| 00:00.29 | *** join/#asterisk sb_mx (n=sb_mx@dsl-200-67-149-101.prod-empresarial.com.mx) |
| 00:01.22 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 00:03.51 | tasat- | Strom_C: are you aware of, or have you used any SIP origination providers that provide the DNIS info in SIP headers? |
| 00:04.05 | Strom_C | tasat-: plenty |
| 00:04.22 | Strom_C | they tell you which DID the call is coming in on |
| 00:06.36 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 00:06.41 | shmaltz | tzfrir ping |
| 00:06.50 | shmaltz | tzafrir_laptop |
| 00:07.14 | shmaltz | ~seen tzafrir_laptop |
| 00:07.19 | jbot | tzafrir_laptop is currently on #asterisk (4h 3m 47s), last said: 'you have Sangoma, not ZapBRI, right?'. |
| 00:07.20 | shmaltz | ~seen tzafrir |
| 00:07.22 | jbot | tzafrir is currently on #asterisk. Has said a total of 50 messages. Is idling for 8h 11m 53s, last said: 'I can't really say. But it should surely be able to manage with a quad PRI'. |
| 00:07.22 | tasat- | Strom_C: humm... scanning the headers and I see the originating caller's number, and the DID, but not the number to which the call was originally directed to and forwarded from... |
| 00:08.02 | Strom_C | tasat-: that's contained in the SS7 ISUP Forwarding Number field, and all bets are off as to whether your ITSP will pass that through in any form |
| 00:08.08 | *** join/#asterisk techie (n=techie@adsl-068-209-242-072.sip.mia.bellsouth.net) |
| 00:08.13 | *** join/#asterisk kaphein (n=kaphein@ottawa-hs-209-217-75-187.d-ip.magma.ca) |
| 00:08.28 | *** part/#asterisk techie (n=techie@adsl-068-209-242-072.sip.mia.bellsouth.net) |
| 00:08.43 | *** join/#asterisk opus_ (i=opus@pabstblueribbon.net) |
| 00:08.47 | opus_ | anyone recommend a cheap channel bank? |
| 00:09.03 | Strom_C | get a used Adtran :) |
| 00:09.09 | Strom_C | cheap AND high-quality! |
| 00:09.12 | shmaltz | opus_, for what purpose? |
| 00:09.22 | shmaltz | or a used Adit 600 |
| 00:09.29 | opus_ | modems |
| 00:10.22 | opus_ | how much is a used adtran? |
| 00:10.33 | Strom_C | opus_: ive seen them on ebay for $500 |
| 00:10.33 | *** join/#asterisk techie (n=techie@adsl-068-209-242-072.sip.mia.bellsouth.net) |
| 00:10.49 | harryvv | thats cheap! |
| 00:10.53 | kaphein | Hi there, newbie to asterisk.... would it be safe to open my asterisk server to the internet but only for IAX ? I googled it but can't find any pros/cons... any pointers |
| 00:11.07 | Strom_C | kaphein: it's fairly secure |
| 00:11.07 | QbY | If anyone can give me some insght to the error and subsquent crash, a section of my log is here: http://pastebin.ca/155328 |
| 00:12.01 | harryvv | Anyone know if there a bulk call genertors in software form vs hardware? I just came across a site that sells isdn/pri/pstn call generators and thought that was interesting. |
| 00:12.48 | QbY | kaphein.. i've been running my asterisk on the public internet for over a year, and never (knocking on wood) had a security incident |
| 00:13.42 | tasat- | Strom_C: OK, so as long as I'm working through a SIP/IAX provider, there's no practical way to detect the forwarding number? |
| 00:13.49 | harryvv | QBy two years here and no issues with security. |
| 00:13.59 | Strom_C | tasat-: it depends. You'll have to ask the provider |
| 00:14.14 | harryvv | http://www.vconsole.com/client/ |
| 00:14.25 | harryvv | interesting telephony products |
| 00:14.46 | JT | opus_: a channel bank that's actually cheap is a zhone zplex 10 |
| 00:14.48 | kaphein | thanks :) |
| 00:14.51 | JT | but they have CID issues |
| 00:16.03 | *** part/#asterisk shwa (n=FreePBX2@62.235.228.237) |
| 00:20.41 | Strom_C | I've never had issues with Adtran stuff ever ever :) |
| 00:22.25 | tessier__ | djpbx*CLI> Aug 30 17:22:26 WARNING[18953]: chan_agent.c:1836 __login_exec: Extension '127' is not valid for automatic login of agent '1000' |
| 00:22.30 | tessier__ | Anyone know how queues work? |
| 00:22.49 | tessier__ | I've read the howto on setting this up but it doesn't make clean this bit or the stuff about "members" in queues.conf |
| 00:26.11 | Nugget | http://creativebastard.com/archives/2006/08/168/ <-- fun idea for your asterisk installation. |
| 00:26.58 | opus_ | Nugget, is that the telemarketing mp3 ? |
| 00:27.01 | Nugget | yeah |
| 00:27.09 | opus_ | great idea |
| 00:27.12 | opus_ | i just scream at them |
| 00:27.46 | opus_ | and play come to daddy |
| 00:27.56 | opus_ | your SOUL :) |
| 00:29.51 | *** part/#asterisk brif8 (n=Administ@ns1.ttienterprises.org) |
| 00:30.08 | *** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 00:33.44 | *** join/#asterisk rikstah (n=rick@c-71-227-234-92.hsd1.or.comcast.net) |
| 00:35.41 | *** join/#asterisk freeepbxxnoobb (n=chatzill@rrcs-67-52-187-18.west.biz.rr.com) |
| 00:37.40 | Strom_C | well that certainly goes on the list of "most annoying handles" :) |
| 00:37.40 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 00:38.28 | *** join/#asterisk Terlouw (n=d3vious@80.126.223.172) |
| 00:39.22 | Terlouw | people... quick question... why when i activate callrecording, do i only hear 1 side of the confersation ? ... what could that be? I am not behind a NAT, and calling works perfectly, only in the recording, i only hear one side... :S |
| 00:39.24 | harryvv | strom or anyone for that fact, if a user is registered against a provider and thay have the context setup in sip and iax.conf even withouth a dial plan will the asterisk box try to process the call knowning it will fail? |
| 00:39.44 | harryvv | nat issue and check your globals |
| 00:40.22 | Strom_C | i don't fully understand the question |
| 00:44.21 | harryvv | Strom my settings were fine and i have made no changes. The context was defalt for the inbound DID calls and went strait to my ivr. I dont think its a dial plan issue but cli should show something! I call the 1800 number I own and the providers servers gave a playback of "Were sorry your call did not go though please check your number" My firewall shows no packet info from the tisp yet tcpdump last night showed something. |
| 00:45.28 | Nugget | Terlouw: that's normal if you don't have soxmix |
| 00:45.51 | Nugget | Monitor() initially records each direction into separate files and then (if you've told it to) merges them after the call |
| 00:46.17 | Nugget | and if you lack soxmix or it's not in asterisk's path it will silently fail |
| 00:52.56 | [shodan] | hey , those sipurai dual fxs ata , they are actually two fxs in one right ? not one fxs with two connectors ? |
| 00:53.26 | Strom_C | [shodan]: the spa-2002? |
| 00:53.40 | Strom_C | the spa-2002 is two individual fxs ports |
| 00:53.45 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 00:53.45 | *** mode/#asterisk [+o russellb] by ChanServ |
| 00:56.02 | freeepbxxnoobb | anyone know anything about partner avaya phone system |
| 00:56.05 | freeepbxxnoobb | when I get a incoming call then it rings on two lines and the second line wont stop ringing |
| 00:56.10 | freeepbxxnoobb | does anyone know if its the automated attendant or the call hunt or dxd???? |
| 00:56.29 | Strom_C | um, what the hell? is this #avaya all of a sudden? |
| 00:57.26 | Strom_C | :) |
| 00:57.28 | Terlouw | ok... how do i get this.. soxmix ? |
| 00:57.46 | freeepbxxnoobb | its connected to my asterisk box and im not sure if its because of that |
| 00:58.00 | [shodan] | ok great , for a second there I thought I had just made a big mistake ;) |
