk Ast001 (n=uros@77-105-53-82.adsl-1.sezampro.yu) |
| 20:34.07 | Ast001 | good evening |
| 20:34.13 | syzygyBSD | ahh, ok, ya, don't see those |
| 20:34.27 | [TK]D-Fender | BlepsoaF: Polycom phones also don't support DOC to ODF coversion, MPEG2 -> DivX conversion, remote HD Defragmentation either. Just in case you were looking for that in the manual as well ;) |
| 20:34.55 | syzygyBSD | lol [TK]D-Fender |
| 20:35.17 | syzygyBSD | isn't someone working on porting linux to polycom phones? |
| 20:35.44 | syzygyBSD | then it would support all of those things |
| 20:35.44 | Ast001 | Is there anyway to force fxo device with ls signalling to hang up when remote phone hang up the line ? |
| 20:35.52 | [TK]D-Fender | syzygyBSD: Nothing I've heard of.... wouldn't think anyone would bother. |
| 20:35.57 | Ast001 | signaling ls |
| 20:36.02 | BlepsoaF | [TK]D-Fender: Nor would I have expected it to |
| 20:36.02 | nortex | When I pick up a parked call, should I be able to use # to transfer the call? |
| 20:36.16 | BlepsoaF | nortex: look at features.conf |
| 20:36.47 | *** join/#asterisk greendisease (n=jack@fedora/greendisease) |
| 20:36.49 | syzygyBSD | [TK]D-Fender: a lot of people do things that I wouldn't think anyone would bother with |
| 20:37.12 | [TK]D-Fender | syzygyBSD: AEL is higher on my list right now.... |
| 20:37.22 | tset | nortex are you using a T or a t in your dial command in extensions.conf? |
| 20:37.32 | nortex | BlepsoaF, For what? The # to transfer is already active. In fact I use #700 to park the caller? |
| 20:38.02 | tset | #700 is to park, #701-720 is to unpark and #700 is to park again (to some park # that is not in use) |
| 20:38.09 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 20:38.40 | nortex | tset, Everywhere I can think of, but when a person answers the parked, say 701, then it appears * quits watching for the # key. |
| 20:39.03 | tset | that depends on if the reciever is allowed to do that function or not |
| 20:39.20 | tset | its in your dial command in extensions.conf where that part is handled |
| 20:39.23 | tset | i just set it up a moment ago :) |
| 20:39.25 | PupenoR | I have created some queues pointing to some SIP extensions, they all show as "(Invalid) has taken no calls yet" and calling the queues shows "Unable to join queue 'all-phones'". Now, what makes it really weird is that if I pick one phone that is in the queue and make calls to the queue untill it tries to ring itself, the queue starts to work. Any idea what's going on here ? |
| 20:40.22 | Aces1Up | is anyone familiar with a2billing? having some issues when printing out invoices when using the calling card id number. |
| 20:40.44 | Aces1Up | getting the following error : Fatal error: Call to a member function on a non-object in /var/www/html/a2billing/Public/invoices.php on line 416 |
| 20:40.46 | killfill_ | hey syzygyBSD |
| 20:40.58 | killfill_ | have you compilet red_jabber.so on FreeBSD? |
| 20:41.00 | nortex | tset, So how do I make sure that when someone picks up 701 they are still able to transfer the call using # |
| 20:41.07 | PupenoR | Agent members of a queue are shown in state "(Unavailable) has taken no calls yet". |
| 20:41.14 | killfill_ | it seem to use libnsl |
| 20:41.16 | killfill_ | wich is a linux thingy?.. |
| 20:41.27 | tset | nortex: this is a quick read and will help you figure it out: http://www.voipplanet.com/backgrounders/article.php/3632556 |
| 20:41.32 | [TK]D-Fender | nortex: * basic parking blows, app_valetparking FTW! |
| 20:41.40 | Aces1Up | killfill you talking to me? |
| 20:42.13 | killfill_ | Aces1Up: you use FreeBSD? |
| 20:42.17 | tset | nortex: [TK]D-Fender is right, the basic parking sucks, its not specific to queues or depts, depending on how you want to seperate things... use this one only as a basis if you need something more complex |
| 20:42.24 | Aces1Up | fill no centos |
| 20:42.54 | killfill_ | ah ok.. |
| 20:43.19 | *** join/#asterisk _paulo_ (n=_PauloS@mail.eletrodireto.com.br) |
| 20:43.25 | syzygyBSD | I don't use freebsd, sorry |
| 20:43.28 | nortex | I only need to basicly put the caller on hold and get another person to pick it up, and I am almost certain this worked before. |
| 20:43.34 | _paulo_ | hi people! |
| 20:43.42 | Aces1Up | killfill you have any suggestions? |
| 20:44.26 | *** join/#asterisk backblue (n=moo@87-196-107-2.net.novis.pt) |
| 20:44.27 | killfill_ | Aces1Up: to you error? nope |
| 20:44.29 | tset | nortex: that link should help you solve your issue |
| 20:44.53 | Ast001 | Someone is calling from PSTN I answer from XLite phone but when he hangs up I still hear pots signall and asterisk is recording.Its loopstart signalling issue.So is there any way to force asterisk to hang up near end when far end hangs up ? |
| 20:46.04 | _paulo_ | Im using the "monitor" feature in queues.conf |
| 20:46.04 | [TK]D-Fender | Ast001: First make sure your telco is using Call Disconnect Supervision |
| 20:46.33 | _paulo_ | but the recording stops if the calee transfers call to another extension. |
| 20:46.41 | Ast001 | I tried to change loop start to kewel start but it can not recieve call in that case |
| 20:46.54 | Ast001 | same is with ground start |
| 20:47.23 | _paulo_ | is there anithing I can do to continue recording even if the call is transfered? |