| 00:58.19 | Strom_C | freeepbxxnoobb: how is it connected? |
| 00:58.49 | freeepbxxnoobb | thru a pap2 adapter |
| 00:58.56 | Strom_C | ..... |
| 00:59.00 | Strom_C | what?! |
| 00:59.07 | Strom_C | are you totally mad? |
| 00:59.16 | Strom_C | asterisk -> analog -> avaya? |
| 00:59.39 | freeepbxxnoobb | I know but thats what the customer wants |
| 00:59.45 | Strom_C | ..... |
| 00:59.54 | freeepbxxnoobb | he wants voip with his legacy pbx |
| 01:00.00 | *** join/#asterisk pigpen2 (n=mark@fw.seamans.cc) |
| 01:00.05 | [shodan] | you should know by now that "the customer is always wrong" ;) |
| 01:00.16 | Strom_C | as a consultant, it's your job to tell the customer when they are making a huge mstake, and i would be shocked if there isn't a better way to do that than with a PAP2 |
| 01:01.19 | freeepbxxnoobb | he doesnt wanna spend more money and doesnt wanna get rid of old equipment |
| 01:01.23 | harryvv | free thats going to be difficult with his old pbx |
| 01:01.28 | harryvv | its possible |
| 01:01.28 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 01:01.29 | *** mode/#asterisk [+o russellb] by ChanServ |
| 01:01.45 | Strom_C | freeepbxxnoobb: and i bet he's giving you the shit end of the stick with regard to consulting fees |
| 01:02.06 | freeepbxxnoobb | ur on the money with that |
| 01:02.10 | Strom_C | i get requests like that all the time, and I say "no" |
| 01:02.32 | Strom_C | or "come back to me when you're ready to do the job right" |
| 01:02.34 | freeepbxxnoobb | i already agreed so i know not to in the future |
| 01:02.54 | Strom_C | anyway, analog line signaling is pretty boring |
| 01:03.09 | Strom_C | ring, voltage, caller ID bursts, etc |
| 01:03.29 | russellb | s/bor/annoy/ |
| 01:03.39 | Strom_C | so if you're behaving exactly as the telephone company does, then it's not your problem; it's the avaya's problem |
| 01:03.43 | Strom_C | russellb: hahaha |
| 01:03.46 | Strom_C | that too |
| 01:04.06 | freeepbxxnoobb | I know plus its all proprietory crap that u cant really see any thing thats going on |
| 01:05.38 | harryvv | Anyone know of a robust 1800 did company that has aviable staff? |
| 01:05.50 | Strom_C | freeepbxxnoobb: of course, you've reserved the right to terminate the consulting agreement at any time and for any reason, right? |
| 01:06.51 | freeepbxxnoobb | yea but i wanna do this till the end . Ill think of it as a learning experience |
| 01:07.25 | Strom_C | fair enough |
| 01:08.21 | freeepbxxnoobb | but i know nothing of legacy systems and dont know where to look for answers |
| 01:09.01 | Strom_C | freeepbxxnoobb: documentation is not going to be easy to come by. Set everything up on your end to work correctly, and then refer the client to an Avaya specialist |
| 01:09.14 | *** join/#asterisk flujan (n=flujan@201-42-43-127.dsl.telesp.net.br) |
| 01:09.55 | freeepbxxnoobb | I got most problems down by now and wanna see it thru. Then ill remember not to take any more jobs like that |
| 01:11.17 | Strom_C | well, you could always work with the avaya specialist |
| 01:11.33 | Strom_C | subcontract it out or something and add a percentage of their consulting fee to your invoice |
| 01:12.42 | rene- | freeepbxxnoobb: have the avaya guy send his own bill, tell your costumer this is going to be his last avaya bill, his way out |
| 01:13.18 | freeepbxxnoobb | I think ill do that thanks |
| 01:13.33 | Strom_C | half the fun of consulting is creative invoicing :D |
| 01:13.58 | rene- | freeepbxxnoobb: also take into account that if you go tru the pap2 way things like faxing and modem will almost certainly wont work with * |
| 01:14.14 | freeepbxxnoobb | creative invoicing ....i like that |
| 01:14.15 | *** part/#asterisk justinu|laptop (n=Justin@12.44.122.130) |
| 01:14.32 | Strom_C | yeah...honestly if you're interconnecting asterisk with something else, the best method is a TDM(2)400P |
| 01:15.14 | Strom_C | or even better, some variety of T1 card if the equipment can do T1 |
| 01:15.35 | freeepbxxnoobb | cant do t1 |
| 01:15.43 | freeepbxxnoobb | only x100p\ |
| 01:15.47 | Strom_C | hence the "if" |
| 01:15.54 | Strom_C | wait wait wait |
| 01:16.08 | Strom_C | x100P? you're running asterisk as a /station/ off the avaya system? |
| 01:16.28 | freeepbxxnoobb | no its only for timing purposes |
| 01:16.35 | Strom_C | oh, ok |
| 01:16.40 | pigpen2 | Strom_C, hey, swb says they are seeing "slips" on our equipemnt (ie: we loose timing with the line). |
| 01:16.56 | Strom_C | oh interesting |
| 01:16.58 | pigpen2 | They did find that the db was -19db....and fixed it... |
| 01:17.05 | Strom_C | cool |
| 01:17.13 | Strom_C | that may be half the problem |
| 01:17.13 | pigpen2 | but..are there any settings for this "slip" issue? |
| 01:17.18 | pigpen2 | yeah.... |
| 01:17.27 | Strom_C | what is the span configuration in zaptel.conf? |
| 01:17.51 | pigpen2 | zapat? |
| 01:17.55 | pigpen2 | zapata? |
| 01:17.59 | Strom_C | no, zaptel |
| 01:18.12 | Strom_C | span => 1,1,0,esf,b8zs or somethign similar |
| 01:18.15 | Doctor | whats the timing do? |
| 01:18.24 | Doctor | I'm still not clear at all on timing :( |
| 01:18.33 | harryvv | Are there any inexpensive pri generators on the market? |
| 01:18.37 | pigpen2 | sorry..head up the ass... |
| 01:18.45 | pigpen2 | span=1,1,0,esf,b8zs |
| 01:18.50 | Strom_C | harryvv: an asterisk box with a TE110P :) |
| 01:18.51 | harryvv | simulate a telco in a room |
| 01:18.58 | pigpen2 | He is removing the pri from the smartjack....and connecting it to his test set. |
| 01:19.06 | pigpen2 | ... to see if it is still happening... |
| 01:19.09 | Strom_C | pigpen2: ok, so your asterisk box is set to receive timing from the telco |
| 01:19.15 | Strom_C | yeah, see if its the T1 itself |
| 01:19.16 | pigpen2 | correct. |
| 01:19.31 | harryvv | Strom so a asterisk to pri to aster would simulate a telco then. |
| 01:19.36 | harryvv | telco in between |
| 01:19.38 | Strom_C | harryvv: indeed |
| 01:19.44 | pigpen2 | but are there any settings for this? I know of none. |
| 01:19.57 | pigpen2 | other than the clock, they provide...we watch them. |
| 01:19.57 | Strom_C | pigpen2: no, zaptel should get the timing automatically |
| 01:20.01 | pigpen2 | k |
| 01:20.08 | harryvv | What about bulk call generation? |
| 01:20.15 | Strom_C | if you're getting frame slips, that's a problem with the T1 most likely |
| 01:20.27 | pigpen2 | k. |
| 01:20.37 | Strom_C | harryvv: predictive dialers and whatnot are the scum of the earth and should all burn in Hell |
| 01:20.44 | harryvv | heheh |
| 01:22.02 | harryvv | at 500 dollars a card on ebay |
| 01:22.48 | pigpen2 | Strom_C, we are getting slips every 3 or 4 seconds. |
| 01:22.58 | pigpen2 | it was worse with his equipment directly connected. |
| 01:23.18 | Strom_C | pigpen2: it sounds like SWBT didn't do their job correctly in provisioning the T1 line |
| 01:23.28 | *** part/#asterisk sb_mx (n=sb_mx@dsl-200-67-149-101.prod-empresarial.com.mx) |
| 01:25.14 | *** join/#asterisk EyeCue (n=eyecue@unaffiliated/eyecue) |
| 01:25.22 | rene- | pigpen2: how are you measuring for those slips? |
| 01:25.52 | pigpen2 | please elaborate. |
| 01:26.14 | rene- | pigpen2: you said that you were getting slipes each 3/4 secs |
| 01:26.20 | pigpen2 | k |
| 01:26.31 | Strom_C | rene-: the telco is measuring it |
| 01:26.38 | rene- | ok |
| 01:26.44 | pigpen2 | yeah...1 timeing slip every 3 or 4 seconds. |