| 20:48.08 | nortex | tset, thanks for the link but the is nothing there I don't already have in play. In fact if you follow their test by calling in via cell phone and have ther person park your call, then you pickup the parked call then try to transfer it with # the DTMF goes right to the cell phone. |
| 20:48.45 | [TK]D-Fender | Ast001: I said check with your TELCO. This is not your server I'm talking about... |
| 20:48.55 | tset | nortex: np, one thing you maybe missing is the extension which is picking up the call maybe missing a include parked calls file |
| 20:49.02 | [TK]D-Fender | Ast001: And if you can't support Kewlstart, this may be a problem. |
| 20:49.58 | Ast001 | I cant.When I change signalling to fxs_ks it can not recieve or place a call. |
| 20:50.20 | Ast001 | Thanks I'll check with telco but I doubt they use that service |
| 20:50.34 | tset | nortex: whatever context that phone uses to make calls, you should test it out by doing a "include => parkedcalls" |
| 20:51.07 | nortex | tset, Nope, all extensions are in the inside context, which includes features, which includes parkedcalls |
| 20:51.12 | tset | nortex: if that doesnt work, no idea unfortunately, mine seems to work, ill have to test it further to make extra certain :) |
| 20:52.13 | killfill_ | woot |
| 20:52.25 | killfill_ | the file makeopts |
| 20:52.29 | killfill_ | contains -lnsl |
| 20:52.36 | killfill_ | wich is not avaible in all plataforms |
| 20:52.43 | nortex | tset, Thanks for trying I'll keep digging, since I have 7 locations doing the exact same thing. |
| 20:52.45 | killfill_ | IKSEMEL_LIB contains it |
| 20:52.59 | killfill_ | it should be IKSEMEL_LIB=-L/usr/local//lib -liksemel -lgnutls -lz -lgcrypt -lgpg-error i think.. :-p i.e. withouth it |
| 20:53.35 | tset | nortex: your welcome, if i can think of something else for you i will let you know. good luck |
| 20:54.46 | killfill_ | not avaible/not necesary |
| 20:55.07 | *** join/#asterisk AlfaScorpii (n=santo@host86.201-252-129.telecom.net.ar) |
| 20:55.22 | AlfaScorpii | hello |
| 20:55.29 | *** join/#asterisk zapp-branigan (n=zapp-bra@81-202-140-56.user.ono.com) |
| 20:55.34 | _paulo_ | is there anithing I can do to continue recording even if the call is transfered (recording stops whe user do atxfer in my setup)? |
| 20:56.27 | _paulo_ | ~/whe/when/ |
| 20:57.13 | AlfaScorpii | Alguien en español |
| 20:57.49 | AlfaScorpii | hola |
| 20:58.07 | zapp-branigan | hi there is a problen of number of calls if i use a asterisk 1.2.14 or i must to install some pach to work fine ? |
| 20:58.12 | zapp-branigan | alga yo |
| 20:58.21 | zapp-branigan | alfa |
| 20:58.25 | zapp-branigan | k tal |
| 20:58.36 | AlfaScorpii | zapp-branigan: hola! muy bien y tu? |
| 20:58.40 | zapp-branigan | je je |
| 20:58.48 | zapp-branigan | usas el asterisk? |
| 20:58.56 | zapp-branigan | o estas aprendiendo ? |
| 20:59.08 | AlfaScorpii | zapp-branigan: estoy intentando poner a funcionar nuevamente un asterisk que quedo sin uso hace tiempo |
| 20:59.22 | AlfaScorpii | zapp-branigan: no logro hacer llamadas salientes |
| 20:59.26 | zapp-branigan | pues yo estoy empezando :( |
| 20:59.35 | zapp-branigan | usas zap |
| 20:59.37 | zapp-branigan | ? |
| 21:00.10 | AlfaScorpii | zapp-branigan: SIP |
| 21:00.11 | *** join/#asterisk [hC] (n=hardcore@S0106000fb51cc225.vf.shawcable.net) |
| 21:00.35 | _paulo_ | AlfaScorpii: show sip peers |
| 21:00.48 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 21:01.34 | zapp-branigan | you have sip ports open ? |
| 21:01.41 | zapp-branigan | try iax |
| 21:01.56 | zapp-branigan | whith ono do not work fine sip |
| 21:02.12 | _paulo_ | AlfaScorpii: sip show registry |
| 21:03.19 | AlfaScorpii | _paulo_: im usig freepbx |
| 21:03.34 | *** join/#asterisk Strom_M (n=strom@dsl092-221-162.lax1.dsl.speakeasy.net) |
| 21:04.06 | syzygyBSD | AlfaScorpii: you poor thing, have you thought about rehab? |
| 21:04.06 | zapp-branigan | hi there is a problen of number of calls if i use a asterisk 1.2.14 or i must to install some pach to work fine ? |
| 21:04.15 | SheriF_SpacE | good night guys |
| 21:05.07 | [TK]D-Fender | AlfaScorpii: ... |
| 21:05.10 | _paulo_ | AlfaScorpii: sorry, I know asterisk only. |
| 21:05.22 | [TK]D-Fender | ~freepbx |
| 21:05.24 | jbot | freepbx is probably , NOT supported here! It is a complex piece of software and can't be easily supported by people who aren't deeply involved. Please join #freepbx and try asking your question there. |
| 21:06.27 | _paulo_ | AlfaScorpii: no soportamos freepbx aqui. Ya intentaste en #freepbx? |
| 21:08.13 | AlfaScorpii | gracias |
| 21:08.19 | AlfaScorpii | seguire intentando |
| 21:09.15 | killfill_ | can i chat (text) with sip sofwtarephones? |
| 21:09.53 | AlfaScorpii | debo usar freepbx para configurar asterisk desde una pc windows via web |
| 21:10.40 | killfill_ | freebsd es una tontera.. use asterisk-gui ;-) |
| 21:10.47 | Aboulafia | Hi.. I need some help about call transfer : I need to make attended call transfer with the manager... it's possible ? |
| 21:11.08 | _paulo_ | AlfaScorpii: hace un rato largo que no charlo en #asterisk, pero me acordo que a la gente aca no les gusta que hablen otro idioma que no sea ingles... |
| 21:11.30 | zapp-branigan | k va son muy majos |
| 21:11.52 | *** join/#asterisk Dr-Linux|home (n=Dreamer@DSL-202-59-73-131.nexlinx.net.pk) |
| 21:12.17 | Dr-Linux|home | anybody tried LumenVox recognition? |
| 21:13.36 | jm|laptop | :) |
| 21:13.56 | jm|laptop | Debian is taking over my home lan :( |
| 21:15.53 | giesen | is there a way to read asterisk variables from outside of asterisk? |
| 21:15.57 | giesen | from a shell script for example? |
| 21:16.08 | _paulo_ | Aboulafia: dont you mean unattended transfer? |
| 21:16.29 | *** part/#asterisk AlfaScorpii (n=santo@host86.201-252-129.telecom.net.ar) |
| 21:16.36 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 21:16.58 | _Sam-- | giesen " |
| 21:16.58 | _Sam-- | Asterisk AGI |
| 21:16.59 | _Sam-- | <PROTECTED> |
| 21:16.59 | _Sam-- | AGI -- |
| 21:16.59 | _Sam-- | AGI is the Asterisk Gateway Interface, an interface for adding functionality to asterisk with many different programming languages. Perl, PHP, C, Pascal, Bourne Shell - it's your choice, really. |
| 21:17.21 | Aboulafia | _paulo_: oops :) i suppose not, I mean that its for a standard, the standard have to ask for whom, call this personn and ask to him if he would like to speak to him... |
| 21:17.33 | *** join/#asterisk _Sam-- (n=sam@fresco.kneedraggers.com) |
| 21:17.43 | _Sam-- | <oops - sorry> |
| 21:17.45 | _Sam-- | obviously copy didnt work :) |
| 21:18.11 | *** join/#asterisk markit (n=konversa@host119-245-static.72-81-b.business.telecomitalia.it) |
| 21:18.22 | _Sam-- | giesen: AGI -- |
| 21:18.23 | _Sam-- | AGI is the Asterisk Gateway Interface, an interface for adding functionality to asterisk with many different programming languages. Perl, PHP, C, Pascal, Bourne Shell - it's your choice, really. |
| 21:18.36 | giesen | yeah but that has to be run from within asterisk |
| 21:18.48 | markit | hi, is it normal that if I try to assign to a variable a database key that has not yet been assigned, it produces a syntax error? any way to "initialize" database variables from dialplan automatically at startup? |
| 21:19.05 | giesen | I want a cron job to run a script |
| 21:19.07 | _Sam-- | i use php to send stuff directly to my asterisk...i dont know what you want |
| 21:19.11 | giesen | to check the status on an asterisk variable |
| 21:19.14 | *** join/#asterisk terrapen (n=cjs@66.29.164.42.static.utahbroadband.com) |
| 21:19.17 | terrapen | howdy |
| 21:19.31 | *** part/#asterisk Ast001 (n=uros@77-105-53-82.adsl-1.sezampro.yu) |
| 21:19.32 | terrapen | has anybody ever seen a commercial, solid-state T3 to SIP gateway? |
| 21:19.34 | Aces1Up | is there a program that will monitor a say broadband connection and let me know when use is beginning to exceed its capabilities? |
| 21:19.49 | terrapen | aces, ntop, cacti, mrtg2, et c |
| 21:19.49 | _Sam-- | giesen: New in Asterisk v1.2.11: GET VARIABLE can now retrieve global variables (see bug 7609) |
| 21:20.01 | Aces1Up | to see if a certain amount of concurrent phone calls going out is reaching the limit of the connection? |
| 21:20.08 | terrapen | cacti |
| 21:20.12 | terrapen | nagios |
| 21:20.42 | giesen | yeah but the AGI has to be called from within asterisk |
| 21:20.46 | giesen | if Im not mistaken |
| 21:21.05 | terrapen | does anybody in here have a voice T3? |
| 21:21.26 | _Sam-- | you can use asterisk -x |
| 21:21.29 | monsted | terrapen: we have lots, but i don't get to play with them :) |
| 21:21.50 | monsted | well, STM1's, i think they are |
| 21:22.08 | terrapen | i need to bring in a whole arseload of channels |
| 21:22.23 | monsted | well, you could get an SS7 link |
| 21:22.27 | terrapen | but i'd rather not bring them in directly to an * server |
| 21:22.38 | terrapen | but rather a media gateway and then through SER |
| 21:22.59 | _paulo_ | terrapen, why not? |
| 21:23.00 | _Sam-- | geisen : example: asterisk -rx 'show channels' |
| 21:23.06 | _Sam-- | from command line |
| 21:23.10 | _Sam-- | <not asterisk console> |
| 21:23.25 | terrapen | scalability... |
| 21:23.33 | terrapen | i want to load balance a bunch of SER gateways |
| 21:23.51 | JerJer | ser is a proxy not a gateway |
| 21:23.58 | terrapen | thats what i maent |
| 21:24.02 | terrapen | proxy |
| 21:24.15 | monsted | terrapen: get a cisco mgw 8880, an SS7 link and SIP away |
| 21:24.23 | terrapen | anyway, when one of my users calls out, i don't want them to touch * |
| 21:24.54 | JerJer | monsted: you mean mgx ? |
| 21:25.01 | monsted | JerJer: err, yeah |
| 21:25.41 | _paulo_ | terrapen, you can have 1 asterisk box acting as a gateway... |
| 21:25.55 | terrapen | are there options other than cisco? |
| 21:26.16 | terrapen | paulo, i'd like something solid state...i wonder what HW i would use if i did run * for this |
| 21:26.39 | terrapen | jerjer how do you guys bring in channels? PRI? |
| 21:26.40 | _paulo_ | terrapen, if you dont use compression when talking to other asterisk box, i think the thing scales well. |
| 21:26.50 | giesen | _Sam--: thanks |
| 21:26.56 | giesen | that may just do the trick |
| 21:27.01 | terrapen | paulo, i forgot to mention....i'm also very concerned about uptime |
| 21:27.07 | terrapen | we're running an inbound call center |
| 21:28.01 | giesen | then buy a tekelec t6000 and be done with it |
| 21:28.03 | giesen | =) |
| 21:28.12 | giesen | erm wait |
| 21:28.18 | giesen | tekelec is a pos |
| 21:28.18 | killfill_ | how should i gdb asterisk?.. |
| 21:28.20 | killfill_ | i get |
| 21:28.22 | _paulo_ | terrapen, I have very long MTBF using * with linux. |
| 21:28.23 | killfill_ | res_crypto.so => (Cryptographic Digital Signatures) |
| 21:28.23 | killfill_ | gdb in realloc(): error: pointer to wrong page |