| 01:28.15 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
| 01:29.56 | tasat- | Strom_C: looks like I'm looking for RDNIS ID, but don't see how Asterisk picks it up from a SIP Invite -- any ideas? |
| 01:30.52 | *** part/#asterisk Terlouw (n=d3vious@80.126.223.172) |
| 01:31.05 | Strom_C | tasat-: like I said, it depends on your provider. Contact them. |
| 01:32.30 | pigpen2 | Ok..update: |
| 01:32.45 | pigpen2 | The tech was running all his tests -through- the smartjack. |
| 01:33.06 | pigpen2 | We had the first pri card toasted about 3 weeks ago... |
| 01:33.16 | pigpen2 | He is replacing the smartjack. |
| 01:33.25 | pigpen2 | (as a test...) |
| 01:33.28 | Strom_C | that's what he should be doing. the smartjack is the demarcation point for that T1 |
| 01:33.43 | Strom_C | so SWBT's responsibility is everything up to and including the smartjack |
| 01:33.51 | pigpen2 | yep. |
| 01:33.57 | pigpen2 | we have seen them all go out... |
| 01:34.04 | Strom_C | why the hell do I keep calling them SWBT? They haven't been Southwestern Bell Telephone Company for ages :) |
| 01:34.16 | pigpen2 | me too.. |
| 01:34.18 | Strom_C | AT&T Southwest |
| 01:34.21 | pigpen2 | SWBT = ATT |
| 01:34.23 | pigpen2 | bastards. |
| 01:34.31 | *** join/#asterisk lo_tech (n=lo_tech@ur45.novo1.com) |
| 01:34.33 | Strom_C | hey, AT&T is a much nicer name than SBC :) |
| 01:34.46 | pigpen2 | definatly...and much better than verizon. |
| 01:35.00 | lo_tech | 'nice' dont make the phones ring :P |
| 01:35.15 | Strom_C | I called SBC California "Pacific Bell" for like three years, but as soon as the AT&T name came around, I dropped "SBC" like a hot potato |
| 01:35.29 | Strom_C | AT&T sounds like a phone company to me :) |
| 01:36.30 | pigpen2 | We have a small texas only telco here called Texlink Communications... |
| 01:36.40 | harryvv | nifty name |
| 01:36.55 | pigpen2 | When we need emergency help, we call and can actually hear the tech "wake up" and get out of bed. |
| 01:37.00 | pigpen2 | ^^^now that is service. |
| 01:37.01 | pigpen2 | hehe |
| 01:37.02 | Strom_C | hahahahaha |
| 01:37.10 | benjk | we have an internet provider here called nifty |
| 01:37.15 | pigpen2 | nifty. |
| 01:37.18 | pigpen2 | sorry...neat. |
| 01:37.21 | Strom_C | exactly why clients must pay me a hefty retainer agreement for 24 hour service |
| 01:37.26 | Strom_C | s/agreement// |
| 01:37.30 | pigpen2 | yeah...same for us. |
| 01:37.51 | pigpen2 | but luckly we have very little issues. |
| 01:37.53 | pigpen2 | :) |
| 01:39.03 | file | Identity acknowledged. Hello Strom. |
| 01:39.33 | Strom_C | Hello file. |
| 01:39.37 | Strom_C | what's new? |
| 01:39.50 | file | I'm about to make lots of people who have trunk checked out minorly unhappy |
| 01:40.11 | file | and yourself? |
| 01:40.19 | benjk | that's boring |
| 01:40.24 | Strom_C | I'm making my apartment a clean and happy place |
| 01:40.28 | lo_tech | file, meet jbot |
| 01:40.29 | benjk | you should do something more exciting |
| 01:40.32 | Strom_C | because right now it's a pigsty |
| 01:40.35 | rene- | cleaning sucks |
| 01:40.40 | rene- | well it doesnt but |
| 01:40.44 | benjk | like make them majorly miserable |
| 01:40.45 | rene- | i have to drag myself to do it |
| 01:40.53 | file | benjk: that would take too much effort |
| 01:41.16 | benjk | rm -rf wouldn't be all that much effort I guess |
| 01:41.51 | benjk | or garble all the files with some random garblizer bash script |
| 01:43.09 | *** join/#asterisk lo_tech (n=lo_tech@ur45.novo1.com) |
| 01:43.58 | *** join/#asterisk lunaphyte (n=lunaphyt@static-71-120-128-10.gdrpmi.dsl-w.verizon.net) |
| 01:44.18 | pigpen2 | Imagine that...the slips are gone. |
| 01:44.20 | pigpen2 | hehe |
| 01:44.29 | *** join/#asterisk flujan (n=flujan@201-42-43-127.dsl.telesp.net.br) |
| 01:44.32 | pigpen2 | Thanks all for helping. |
| 01:45.00 | Strom_C | very welcome :) |
| 01:45.04 | flujan | guys... I am trying like a ginea pig to make the get data "beep" 5000 2 agi command to return something to me... |
| 01:45.28 | flujan | i tried the perl agi-script from asterisk ... I always get the very old result... 0 |
| 01:46.01 | flujan | Shall I give some kind of permission in my linux system to have it working? |
| 01:46.18 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 01:46.21 | rene- | flujan sure |
| 01:46.39 | rene- | you need to have your script be executable |
| 01:47.03 | rene- | your script needs to be executable, i.e chmod 755 |
| 01:47.18 | benjk | yeah, you'll need this ... |
| 01:47.44 | benjk | set permission to work : granted |
| 01:47.51 | benjk | set permission to crash : granted |
| 01:47.59 | benjk | etc |
| 01:54.36 | flujan | rene-, could you tell me which permissions in which devices? |
| 01:54.46 | flujan | rene-, I am currently using slackware linux. |
| 01:57.59 | flujan | rene-, i already have the script running with such permissions... :( |
| 02:00.21 | flujan | rene-, are u here? |
| 02:00.42 | rene- | sorry |
| 02:00.48 | rene- | mm |
| 02:01.06 | rene- | try to use the debug facilities of your agi script |
| 02:01.19 | rene- | and why arent you using ruby, that was the first home run |
| 02:01.49 | rene- | the ragi api is super easy |
| 02:01.49 | rene- | and ruby-agi shouldnt be more diificult |
| 02:01.53 | flujan | rene-, i am using ragi. But i have problems with the get_data method |
| 02:01.57 | rene- | ok |
| 02:02.25 | flujan | rene-, it returns nothing... The get_data playback the message for example, but if i try to use say_digits on it I have error messages |
| 02:02.35 | flujan | rene-, do you use ragi? |
| 02:02.43 | rene- | no but try this |
| 02:03.06 | rene- | use get_data to read DTMF digits from the telephone |
| 02:03.16 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 02:03.18 | rene- | then use ragi debug facility to print the digits on the cli scree |
| 02:03.20 | rene- | screen |
| 02:03.33 | rene- | say_digits not working could be because the sound files arent right |
| 02:03.51 | flujan | hum... how can I print the digits in the cli? |
| 02:04.04 | rene- | there has to be a way, perl agi could do it |
| 02:04.54 | *** join/#asterisk novafirst (n=kosta@wrt1.niclab.com) |
| 02:05.05 | flujan | rene-, i try to print the digits using the RAGI.LOGGER.info() which print the digits in the ruby screen... |
| 02:05.09 | lo_tech | $AGI->verbose("DNID: $DNID",15); will 'echo' to CLI |
| 02:05.24 | novafirst | any good open-source windows based sip-phone??? |
| 02:05.25 | JunK-Y | just agi debug? |
| 02:05.28 | *** join/#asterisk castellvector (n=Administ@adsl-70-240-182-197.dsl.hstntx.swbell.net) |
| 02:05.34 | *** part/#asterisk castellvector (n=Administ@adsl-70-240-182-197.dsl.hstntx.swbell.net) |
| 02:05.52 | rene- | flujan: so you are getting them right? |
| 02:06.12 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 02:06.30 | lo_tech | nova: tons, check http://www.voip-info.org/wiki-VOIP+Phones |
| 02:06.57 | flujan | rene-, no the problem is that i am not getting them... :( |
| 02:07.07 | *** join/#asterisk spr1te (i=spr1te@194.187.130.227) |
| 02:07.32 | rene- | well you might have a separate issue |
| 02:07.46 | rene- | are you testing with a voip phone? |
| 02:08.42 | rene- | what phone are you using to test this |
| 02:08.57 | rene- | Hello JunK-Y |
| 02:09.16 | flujan | x-lite |
| 02:09.36 | rene- | flujan: then you might have different DTMF settings in sip.conf and XLITE configuration |