| 21:28.23 | killfill_ | Abort trap: 6 (core dumped) |
| 21:28.25 | killfill_ | :-S |
| 21:28.29 | _Sam-- | i have connected my asterisk fine to t6000s |
| 21:28.30 | killfill_ | cannot get to the prompt. |
| 21:28.37 | killfill_ | (it works withouth debuggin tho.. :-p |
| 21:28.38 | _Sam-- | it wasnt MY t6000s....but it worked well. |
| 21:28.42 | giesen | _paulo_: I think the idea he wants is failover without dropping calls |
| 21:28.57 | giesen | I got a redundant asterisk setup going |
| 21:29.07 | giesen | but it still causes about 10s of downtime when it fails over |
| 21:29.41 | giesen | was hoping to put ser in front of it but ran out of time |
| 21:30.17 | _paulo_ | giesen: do you have both servers always connected, or are you switching? |
| 21:30.18 | giesen | _Sam--: I know people that put asterisk in front of a t6k just to handle SIP uri dialing |
| 21:30.28 | giesen | _paulo_: one is hot and one is standby |
| 21:30.32 | giesen | so when it fails over |
| 21:30.40 | giesen | the standby becomes active |
| 21:30.47 | giesen | takes about 10 seconds to fail over |
| 21:30.53 | _Sam-- | giesen: the t6000 i connected to -- it could have been replaced by asterisk for what it was doing. |
| 21:30.57 | *** join/#asterisk linlin (i=will@c-71-194-70-13.hsd1.il.comcast.net) |
| 21:31.08 | _Sam-- | i was the first person to ever connect to it with asterisk....my clec was clueless |
| 21:31.09 | giesen | t6k is pretty good at partitioning |
| 21:31.13 | giesen | which asterisk definitely isnt. |
| 21:31.15 | _Sam-- | it sits at the other side of my data T1 |
| 21:32.12 | _paulo_ | giesen: 10s seems very good. |
| 21:32.36 | _Sam-- | what causes the switch to failover? lack of ping repsonse, or what is the criteria? |
| 21:32.42 | _paulo_ | giesen: at least for commodity hardware and software like *. |
| 21:33.09 | giesen | _Sam--: you can specify a bunch of criteria |
| 21:33.17 | giesen | but it does a bunch of ethernet broadcasts |
| 21:33.20 | giesen | on a second nic |
| 21:33.26 | giesen | with a crossover cable between the two boxes |
| 21:33.30 | _Sam-- | the failover comes up with the same IP as the original host? |
| 21:33.32 | giesen | it can also check running processes |
| 21:33.35 | giesen | yes |
| 21:33.39 | giesen | they each have their own IP |
| 21:33.43 | giesen | plus a 3rd shared IP |
| 21:33.51 | giesen | that the active host takes over |
| 21:33.59 | _Sam-- | how do they stay synced? rsync and / or mysql replication? |
| 21:34.07 | giesen | I use drbd |
| 21:34.07 | _paulo_ | giesen, they also share the MAC address? |
| 21:34.10 | giesen | shared disk |
| 21:34.20 | giesen | it's basically a network shared disk |
| 21:34.32 | giesen | that is readonly on the standby |
| 21:34.36 | giesen | and read-write on the master |
| 21:34.40 | _Sam-- | i havent worked with drbd at all yet |
| 21:34.43 | giesen | so as you make changes to configs |
| 21:34.46 | giesen | or voicemails are left |
| 21:34.47 | giesen | etc |
| 21:34.59 | giesen | it's all copied in realtime to the standby |
| 21:35.04 | giesen | when it faisl over |
| 21:35.10 | giesen | the disk becomes read-write on the standby |
| 21:35.17 | giesen | it's actually a very simple setup |
| 21:35.21 | _Sam-- | if you control the failover criteria -- why is it taking 10 seconds to switch? |
| 21:35.22 | giesen | intentionally |
| 21:35.33 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 21:35.33 | giesen | _Sam--: to startup asterisk, etc |
| 21:35.37 | _Sam-- | i keep a hot backup running of my asterisk stuff, but not with an auto failover |
| 21:35.47 | *** join/#asterisk xezz (n=drftr@b-h7.ektheseis.otenet.gr) |
| 21:35.56 | xezz | hello |
| 21:36.10 | giesen | I cant keep asterisk running because it woudlnt have state information anyways |
| 21:36.17 | giesen | or be able to bind to the proper ip |
| 21:36.25 | LeadCrayon | Hello. I've just tried to get the Asterisk manager interface operating via the web, but when I try to access it Asterisk shuts down. Has anyone seen this behavior before and know what's causing it? |
| 21:36.30 | xezz | im running an asterisk server on debian , how can i record my incoming/outgoing calls without any web interfaces etc ? |
| 21:36.34 | _Sam-- | giesen: how many people rely on your asterisk? |
| 21:36.39 | giesen | 10 or so |
| 21:36.46 | giesen | def not a high volume box |
| 21:36.49 | _Sam-- | must be some important calls |
| 21:36.56 | giesen | of course |
| 21:37.04 | giesen | that's why we have redundancy |
| 21:37.08 | _Sam-- | xezz; asterisk addons - cdr_mysql |
| 21:37.20 | giesen | and the 10 second failover time was an acceptable compromise |
| 21:37.25 | _paulo_ | xess: http://voip-info.org/wiki/view/Asterisk+-+documentation+of+application+commands#SoundsRecordingandmonitoringlisteningin |
| 21:37.31 | giesen | in terms of cost and features |
| 21:37.37 | _Sam-- | i could manually switch over in about 10 seconds...but if i wasnt here, it wouldnt happen in 10 seconds. |
| 21:37.38 | giesen | I built the whole system on $400 worth of hardware |
| 21:37.47 | giesen | yeah this is fully automatic |
| 21:37.56 | giesen | and anything you can write a script for |
| 21:37.59 | giesen | can become a failover condition |
| 21:38.18 | _Sam-- | how often does the script check the conditions? |
| 21:38.21 | terrapen | hmmm |
| 21:38.23 | giesen | Im using linux heartbeat for the actual failover |
| 21:38.26 | giesen | it's configurable |
| 21:38.31 | *** join/#asterisk FaithX (n=FaithX@ns.linuxterminal.com) |
| 21:38.37 | terrapen | i have to build an incredibly bomber call center |
| 21:38.45 | terrapen | we use avaya right now and it never breaks |
| 21:38.49 | terrapen | so * has to be the same |
| 21:39.01 | _Sam-- | terrapen: my * never breaks (knocks wood -- been up 25 weeks?) |
| 21:39.01 | xezz | ty |
| 21:39.10 | giesen | asterisk rarely breaks except when you change something |
| 21:39.10 | _paulo_ | avaya costs an leg and an arm... |
| 21:39.25 | giesen | or use agents in queues and try to transfer a call =) |
| 21:39.26 | _Sam-- | phone*CLI> show uptime |
| 21:39.26 | _Sam-- | System uptime: 24 weeks, 16 hours, 39 minutes, 24 seconds |
| 21:39.26 | _Sam-- | Last reload: 2 weeks, 2 days, 19 hours, 21 minutes, 26 seconds |
| 21:39.26 | terrapen | * runs great here already...we run one of our offices on it |
| 21:39.34 | terrapen | <PROTECTED> |
| 21:39.39 | `Sean | show uptime |
| 21:39.40 | `Sean | System uptime: 11 hours, 58 minutes, 12 seconds |
| 21:39.41 | `Sean | *CLI> |
| 21:39.51 | terrapen | System uptime: 8 weeks, 5 days, 17 hours, 51 minutes, 44 seconds |
| 21:39.54 | terrapen | yeah, we're happy |
| 21:39.59 | Grnd-Wire | terrapen: Are you running in some sort of key mode, or hybrid/pbx ? |
| 21:40.17 | giesen | 16:40:01 up 202 days, 19:05, 4 users, load average: 0.00, 0.00, 0.00 |
| 21:40.19 | terrapen | that system right there is a mix of stuff |
| 21:40.22 | *** join/#asterisk trelane_ (i=trelane@unaffiliated/trelane) |
| 21:40.24 | _Sam-- | my upstream telephony providers are way less reliable than my actual * |
| 21:40.26 | giesen | sadly the last time that system went down |
| 21:40.30 | terrapen | it has a bunch of polycom ip501s connected to it |
| 21:40.34 | giesen | was to install an ATM card |
| 21:40.35 | terrapen | but also a crappy NEC PBX |
| 21:40.48 | terrapen | but the NEC must traverse the * to call out |
| 21:40.59 | terrapen | we use a redfone 4-port fonebridge |
| 21:41.00 | trelane_ | I'm getting invalid module formats after building zaptel 1.4.0 against linux 2.6.16-gentoo3. Has anyone seen this issue as yet? |
| 21:41.02 | terrapen | which seems to work well |
| 21:41.27 | *** join/#asterisk sbingner (n=sam@pdpc/supporter/sustaining/sbingner) |
| 21:41.41 | terrapen | for this new system, I think we'll be 100% polycom |
| 21:41.59 | terrapen | with some SER proxies load balanced behind OpenBSD and hostated |
| 21:42.20 | _Sam-- | terrapen: build it solid state using some kind of flash memory / card to boot from, and mount some crazy raided drive for your system drive ....your * will never break |
| 21:42.33 | _Sam-- | mount a network netapp or something |
| 21:42.39 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 21:43.01 | _Sam-- | i built a few solid state versions, but none that mounted good raided network storage |
| 21:43.39 | terrapen | i've yet to find a high quality 2 proc server that can easily accept a CF drive |
| 21:43.49 | terrapen | we usually buy Sun Fire servers |
| 21:44.02 | _Sam-- | be a waste of money (only in MY own opinion) for a * server |
| 21:44.06 | terrapen | but the CF adapters won't fit in them...at least some of them |
| 21:44.20 | _Sam-- | i see what you're saying. |
| 21:44.22 | terrapen | this server will take a T3 in |
| 21:44.29 | terrapen | so it needs to be hella bomber |
| 21:44.37 | terrapen | ideally, i'd like a failover of the T3 |
| 21:44.52 | _paulo_ | trelane_ : didi you manually rm all files in your modules folder? |
| 21:44.55 | danp | i'm trying to set up an extension in AEL that rolls through all my zap channel and turns on call forwarding (basically *72 and then a number)...any tips on how to do that? |
| 21:44.55 | terrapen | what's the breaking point for T3 versus PRI anyway? |
| 21:45.19 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 21:45.24 | danp | i tried somethning like Dial(Zap/1/*72...); Wait(5); Hangup(); ... but Hangup hangs up my SIP call |
| 21:45.26 | terrapen | maybe we *would* do better with a bunch of PRIs and some of those four-port PRI <-> SIP gateways |
| 21:45.42 | trelane_ | _paulo_: no but I specifically went back and rm'd those modules |
| 21:45.48 | terrapen | rather than bet the whole kitty on one asterisk box with a T1 card |
| 21:47.32 | *** join/#asterisk pgp4privacy (n=Administ@216.201.219.105) |
| 21:47.58 | trelane_ | _paulo_: I went back and nuked the whole directory, to no avail |
| 21:48.16 | _paulo_ | trelane_ : I use Debian. |
| 21:48.29 | pgp4privacy | ok, what I need to do is create a feature code that when a user enters it on their phone..the system changes inbound routing from a call group, to an after hours ivr |
| 21:48.37 | pgp4privacy | any help, or suggestions? |
| 21:48.47 | trelane_ | _paulo_: the module format shouldn't be any idfferent |
| 21:50.19 | _paulo_ | pgp4privacy: have you seen applicationmap ? http://voip-info.org/wiki/index.php?page=Asterisk+config+features.conf |
| 21:51.18 | pgp4privacy | _paulo: I just googled that earlier, however, I am unsure of what to put in to change an inbound route |
| 21:51.31 | pgp4privacy | would I need to create a batch file that interfaces with the management interface? |
| 21:51.54 | pgp4privacy | I have written a similar bourne script that checked the time, and routed appropriately..but I want to let the user set it |