| 02:09.44 | rene- | that can be an issue |
| 02:09.51 | flujan | hum... |
| 02:12.09 | rene- | i will be right back |
| 02:12.56 | flujan | BAH |
| 02:13.03 | flujan | i tried other softphone |
| 02:13.06 | flujan | and it worked |
| 02:13.34 | flujan | the xten DTMF is not working propertly |
| 02:13.39 | rene- | ;) |
| 02:13.43 | novafirst | is there a way to encrypt-decrypt user pwd's using md5 or any other encryption ??? |
| 02:13.47 | rene- | dont blame ruby! |
| 02:14.06 | JT | encrypt, yes |
| 02:14.10 | JT | decrpty, not easily |
| 02:14.14 | JT | -y |
| 02:14.40 | bkw_ | echo -n username:realm:password | md5 |
| 02:14.49 | *** join/#asterisk vexorg (n=vexorg@CPE0003478eef7c-CM0016b531e87c.cpe.net.cable.rogers.com) |
| 02:14.51 | bkw_ | md5secret= that |
| 02:16.07 | novafirst | I noticed all user pwd inside sip.conf are regular text so I want to implement some kind of encryption |
| 02:16.56 | bkw_ | md5secret is what you want |
| 02:17.02 | bkw_ | instead of secret |
| 02:17.05 | bkw_ | echo -n username:realm:password | md5 |
| 02:17.10 | bkw_ | that will output the string to use |
| 02:17.14 | bkw_ | pay attention ;) |
| 02:17.25 | bkw_ | the default realm is asterisk |
| 02:17.29 | bkw_ | so |
| 02:17.46 | bkw_ | <688>:echo -n brian:asterisk:test | md5 |
| 02:17.46 | bkw_ | 1de0ae23541ff62118c4463b610e13b1 |
| 02:18.23 | novafirst | ok |
| 02:18.50 | novafirst | realm stand for? |
| 02:19.23 | [shodan] | is there anything fun to do with bluetooth headsets/phones & asterisk ? |
| 02:20.17 | lo_tech | shodan: yep... you can mod the dialplan to notice where you are and have it ring the nearest phone when your bluetooth device is near |
| 02:21.05 | lo_tech | shodan: is more of a 'neat trick' than anything useful.. and people will tire of having your calls delivered to their extension just because you're in the same room |
| 02:22.21 | [shodan] | yeah, I heard of that , I was more thinking of receiving calls via the bluetooth device , since most headset have at least a pickup/hangup button |
| 02:23.48 | [shodan] | like I bought one of those, http://www.msicomputer.com/product/p_spec.asp?model=FS300&class=com , to turn my wired headphones into wireless ones , but it'd be neat if I could receive calls through them too |
| 02:24.07 | lo_tech | well, you can also have the calls routed to your bluetooth device when it's detected, provided that it's a bluetooth phone... afaik, jsut because it's bluetooth doesnt mean it can do just anything... |
| 02:25.37 | lo_tech | I know you can get your Bluetooth phone to be a headset, but not the other way around... the headset doesnt have the guts to be a phone |
| 02:28.52 | [shodan] | is there software to do that on windows or is that a linux only thing ? |
| 02:29.52 | lo_tech | software to do which? |
| 02:30.48 | [shodan] | to use the bluetooth phone as a headset |
| 02:32.42 | *** join/#asterisk _deg_ (n=deg@201-40-223-25.ctame704.dsl.brasiltelecom.net.br) |
| 02:32.50 | lo_tech | http://answers.google.com/answers/threadview?id=334278 |
| 02:33.23 | [shodan] | thanks ! |
| 02:35.59 | flujan | rene-, sure... i always will blame proprietary software |
| 02:36.05 | flujan | rene-, I am a rails lover too. :) |
| 02:36.35 | flujan | guys someone knows how to configure dtmf tones in xten-xlite? |
| 02:40.17 | novafirst | is there other way to create md5secret instead this ??? echo -n brian:asterisk:test | md5 ... ? |
| 02:40.52 | russellb | sure, any other way that you can conceive of that does the same operation |
| 02:41.00 | russellb | pick a programming language |
| 02:41.23 | novafirst | bash? |
| 02:41.41 | russellb | well, it's already a shell command ... |
| 02:41.47 | novafirst | I don't have md5 installed :( |
| 02:44.08 | Nugget | if you have openssl installed you do. |
| 02:44.24 | Nugget | echo -n brian:asterisk:test | openssl md5 |
| 02:45.16 | Nugget | md5 might be installed as "md5sum" on your system. I've seen that sometimes on linux systems. |
| 02:45.34 | Nugget | but openssl can do it in any event. if you don't have openssl then md5 is the least of your problems. ;) |
| 02:50.04 | lo_tech | echo -n brian:asterisk:test | md5sum |
| 02:51.41 | lo_tech | echo -n brian:asterisk:test | openssl md5 |
| 02:52.31 | novafirst | md5sum works perfect. |
| 02:52.53 | Nugget | apparently lo_tech took the "echo" seriously. ;) |
| 02:53.09 | lo_tech | ya, but the trailing ' - ' might get you in trouble if you dont clip it |
| 02:55.12 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 02:56.37 | novafirst | lo_tech: yup, ' - ' could (will) be a problem .. openssl md5 works perfect |
| 02:58.05 | novafirst | now, x-lite users will need to update their pwd ??? |
| 02:58.15 | flujan | guys... i am having problems with dtmf detection |
| 02:58.17 | [shodan] | when I call a sip phone with Dial() is there a way I can tell the phone to use a different ring ? |
| 02:58.58 | flujan | how can I configure it? |
| 02:59.18 | flujan | I have problems with the dtmf comming from the pstn |
| 02:59.49 | [shodan] | flujan, I'd say it's something in /etc/asterisk/zapata.conf |
| 03:00.55 | flujan | [shodan], I already set relaxdtmf=yes |
| 03:01.05 | flujan | in the zapata.conf but still have problems... :( |
| 03:02.27 | flujan | I am really deseperated here... I need dtmf detection to create a IVR system... :( |
| 03:02.58 | *** join/#asterisk Splas (n=jwb@brooklyn.paravolve.net) |
| 03:04.04 | [shodan] | try to play with your echo cancelling or gain settings ? (also in zapata.conf) |
| 03:04.47 | *** join/#asterisk tempest1 (n=asf13@c-68-58-187-78.hsd1.sc.comcast.net) |
| 03:05.06 | [shodan] | maybe try to find the cause of the problem ? bad wiring, EM noise source near cables etc.. ? |
| 03:07.13 | lo_tech | shodan: from http://www.voip-info.org/wiki-Asterisk+SIP+channels... search for distinctive ring |
| 03:07.42 | novafirst | Nugget: if I use md5secret how will x-lite users authenticate ?? |
| 03:08.36 | rene- | wtf my intel board has tons of irqs free but insists in give my digium board the same irq and it is sharing it with the smbus device |
| 03:08.48 | rene- | is it possible to disable this smbus thing? |
| 03:09.51 | flujan | [shodan], I removed my echo cancel support now |
| 03:10.55 | flujan | [shodan], http://pastebin.ca/155513 |
| 03:11.03 | flujan | [shodan], please, look my zapata.conf |
| 03:11.10 | Nugget | hell if I know. I was just assisting with getting the md5 hash calculated. |
| 03:11.19 | Nugget | I've never used x-lite or md5 authentication |
| 03:12.23 | [shodan] | oh wait isdn ? I'm sorry I can't help you I only used plain fxos , maybe someone else can help |
| 03:12.26 | lo_tech | rene: yes, but you're prolly going to not like the results... it's an Intel thing so I cant say much about it. Intel has an App Accelerator widget that lets you jiggle it's IRQ tho |
| 03:12.41 | flujan | [shodan], ok... :) thanks for the help. |
| 03:14.23 | lo_tech | flujan: well, it's not really 'DTMF' over isdn, it's D channel signalling... what is happening when you receive a call that leads you to believe DTMF is goofed? |
| 03:15.06 | lo_tech | FYI, tx/rxgain is not really applicable to PRI |
| 03:15.36 | flujan | i am using get data... when i used it with a softphone(moziax) it works. When I use x-lite or the pstn it do not work. |
| 03:15.42 | flujan | returns null. |
| 03:16.05 | lo_tech | overlap dial is prolly eating you up |