| 21:51.59 | pgp4privacy | via a feature code |
| 21:52.18 | _Sam-- | you wouldnt need a bash script....that is what GOTOIFTIME does |
| 21:52.32 | _paulo_ | pgp4privacy: you can execute any command in asterisk from shell with something like "asterisk -r -x 'show channels'" |
| 21:52.55 | _Sam-- | asterisk -rx will save two keystrokes |
| 21:53.07 | pgp4privacy | _paulo: hmm, I see what you are saying...looks like I just need to research a bit more before bothering you all |
| 21:53.23 | wunderkin | pgp4privacy, i'm not sure where these people are coming from... but you want to look at getting and setting a variable |
| 21:53.45 | _Sam-- | i was only commenting about using a bash script to change items based on time of day |
| 21:53.52 | _Sam-- | that is what GotoIfTime does |
| 21:53.56 | pgp4privacy | wunderkin: use the variable to store what exactly? |
| 21:53.56 | wunderkin | that's not what he asked for |
| 21:53.57 | _Sam-- | in asterisk |
| 21:54.09 | _paulo_ | pgp4privacy: exten => s,1,GotoIfTime(9:00-18:00|mon-fri|*|*?businesshours,s,1) |
| 21:54.36 | wunderkin | if true, use this, if false, go elsewhere |
| 21:54.47 | pgp4privacy | my problem is, that the hours are different every day |
| 21:55.06 | pgp4privacy | which is why I liked the flexibilty of a feature code to set and unset the after hours IVR for inbound routing |
| 21:55.21 | pgp4privacy | previously, I did the bash script that used gotoiftime, to set different time schedules for each day |
| 21:55.35 | pgp4privacy | I suppose I could go back to that, but they are wanting something they can interface with |
| 21:56.23 | _Sam-- | from what i can tell, you should be able to do it with AGI, but i dont have steps HOW |
| 21:56.41 | _paulo_ | pgp4privacy: use applicationmap from features.conf with some macro to change the route. |
| 21:56.51 | wunderkin | yuck yuck yuck, all you need is to set a variable and check for the existance of the variable |
| 21:57.17 | pgp4privacy | wunderkin: you are saying set some type of state variable used to tell if it is biz hours or not |
| 21:57.44 | _Sam-- | so, when you dial an extension, it set $open=n |
| 21:57.44 | _paulo_ | bye people! |
| 21:57.44 | _Sam-- | and then if $open=n , then play IVR |
| 21:57.45 | pgp4privacy | _paulo: thank you for yoru help!! |
| 21:58.08 | _Sam-- | and you could dial another extension and set $open=y |
| 21:58.36 | pgp4privacy | I think I see the logic you are using, I just need to get to the command line and bang out some tests |
| 21:58.57 | _Sam-- | exten => 1,Set(open=1) |
| 21:59.12 | _Sam-- | http://www.voip-info.org/wiki/view/Asterisk+cmd+Set |
| 21:59.42 | pgp4privacy | but, just to throw this out there: if I wished to due a feature code...I coudl use featuremap which executed a macro I build...which in turn uses the management interface to change routing, yes? |
| 21:59.50 | pgp4privacy | use, not due |
| 22:00.25 | syzygyBSD | what do you mean by "feature code" |
| 22:00.47 | pgp4privacy | like "*889" |
| 22:00.51 | pgp4privacy | that's all I mean, really |
| 22:01.01 | pgp4privacy | just a predefined key sequence to set and unset the after hours IVR routing |
| 22:01.47 | _Sam-- | that is what i was describing |
| 22:01.48 | syzygyBSD | well, I know it is possible, I just don't know about using featuremap, I have always done it other ways |
| 22:01.53 | _Sam-- | just set an extension to dial |
| 22:02.16 | _Sam-- | exten => 999,1,Set(open=1) |
| 22:02.23 | _Sam-- | exten => 000,1,Set(open=0) |
| 22:02.27 | pgp4privacy | _Sam: ahh, I see...sorry I misunderstood |
| 22:04.01 | pgp4privacy | thank you all very much for your help, I really appreciate it! |
| 22:08.12 | *** part/#asterisk markit (n=konversa@host119-245-static.72-81-b.business.telecomitalia.it) |
| 22:08.16 | codefreeze | danp: any revelations as to how to do what you want? |
| 22:08.53 | *** part/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 22:10.25 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 22:10.33 | *** join/#asterisk tomcruize (n=elgroper@wireless-net143-ip170-toronto.ica.net) |
| 22:10.41 | tomcruize | hi everyone |
| 22:10.53 | tomcruize | has anyone used spandsp ? |
| 22:10.58 | tomcruize | with asterisk? |
| 22:11.00 | syzygyBSD | yes |
| 22:11.23 | tomcruize | can you resize the image coming in to 45.2% of it's original size? |
| 22:11.34 | *** join/#asterisk Teeli (n=tili@172.Red-88-0-147.dynamicIP.rima-tde.net) |
| 22:12.02 | syzygyBSD | um.. I don't think using spandsp, I used another program to do all my modifications after I saved the image |
| 22:14.16 | syzygyBSD | why 45.2% |
| 22:15.07 | tomcruize | can't you just pass the parameters as to the image size. |
| 22:16.05 | *** join/#asterisk Dr-Linux|home (n=Dreamer@DSL-202-59-73-131.nexlinx.net.pk) |
| 22:16.09 | Dr-Linux|home | anybody tried LumenVox recognition? |
| 22:18.40 | *** join/#asterisk ivanfm (n=ivanfm@c93481ec.virtua.com.br) |
| 22:19.26 | *** join/#asterisk DocHolliday (i=RogerRab@gateway/gpg-tor/key-0x0E4F6D6C) |
| 22:20.19 | tomcruize | 45.2 is the exact size of the page being viewed |
| 22:20.28 | Dr-Linux|home | i purchased lumenvox starter package, everything is fine, but i can't load res_speech_lumenvox.so module |
| 22:20.35 | Dr-Linux|home | not sure what's the reason |
| 22:20.40 | tomcruize | when sending it out through a third party, only 1/2 shows up. |
| 22:20.44 | file | have you contacted technical support? |
| 22:20.47 | DonX | Per Goto's documentation, it's supposed to failover to the next priority if the context/extension/priority cannot be reached. Anyone know why this is? |
| 22:20.58 | DonX | I meant why this isn't working |
| 22:21.07 | Dr-Linux|home | file: me? |
| 22:21.12 | DonX | sent into invalid extension '2356' in context 'site4', but no invalid handler |
| 22:21.13 | file | Dr-Linux|home: yes |
| 22:21.53 | Dr-Linux|home | file: i'm using opensource asterisk, not sure if digium support will help me with that |
| 22:21.55 | Dr-Linux|home | :S |
| 22:22.13 | file | they will help you get the connector installed |
| 22:22.16 | Dr-Linux|home | i tried my best to fix this ONLY issue but failed, so came here to ask |
| 22:22.25 | *** part/#asterisk rbrindley (n=ryan@user-69-73-26-211.knology.net) |
| 22:22.54 | Dr-Linux|home | file: that support is for free? |
| 22:23.02 | file | it's installation support |
| 22:23.11 | Dr-Linux|home | file: correct |
| 22:23.32 | file | (that means yes) |
| 22:23.58 | Dr-Linux|home | then i'll call them tomorrow |
| 22:24.14 | Dr-Linux|home | also i coudn't find anything on google |
| 22:24.30 | Dr-Linux|home | once i'm done with this, i'll write a blog on it |
| 22:24.35 | *** join/#asterisk malcolmd (i=malcolmd@pdpc/sponsor/digium/malcolmd) |
| 22:25.46 | malcolmd | aloha :) |
| 22:28.29 | *** join/#asterisk grandy (n=chatzill@c-71-198-130-108.hsd1.ca.comcast.net) |
| 22:29.58 | grandy | Hello. I ordered some g729 codec licenses from Digium... and I'm wondering if anyone knows how they are delivered? I haven't received an email with any registration codes yet... |
| 22:30.42 | Qwell[] | grandy: via email |
| 22:30.58 | grandy | Qwell[]: ok... so they will eventually email me the info to get the licenses set up? |
| 22:30.59 | Qwell[] | if it's been a while (ie; something that wouldn't be considered "reasonable"), call support and ask them |
| 22:31.01 | DocHolliday | people actually buy those things @_@ |
| 22:31.17 | grandy | DocHolliday: yeah i'm trying to be supportive of Digium's profit model |
| 22:31.37 | Qwell[] | and legal ;) |
| 22:31.49 | grandy | that too, |
| 22:31.49 | DocHolliday | grandy, i'm sure its well appreciated. |
| 22:32.27 | DocHolliday | Qwell, is the call quality substantially better G711 -> 29? |
| 22:32.43 | Qwell[] | no, it's worse, but that isn't the reason you'd use g729 |
| 22:32.43 | Juggie | obviously it is not. |
| 22:33.04 | DocHolliday | heh |
| 22:33.23 | *** join/#asterisk uwe (n=uwe@a40-45.adsl.paltel.net) |
| 22:33.49 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 22:34.38 | tomcruize | spandsp default image size LxW where can that be set anyone know? |
| 22:37.51 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 22:39.15 | Dr-Linux|home | Qwell[]: since Sergio is dead, can i try my conference phone 1.4 ? |
| 22:39.25 | Dr-Linux|home | suggestions? |
| 22:39.54 | Qwell[] | Dr-Linux|home: I need packet captures from ccm for it, before it can be supported in chan_skinny |
| 22:40.23 | Dr-Linux|home | ccm? :S |
| 22:40.38 | Qwell[] | ~ccm |
| 22:40.39 | jbot | methinks ccm is Cisco Call Manager |
| 22:40.40 | Dr-Linux|home | Qwell[]: what logs i can send you to look into? |
| 22:40.50 | Dr-Linux|home | aww |
| 22:40.51 | bkruse | Qwell > all |
| 22:40.54 | bkruse | Qwell > jbot |
| 22:40.55 | bkruse | ~qwell |
| 22:40.56 | jbot | [qwell] a patented liquid formula that contains three plant-based bio-active agents that work together in a perfectly balanced combination. These agents act synergistically to boost your good cholesterol and slash the bad. |
| 22:41.03 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 22:43.52 | *** join/#asterisk X-Rob (n=Rob@dsl-202-173-151-24.qld.westnet.com.au) |
| 22:46.39 | *** join/#asterisk jarg (n=jarg@200.56.225.61) |
| 22:50.09 | monsted | Console> (enable) sh port voice active call |
| 22:50.13 | monsted | Total: 0 call |
| 22:50.15 | monsted | heh |
| 22:50.19 | *** part/#asterisk pgp4privacy (n=Administ@216.201.219.105) |
| 22:50.28 | monsted | good thing i have 2500 DS0 channels |
| 22:50.43 | monsted | wouldn't want it to overload! |
| 22:50.45 | Katty | hi! |
| 22:52.04 | danp | if i use Dial with g option (execute more commands when the called party hangs up), can i make more calls? |
| 22:52.58 | danp | i'm trying to find a way to use Dial multiple times to roll through my zap lines and set up telco call forwarding |
| 22:53.38 | Katty | what happens if you setup a time out? |
| 22:53.49 | Katty | and then go to zap2 or whatever |
| 22:56.14 | danp | won't that disconnect my call as well? |
| 22:56.20 | *** join/#asterisk nvzn (n=nvzn@dormir.dreaming.org) |
| 22:56.38 | nvzn | anyone know where to get the best deal on an ATA in canada? |
| 23:00.26 | codefreeze | danp: what if you use the 'h' extension to dial? You can increment the number and dial again...? |
| 23:03.07 | *** join/#asterisk obnauticus (n=aao_pwne@c-24-21-116-29.hsd1.wa.comcast.net) |
| 23:06.38 | *** join/#asterisk lullabud (n=lullabud@12.24.42.67) |
| 23:10.04 | danp | codefreeze: i'll give that a shot, thanks! i got it working for one extension so now i'm messing around with automating all of them |
| 23:12.21 | *** join/#asterisk existance (n=existanc@cpe-24-242-205-133.houston.res.rr.com) |
| 23:12.25 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 23:12.40 | *** join/#asterisk gopher1 (i=gopher@sickos.org) |