| 03:16.25 | cy3o3 | Can you guys reccommend some cheap digium timing hardware for production use? All we'll be doing is some SIP conferencing.. |
| 03:16.54 | JT | lo_tech: actually i think dtmf is sent in band |
| 03:17.12 | StewLG | cy303: Why do you need digium hardware then? |
| 03:17.29 | StewLG | You can do SIP w/o any hardware, I'd think. |
| 03:17.32 | *** join/#asterisk Brijn (n=Bas@S0106004063c0fa1f.vn.shawcable.net) |
| 03:17.40 | flujan | lo_tech, what do you mean by overlap dial? |
| 03:17.41 | Brijn | hello all |
| 03:17.50 | cy3o3 | StewLG: Just need something for timing. We'll likely have 5 clients in the conference at any given point, so I don't want to use the kernel ztdummy route.. |
| 03:17.53 | *** join/#asterisk vexorg (n=vexorg@CPE0003478eef7c-CM0016b531e87c.cpe.net.cable.rogers.com) |
| 03:18.29 | *** join/#asterisk linagee (n=linagee@cpe-66-75-142-207.san.res.rr.com) |
| 03:18.34 | cy3o3 | StewLG: Yeah we can do SIP, but we want to do conferencing (about 5 clients in a MeetMe conference, not just SIP to SIP) |
| 03:18.53 | flujan | lo_tech, do you think it can be a problem with my digium board? |
| 03:19.01 | flujan | lo_tech, or with asterisk? |
| 03:19.13 | JT | flujan: what board are you using? |
| 03:20.07 | lo_tech | sec... inbound calls delivered on a PRI circuit are not 'giving' you DTMF, correct? |
| 03:20.23 | flujan | 406P but we removed the echo cancellation module ( it was making the card do work bad) |
| 03:20.40 | linagee | is SIP over the intarweb secure? should i care necessary? (is POTS secure?) |
| 03:20.42 | flujan | lo_tech, correct. :) |
| 03:21.00 | cy3o3 | StewLG: Any input? :P |
| 03:21.02 | JT | linagee: no |
| 03:21.06 | JT | linagee: it's not secure |
| 03:21.13 | linagee | JT: should i care? heh |
| 03:21.20 | JT | in terms of transmission data |
| 03:21.37 | JT | if you don't trust the path that the packets go over, and the conversations are important, then yes |
| 03:21.38 | lo_tech | kk, that DTMF is sent in-band. any particular reason why you would enable overlapdial=yes? |
| 03:21.49 | linagee | JT: important... hrm... |
| 03:21.56 | JT | secret |
| 03:22.39 | linagee | sexual? |
| 03:22.44 | linagee | lol. j/k |
| 03:23.00 | Brijn | The info on voip-info about a zaptel device needed for TDMoE is still correct? |
| 03:23.20 | linagee | JT: yeah, what if john b. hacker caught you phone sex0ring |
| 03:23.44 | JT | probably not much if it was a random attack and you were a nobody |
| 03:23.51 | JT | may be an issue if you were a senator |
| 03:23.53 | JT | or such |
| 03:23.54 | flujan | lo_tech, i dunno what this option do. :) I will try to enable it now. |
| 03:24.20 | linagee | JT: so you're saying once i start passing voip over the intarweb with no encryption, i shouldn't run for senate? |
| 03:24.37 | JT | hah, do as you wish :) |
| 03:24.50 | JT | is your intarweb super untrustworthy? |
| 03:25.04 | JT | it takes a bit of effort to intercept the RTP voice traffic |
| 03:25.12 | linagee | JT: what if i have no line and it will be voip in, voip out. is it possible to get a voip provider that has a secure connection? |
| 03:25.16 | StewLG | cy303: I'm not expert enough to help. Is there any harm in trying it first, or is your timeframe super critical? |
| 03:25.37 | JT | linagee: don't really know of any that offer secure connections normally |
| 03:25.47 | JT | best way to secure connections atm is with a vpn |
| 03:25.56 | JT | as encrypted sip support is patchy at best |
| 03:25.57 | lo_tech | flujan, based on your pastebin, it's already enabled... we had problems with DTMF on PRI using overlapdial=yes on TE405Ps, so... my suggestion is that you disable overlap and test |
| 03:26.16 | linagee | JT: i guess people just don't care, or it's like, "it's already secure, we know what you're saying and ma bell knows what you're saying..." |
| 03:26.29 | linagee | JT: er, already insecure |
| 03:26.40 | JT | well if the government wants to listen to what you say, they can |
| 03:26.47 | JT | unless you have end to end encryption |
| 03:26.51 | JT | and trust the ends |
| 03:26.52 | JT | :) |
| 03:27.02 | linagee | JT: QUITE EASILY from strom's speeches. :-D |
| 03:27.11 | JT | eh? |
| 03:27.11 | linagee | JT: they make phone switches with built in govt hooks. lol. |
| 03:27.14 | JT | heh |
| 03:27.17 | JT | yeah |
| 03:27.21 | JT | legal requirements |
| 03:27.22 | Strom_C | what are you attributing to me? |
| 03:27.39 | linagee | Strom_C: hehehe. my knowledge of something i only thought as a maybe previously? |
| 03:27.51 | Strom_C | be more specific, please |
| 03:28.00 | linagee | Strom_C: er... calela (sp) |
| 03:28.00 | lo_tech | JT: what makes you think that the gov would allow encryption into the wild that thwy couldnt already spank with their supa-computa-of-doom at NSA? :P |
| 03:28.10 | flujan | lo_tech, ok... Thanks it worked... thanks so much for the help |
| 03:28.11 | linagee | Strom_C: sounds like |
| 03:28.15 | Strom_C | CALEA |
| 03:28.20 | linagee | yes. :-D |
| 03:28.26 | lo_tech | your're not paranoid enough :P |
| 03:28.28 | JT | because it's pretty hard for the government to block useful encryption |
| 03:28.30 | Strom_C | the Communications Assistance for Law Enforcement Act |
| 03:28.32 | lo_tech | welcome flujan |
| 03:28.34 | linagee | Strom_C: should it out! CALEA!!! (if they heard you now..) |
| 03:28.39 | linagee | er, shout |
| 03:28.50 | rene- | they cant eavesdrop on you if you know what you are doing |
| 03:28.55 | JT | the strength of publically available encryption outstrips the NSA's computing power these days, i believe |
| 03:29.20 | linagee | rene-: sure they can. if you piss off the govt enough, they will throw money into resources to do anything to you |
| 03:29.24 | JT | rene-: well they'll go for the lowest effort option, if that means installing a physical bug at your place, then so be it |
| 03:29.30 | lo_tech | rene, you're just decreasing your chances of being listened to... but they're always listening <shudder> |
| 03:29.31 | novafirst | x-lite working fine with md5secret ! |
| 03:29.32 | rene- | thats why the govt want to have backdoors, and wants to make encryption they cant easily |
| 03:29.39 | *** join/#asterisk orlock (i=[2cbxF64@202.44.174.4) |
| 03:29.41 | JT | linagee: doesn't mean they'll crack 2048bit encrytpion |
| 03:29.47 | rene- | break into, ilegall |
| 03:29.47 | linagee | JT: exactly. install a bug at your place, bruteforce with some big govt machines... |
| 03:29.48 | Strom_C | my general theory is this: yes, they can, in theory eavesdrop whenever they like. However, society seems to be doing a reasonably decent job of keeping the government in check. If things start rapidly sliding towards true totalitarianism, then I will be worried. |
| 03:30.06 | JT | break into what? |
| 03:30.16 | JT | b&e is permitted with a warrant |
| 03:30.22 | JT | secret b&e, too |
| 03:30.33 | rene- | they want to declare illegal all encryption they cant easily break |
| 03:30.34 | linagee | JT: more bits doesn't necessarily mean more security. i'm sure they have at least a dozen of the world's smartest mathematicians working for them |
| 03:30.53 | flujan | lo_tech, just a last question Does overlapdial option make asterisk wait until it receives a last digit? |
| 03:31.01 | JT | linagee: sure |
| 03:31.12 | lo_tech | strom: cant be doing much of a job of keeping the gov in check when presidential signing statements are not actionable in court |
| 03:31.22 | JT | but they're going to take the path of least resistance in the course of an investigation |
| 03:31.29 | rene- | with all the fuss about terrorism i guess they can plant you a bug even without a warrant |
| 03:31.48 | JT | heh |
| 03:31.49 | linagee | rene-: i think you should switch to a nice AES crypto. (american encryption equivalency standard. :-D) |
| 03:32.15 | lo_tech | flujan: i didnt really research it much flujan... i just noticed that we had to make that change to get our servers to talk with AT&T and Definity |
| 03:32.19 | JT | here in australia, the ASIO (spys) can enter without a warrant if the director general deems it too urgent to go before a judge |
| 03:32.23 | JT | i'm sure the us has similar |
| 03:32.54 | flujan | lo_tech, ok... So do I learn it too... :) |
| 03:32.57 | linagee | JT: i'm sure bush is going to have all phones tapped in the name of terrorism, then do other searches for things like pornography and such. (google.com anyone? hehehe) |
| 03:33.09 | flujan | lo_tech, thanks again and farewell... :D |
| 03:33.26 | lo_tech | fludan: ya, you research and teach me about overlapdial :) |
| 03:33.35 | JT | phones tapped would imply voice recognition |
| 03:33.42 | orlock | Hmm, does anybody here have remote pickup working? |
| 03:34.03 | linagee | JT: put a firewall on your door. deny access to all traffic. |
| 03:34.20 | JT | ok |
| 03:34.28 | lo_tech | NSA doesnt scare me... I'm harmless. it's the Game Wardens of TX that spook me to death. they never need a warrant |
| 03:34.43 | novafirst | how to control VM menu ?? |
| 03:34.43 | linagee | lo_tech: game warden? |
| 03:35.04 | rene- | they cant, spy on us all.. i dont think is feasible technically or economically, they can spy on some of us, except those who know how to keep themselves safe, and hackers are for the most part non-criminals and non terrorists |
| 03:35.05 | linagee | lo_tech: texas is a spooky place all by itself |
| 03:35.10 | lo_tech | ya, state employees empowered to search/seize without warrant |
| 03:35.17 | linagee | rene-: wtf? how would you know |
| 03:35.17 | lo_tech | anytime/anywhere |
| 03:35.23 | linagee | rene-: it just takes enough money. |
| 03:36.08 | linagee | rene-: what if the govt inserted monitors all over the internet intercepting people's emails. LOL |
| 03:36.09 | lo_tech | for the record, TX was the last state to allow you to carry a loaded wep in a gun-rack in your auto... while driving with an open container of alcohol |
| 03:36.24 | linagee | lo_tech: still legal? yikes |
| 03:36.24 | rene- | linagee: vpn? pgp? |
| 03:36.33 | linagee | lo_tech: and while talking on a cell phone? LOL |
| 03:36.43 | rene- | it is easier for them just to drop you in guantanamo |
| 03:36.43 | linagee | rene-: pretty not good anymore privacy? |
| 03:36.56 | lo_tech | not 'is the last' 'was the last'.... now no open containers, but the gunrack stays |
| 03:36.58 | Nugget | for the record, texas's carry laws are considerably more restrictive than many other states. its reputation isn't deserved. |
| 03:37.01 | JT | what is the purpose of game wardens? |
| 03:37.15 | Nugget | in alabama you can apply for, and receive, a concealed carry permit by mail. |
| 03:37.34 | novafirst | <PROTECTED> |
| 03:37.35 | linagee | JT: to shoot at things when they need shooting at. LOL |
| 03:37.59 | linagee | Nugget: completely by mail? do they allow people to use a fake ID?? |
| 03:38.06 | Nugget | completely by mail. |
| 03:38.18 | linagee | Nugget: do they at least make a notary public witness the ID? lol |
| 03:38.20 | lo_tech | well, in TX you dont need a permit provided you're not in your home county... there is an exemption written into law a couple of centries back that exempt people who are 'traveling' from having wep rights curtailed |
| 03:38.51 | Nugget | tx carry laws are very restrictive, and texas allows cities and towns to override them. |
| 03:38.52 | JT | it is technically feasible for the govt to monitor most Internet traffic, but it's not feasible in terms of the high noise rate they would get for the analysts, trying to find information that actually constitues a threat to national security |
| 03:39.19 | *** part/#asterisk tasat- (n=tasat-@c-67-180-181-221.hsd1.ca.comcast.net) |
| 03:39.22 | linagee | JT: why so? just use md5sum on all the messages and cut out the ones with a high rate of match |
| 03:39.27 | lo_tech | Game Wardens keep people from hunting on state or otherwise posted land, or hunting/fishing out-of-season, etc |
| 03:39.32 | linagee | JT: like if a message is sent to 100 people, ignore that one |
| 03:39.34 | *** join/#asterisk LoneShadow (n=duh@59.92.197.166) |
| 03:39.38 | rene- | linagee: i think that if anything the government has access to stronger crypto than the general populace, and that does not mean they can break easily strong and readily available open source encryption |
| 03:39.41 | LoneShadow | anyone using a callback agi script without freepbx ? |
| 03:40.40 | orlock | man, why isnt pickupexten working for me? |
| 03:40.47 | linagee | rene-: if the govt was smart enough, they would have a queue system for their crypto breaking computers. anything that is actually a terrorist would get the highest spot in the queue. anything where they suspect terrorists or just want to spy on the population would get a lower queue |
| 03:41.30 | lo_tech | rene: do you really believe that the gov would permit 'strong' encryption that it couldn't break? open-source doesn't mean it's necessarily more secure, just that more people have/can review it to find chinks |
| 03:41.56 | orlock | lo_tech: the government doesnt have to "permit" it.. people dont need permission to write code |
| 03:42.00 | rene- | lo_tech: as much as the usa arm wrestles the rest of the world it does not hold a monopoly in science |
| 03:42.33 | linagee | rene-: you're saying another country might clone a human? oh the humanity! LOL |
| 03:42.46 | linagee | rene-: someone please think of the (mutated) children! |
| 03:42.53 | rene- | strong encryption can and has been developed outside of the usa |
| 03:42.54 | lo_tech | orlock: I'm not sure that I agree... the gov has too many levers to pull to get really 'hard' encryption out of the public domain |
| 03:43.12 | JT | but they're not handing it out, ffs |
| 03:43.24 | JT | there is more to the planet than USA |
| 03:43.25 | linagee | rene-: to be safe you can use multiple encryption from multiple sources. :-D |
| 03:43.38 | linagee | layer the encryption like butter on toast. :-D |
| 03:43.56 | rene- | :) |
| 03:43.57 | *** join/#asterisk linlin (i=linlin@c-67-173-38-87.hsd1.il.comcast.net) |
| 03:44.05 | JT | like anzils in quicksand |
| 03:44.06 | rene- | lo_tech: if it is out is out, think decss |
| 03:44.07 | JT | anvils |
| 03:44.49 | rene- | lo_tech: as much as they wanted they cant make the planet spin backwards and trace that evildoer that released stronger crypto to the internet |
| 03:44.53 | cy3o3 | StewLG: Well, just from everything I've read.. I'd expect that we'd run into some timing problems without some timing hardware.. |
| 03:44.58 | linagee | JT: you are the weakest link. goodbye! :-D |
| 03:45.12 | JT | what? |
| 03:45.18 | linagee | it's an old show |
| 03:45.23 | JT | i know |
| 03:45.30 | linagee | it was on television |
| 03:45.40 | linagee | :-D |