| 23:12.43 | gopher1 | hello |
| 23:12.59 | gopher1 | i've problems with debian sarge and a digium te110p, revision d |
| 23:13.18 | gopher1 | the d channel flappers; is this a known bug? |
| 23:13.38 | existance | How can I connect to asterisk through the Manager API using PHP? I've gotten as far as using fsockopen() to open a connection with the server, but I can't figure out what to send through the connection. |
| 23:13.46 | gopher1 | a rev-b card works in the same box, and rev-d is from 2006-10-10 which is newer than debian sarge |
| 23:16.00 | *** join/#asterisk SomeOne1 (n=SomeOne1@pool-71-126-150-34.washdc.fios.verizon.net) |
| 23:16.05 | SomeOne1 | cant you make it jump priorities? |
| 23:16.17 | SomeOne1 | like if its missing a priority in the middle |
| 23:16.19 | SomeOne1 | it gets stucks |
| 23:16.22 | SomeOne1 | stuck* |
| 23:16.24 | SomeOne1 | and times out |
| 23:16.37 | Qwell[] | SomeOne1: use n, and you won't have that problem |
| 23:16.49 | SomeOne1 | theres no other way? |
| 23:16.54 | SomeOne1 | and in what way can i use n? |
| 23:16.59 | Qwell[] | Don't skip priorities? |
| 23:17.07 | Qwell[] | s,1,Foo() |
| 23:17.10 | Qwell[] | s,n,Bar() |
| 23:17.11 | Qwell[] | s,n,Baz() |
| 23:17.40 | codefreeze | danp: if that fails, try the call q and a script to submit the calls...? |
| 23:18.42 | SomeOne1 | so n basically makes it just like go in the order its listed? |
| 23:22.14 | *** join/#asterisk Flauto (n=zhao@adsl-75-3-147-195.dsl.chcgil.sbcglobal.net) |
| 23:24.02 | SomeOne1 | so n basically makes it just like go in the order its listed? |
| 23:28.10 | *** join/#asterisk arguile (i=user224@KTNRON06-1242488957.sdsl.bell.ca) |
| 23:28.34 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 23:29.12 | *** part/#asterisk hoobastooba (n=ckwall@63.149.122.93) |
| 23:29.13 | SomeOne1 | so n basically makes it just like go in the order its listed? |
| 23:31.54 | grandy | Hello... can anyone recommend an easy/friendly web interface for webifying asterisk voicemail? |
| 23:32.26 | nortex | grandy, Check out ARI |
| 23:32.40 | grandy | nortex: ahh thanks |
| 23:33.43 | *** join/#asterisk Hmmhesays (n=Neg@24-117-135-28.cpe.cableone.net) |
| 23:35.19 | Flauto | is there anyone using voicestick? |
| 23:35.41 | Flauto | i have a problem using it with my asterisk |
| 23:35.46 | existance | If I send an originate action to the asterisk server which originates on an extension that is currently in use, what happens? |
| 23:35.49 | Flauto | i mean this problem is new |
| 23:36.46 | grandy | hello... I want to have asterisk try a user's cell phone before going to voicemail... is there a way to have it hang up if it gets the cell phone's own voicemail? |
| 23:37.08 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-140-124.ks.ok.cox.net) |
| 23:37.09 | Hmmhesays | require user input to accept the call |
| 23:37.48 | tset | setup a maximum time before asterisk hangs up and moves onto the next increment in your context |
| 23:38.01 | tset | mine is 15/20 seconds |
| 23:38.32 | tset | some phones allow up to 30 seconds before voicemail hits, its user configurable |
| 23:38.44 | *** join/#asterisk elriah (i=elriah@adsl-072-149-159-016.sip.bhm.bellsouth.net) |
| 23:39.18 | elriah | Hi all. Anyone know how to limit calls to a asterisk context? I know how to limit them with sip peers.. Any help would be appreciated.. |
| 23:39.21 | grandy | tset: ahh so modify the cell phone's setting to make it ring like 5 times before it goes to vm |
| 23:39.52 | tset | grandy if your cell provider allows it |
| 23:40.14 | grandy | tset: hmmm... not sure... |
| 23:40.45 | tset | and in your dial out command on the asterisk server you add a timeout value of say 15 seconds (so if no answer is in 15 seconds it will go to the next command it needs to run) |
| 23:40.59 | Dr-Linux|home | any single person ever tried LumenVox? :) |
| 23:42.24 | nortex | Yup |
| 23:42.30 | tset | grandy: example: exten => s,1, Dial(SIP/number@provider,timeoutvaluegoeshere,options) |
| 23:42.37 | nortex | I have a starterpak |
| 23:42.42 | grandy | tset: ahh... ok... i'll try it |
| 23:42.48 | elriah | Or maybe the question should be -- is there any way to limit the number of outbound calls in asterisk? |
| 23:42.57 | existance | I am implementing click-to-dial on a remote machine using the Asterisk Manager API. Would it be appropriate to keep a login to the asterisk server up persistently, or to open and close a connection as calls need? |
| 23:43.14 | tset | grandy should work for ya, i do the same deal here, i use a ringdown (or 'follow me') service to track me down :) |
| 23:43.25 | grandy | tset: yeah that's what i'm trying to do... |
| 23:43.35 | *** join/#asterisk sethwhit (n=SethWhit@70-56-234-164.clsp.qwest.net) |
| 23:44.02 | tset | grandy ensure when testing you login to asterisk console with high verbose so you can watch what it does |
| 23:44.08 | grandy | ahh ok |
| 23:53.10 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 23:54.25 | *** join/#asterisk izaak (n=izaak@modemcable097.151-202-24.mc.videotron.ca) |
| 23:54.45 | *** join/#asterisk lowlevel (n=Stuart@CPE000e0c057fad-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 23:58.42 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-153-99-104.red.bezeqint.net) |
| 23:58.58 | *** join/#asterisk oQPa (n=uawename@242.Red-83-35-130.dynamicIP.rima-tde.net) |