| 03:45.42 | JT | your use of it against me was.. lame :P |
| 03:45.49 | JT | weak even |
| 03:45.54 | linagee | JT: it was random at best. :-D |
| 03:45.54 | rene- | the lady that hosted that show in mexico was hot |
| 03:45.58 | cy3o3 | Anyone have any experience with MeetMe conferencing in a production environment? I'm simply looking for a good card that provides some timing.. this is for a production environment with about 5 SIP clients in a MeetMe conference. *shrug* I might just go for the TDM400P |
| 03:46.11 | JT | the lady that hosted that show in australia was not |
| 03:46.31 | [TK]D-Fender | cy3o3 : Anything will do. ZTDUMMY alone can do without the need for special hardware even. |
| 03:46.51 | cy3o3 | [TK]D-Fender: but from what I've read, we'll run into some timing problems |
| 03:47.00 | cy3o3 | I have a budget, so I may as well get some real hardware.. |
| 03:47.13 | [TK]D-Fender | cy3o3 : Any card can provide a timing source. |
| 03:47.22 | lo_tech | cy3o3: we have a farm of servers, some using Sangoma Quad-Ts, some using Digium, some just ztdummy... all are doing MeetMe, none of them are having the age-old problem of timing source |
| 03:47.23 | linagee | JT: if i was a television exec, i'd want the hotornot list of people that are all 9's. turn them into actors. :-D |
| 03:47.27 | [TK]D-Fender | cy3o3 : So buy what your solutuion requires. |
| 03:47.33 | [TK]D-Fender | solution* |
| 03:48.34 | linagee | mixture |
| 03:48.45 | cy3o3 | Okay |
| 03:49.04 | novafirst | [TK]D-Fender: can I customize VM menu when I call in to check my VM messages ? |
| 03:49.38 | [TK]D-Fender | novafirst : Practically : No. If you want to rip open the C source and start modding, then go for it. |
| 03:49.53 | sevard | tkkkkkkkkkkkkkkkkkkkk |
| 03:49.57 | sevard | i need cedegaaaaa |
| 03:50.13 | [TK]D-Fender | sevard : I recognize that name... what do they do again? |
| 03:51.12 | JT | linagee: i wouldn't want the hotornot list |
| 03:51.12 | novafirst | [TK]D-Fender: so I am stuck with "password" greating ? |
| 03:51.13 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 03:52.08 | orlock | lo_tech: its already happened |
| 03:52.14 | orlock | lo_tech: dont you remember zimmerman? |
| 03:52.42 | linagee | JT: no |
| 03:52.43 | linagee | ? |
| 03:53.08 | *** join/#asterisk techie (n=techie@adsl-068-209-242-072.sip.mia.bellsouth.net) |
| 03:53.21 | JT | linagee: bit too populust |
| 03:53.35 | JT | and easily tricked with photochopped or "angle" photos |
| 03:53.36 | orlock | Damn, is anybody here using remote pickups? |
| 03:53.50 | linagee | JT: populust is exactly what TV is after! lol |
| 03:54.18 | linagee | JT: execs want people lusting after the actors. it draws them back to watch the show each week. if that's possible, the show could be total crap and still get ratings. LOL |
| 03:54.29 | JT | well you wouldn't want to get an "angle" on tv would you? |
| 03:55.01 | linagee | angle? |
| 03:55.17 | JT | people who only look good when photographed from very specific angles |
| 03:55.28 | LoneShadow | anyone using callerid match on thier spa3k ? |
| 03:55.39 | linagee | JT: everyone is like that. everyone has the best way to photograph them |
| 03:55.48 | *** join/#asterisk r0d3nt|m (n=RatMan@foster.stonedcoder.org) |
| 03:55.55 | LoneShadow | or is an expert in spa3k in here ? |
| 03:56.01 | JT | err some people who are actually attractive look good from almost any angle :) |
| 03:56.16 | JT | compared to some who you have to struggle for a good angle |
| 03:57.16 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 03:57.20 | Brijn | The host= field in iax.conf is not the IP address of the remote end? |
| 03:58.47 | Strom_C | it is |
| 03:58.58 | Strom_C | or the hostname |
| 03:58.59 | Strom_C | etc |
| 03:59.39 | sevard | [TK]D-Fender: Transgaming's cedega |
| 03:59.49 | sevard | it's like wine on crack plus not confusing as hell and actually works |
| 03:59.59 | Brijn | Hmm, I get "Peer xxx is not dynamic" if I change it to host=dynamic, the registration is OK |
| 04:00.08 | *** join/#asterisk WEH (n=bob@pool-71-250-60-139.nwrknj.east.verizon.net) |
| 04:00.12 | WEH | hello! |
| 04:00.16 | WEH | aloha! |
| 04:00.37 | *** join/#asterisk delmar (n=delmar@60-234-168-34.bitstream.orcon.net.nz) |
| 04:01.15 | WEH | is it necessary to use __TRANSFER_CONTEXT? |
| 04:02.46 | JT | Brijn: because a peer's host cannot by dynamic if you specify it |
| 04:02.58 | JT | Brijn: dynamic implys the hostname/ip is unknown |
| 04:03.05 | JT | and will be determined by registration |
| 04:04.11 | Brijn | JT: I'm confused, in IAX.conf I entered the remote ends hostname in host= , then the registration fails with that error. The error even show the same IP.. Where does the dynamic part come into play then? |
| 04:04.18 | novafirst | I need tv repair ??? |
| 04:04.25 | sevard | .... |
| 04:04.45 | WEH | is it necessary to use __TRANSFER_CONTEXT on a transfer |
| 04:05.13 | JT | Brijn: same result of you put ip instead of dns? |
| 04:05.19 | GlobbeTotssz | hi,, i have my * conectted to a legacy PBX,., i can now reach any legacy extensiion from my *. how do i configgure my * so that a legacy PBX extension can reach any * extension? |
| 04:05.44 | Brijn | JT: Oh hold on, I should not have a register statement on the other end? |
| 04:05.52 | lo_tech | globbe: which PBX? |
| 04:05.59 | JT | Brijn: that might be it, not 100% on that |
| 04:06.00 | GlobbeTotssz | NEC |
| 04:06.10 | JT | but probably |
| 04:06.35 | lo_tech | no clue on NEC... but you basically need to force a group of extensions out on the * server's trunk |
| 04:06.39 | Brijn | JT: Hmmm, but where do I specify the username and password then on the other end, now it's in the register |
| 04:07.16 | Brijn | JT: same as in SIP.conf I gues, with type=friend |
| 04:07.30 | JT | you probably need to register, or use public key auth, or no auth |
| 04:07.43 | orlock | GlobbeTotssz: heh, you know NEC are now doing voip/sip pabx stuff? |
| 04:07.47 | lo_tech | on Avaya PBXs, you can do it by modifying the automatic route selection and the uniform dialing plan.. no clue what the NEC equiv is |
| 04:07.54 | GlobbeTotssz | i can already call from the NEC to any * extension.. but i need to know the configuration to configure my * for incoming calls from the NEC |
| 04:07.56 | delmar | hey everyone. I have a wierd problem to do with the amount of time it is taking to pass Busy/Congestion progress from the carrier to the caller.....when IAX is in use... it seems to take far too long |
| 04:08.49 | lo_tech | Globbe: using PRI to connect to the NEC? |
| 04:09.53 | GlobbeTotssz | span=1,0,0,d4,ami |
| 04:10.05 | GlobbeTotssz | no not pri |
| 04:10.16 | lo_tech | E&M? |
| 04:10.22 | GlobbeTotssz | yes |
| 04:10.47 | lo_tech | so, an extension on the * can call any NEC extension, yes? |
| 04:10.57 | delmar | When connected like... Polycom501(SIP) => Asterisk => DID/VoIP carrier(SIP) ... if there is an engaged number... it's instant to the Polycom. when things are connected like... Polycom501(SIP) => Asterisk1(local) => IAX2 => Asterisk2(remote) => DID/voip Carrier (SIP) ... the BUSY response from the carrier to the remote Asterisk box is instant... the response to the next Asterisk box is delayed, then the response to the Polycom is a |
| 04:10.57 | delmar | lso delayed. Why does it take so long for the BUSY to get to the phone where IAX is being used? |
| 04:11.02 | GlobbeTotssz | i can program NEC to ring on one extension on the * ,, but i wan t to be able to dial any extension from the NEC |
| 04:11.09 | GlobbeTotssz | yes |
| 04:11.44 | lo_tech | glob, can you pastebin your extensions.conf? |
| 04:12.48 | lo_tech | and by 'i want to be able to dial any extension from the NEC' i take it to mean you want to dial non-NEC extensions on the asterisk server.. |
| 04:12.59 | [shodan] | hmm, here's a fun one , how do you return to manufacturer settings on a linksys spa-2102 ? |
| 04:13.17 | clyrrad | is anyone else having trouble with MOH MP3's since upgrading to 1.2.11? |
| 04:13.20 | [shodan] | there's no reset button and the 30 page user manual is no help at all :( |
| 04:13.34 | WEH | four arrow keys or 2+4+6+8 |
| 04:13.43 | WEH | or that's polycom |
| 04:13.53 | GlobbeTotssz | vice verssa.,. i want to dial * extensions from the NEC |
| 04:14.06 | GlobbeTotssz | paste my entire file here? |
| 04:14.16 | Strom_C | ~pb |
| 04:14.19 | jbot | from memory, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/, or http://www.paste-it.net/ |
| 04:14.19 | clyrrad | anyone using Asterisk 1.2.11? |
| 04:14.27 | lo_tech | yes Globbe.. i'm just making sure that you are not trying to dial from the NEC to the asterisk using an extension that is IN the NEC |
| 04:14.36 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 04:14.48 | clyrrad | Strom_C - are you aware of any MOH MP3 problems with Asterisk 1.2.11? |
| 04:15.00 | Strom_C | clyrrad: seems to work for me |
| 04:15.18 | clyrrad | since I upgrade I am getting errors that the mp3's do not exist in any format |
| 04:15.20 | lo_tech | glob: please dont post it here... link it here but post it to pastebin.com or some such |
| 04:15.27 | *** part/#asterisk techie (n=techie@adsl-068-209-242-072.sip.mia.bellsouth.net) |
| 04:15.32 | clyrrad | when they do - I thought at first it may be a permissions issue.... |
| 04:15.37 | clyrrad | but it all works with 1.2.10 |
| 04:15.56 | GlobbeTotssz | that is waht i am trying to do,, dial from the the NEC pbx using an NEC extesiion into any * extension |
| 04:17.52 | [shodan] | oh nvm , found a better manual |
| 04:19.00 | clyrrad | Strom_C any idea why its saying the file does not exist when it does? |
| 04:19.17 | Strom_C | I'm too tired and hungry to debug things right now |
| 04:19.25 | clyrrad | alright |
| 04:19.27 | Strom_C | Please try your call again later |
| 04:19.31 | Strom_C | or ask your operator to help you |
| 04:19.33 | Strom_C | this is a recording |
| 04:19.36 | Strom_C | 213-15 |
| 04:20.08 | lo_tech | i get that Globbe, my point is that the NEC needs to know that the extension being dialed needs to go to the * server. AFAIK, the NEC PBX will not send the call to the * server if the dialed extension exists on the NEC pbx |
| 04:21.05 | lo_tech | so, lets say you have extensions 55000 - 55999 and you want those on the * server... they _cant_ exist IN the NEC as an extension or it will try to send the calls to the NEC extension |
| 04:22.11 | lo_tech | now, if you are able to dial a single extension on the NEC to the * server, you're telling the NEC to send the call to the *... and that's what you'll need to do for the remaining extensions that you want the NEC to send to * |
| 04:22.58 | lo_tech | in short, the NEC needs to know which extensions to send to *... and how that's done is within the NEC itself |
| 04:23.54 | bkw_ | Was Asterisk written in Italy? |
| 04:24.02 | bkw_ | :P |
| 04:24.07 | bkw_ | how is everyone tonight? |
| 04:24.36 | lo_tech | only the parts that like wine, the rest was written in beer-country |
| 04:25.17 | bkw_ | I thought it had something to do with Sphegetti |
| 04:25.43 | lo_tech | soo, I learn something new everyday... actually, it's prolly more 'expresso' than either, knowing the coders I do |
| 04:26.36 | bkw_ | ah |
| 04:26.39 | bkw_ | good point |
| 04:26.43 | bkw_ | but I do neither and I can code |
| 04:26.43 | JT | uhh |
| 04:26.50 | JT | "expresso"? |
| 04:26.50 | JT | ... |
| 04:26.52 | bkw_ | expresso would have me bouncing off the walls |
| 04:26.56 | GlobbeTotssz | the NEC is progemmed so that when i dial 82 i get to the t1 trunck.. right now i have all call coming in from the NEC extension into the * going to extension 400 on the *.. i want Poeple on the NEC to be able to call poeple on * by dialing their extension number |
| 04:27.04 | bkw_ | anyway i'm going to watch teevee have fun |
| 04:27.22 | GlobbeTotssz | the NEC extensions are 2xxx the * is 4xx |
| 04:27.33 | JT | expresso is a common mispronounciation of espresso |
| 04:27.38 | JT | there's no x sound :) |
| 04:27.46 | lo_tech | so, they dial 82400 (82 is trunk access code, 400 is the extension) and that calls the * on extension 400, correct? |
| 04:27.54 | bkw_ | JT this is america.. we say their is :P |
| 04:28.02 | JT | hah |
| 04:28.03 | bkw_ | ok teevee time |
| 04:28.26 | lo_tech | what is this? a speeling bee? |
| 04:28.58 | GlobbeTotssz | yes, but right now if you dial 82 402 or 82 401 it still goes to 400 because that is the way that i programmed it |
| 04:29.00 | JT | a lot of people don't even realise they're mispronouncing it :P |
| 04:29.25 | GlobbeTotssz | i dop not know how to program is so that they can dial the extension that they want |
| 04:29.54 | lo_tech | so, are you wanting to force 4-digit extensions from the NEC to the *? because it sounds like you could dial any asterisk * by using 824xx |
| 04:30.02 | GlobbeTotssz | eg: 82402 to extesion 402, 82400 should go to ext 400 on the * |
| 04:30.48 | lo_tech | when you dial 82409 on an NEC extension, what do you hear and what is on the * CLI? |
| 04:31.41 | GlobbeTotssz | it goes to extension 400 |
| 04:32.03 | GlobbeTotssz | because i opnly know how to send it to one extesion |
| 04:32.21 | GlobbeTotssz | 824xx goes to 400 |
| 04:32.40 | lo_tech | so, on the * server you see that the dialed number is 400, regardless of the number you dialed from the NEC extension? |
| 04:32.57 | lo_tech | by 'you see' i mean in the * CLI |
| 04:33.13 | *** part/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 04:33.14 | nextime | bkw_ : ... and "sphegetti" is write "spaghetti" |
| 04:33.34 | lo_tech | s/spelling Nazi// |
| 04:34.13 | GlobbeTotssz | if i dial 824098 is shows that i dial 409,, but i have all calls coming in on that trunk goingot extension 400 |
| 04:34.22 | GlobbeTotssz | because that is all i kow how to do |
| 04:34.42 | GlobbeTotssz | correction > if i dial 82409 is shows that i dial 409 |
| 04:35.53 | lo_tech | ahh, so ... you just need the extensions defined in your * server.... if the incomming dialed number is the 40x number then you're PBXs are squared away, you're just sending all calls to the 400 extension in the extensions.conf |
| 04:37.19 | GlobbeTotssz | yes,, i want to know what do i need to do on my * to hav it route to the appropriate extension |
| 04:37.42 | lo_tech | when the calls come in on the NEC->* trunk, the * server needs to know which device to send the call to on the * server |
| 04:38.58 | GlobbeTotssz | ok,, how do i do that please? |
| 04:39.06 | lo_tech | so you'll need to tell it where to do within your dialplan... if you're using sip, its something like 'exten => _409,1,Dial(SIP/someextension)' |
| 04:39.18 | lo_tech | do you have phones connected to the * server? |
| 04:39.48 | GlobbeTotssz | yes,, i have 5 extensions 400-405 |
| 04:39.58 | lo_tech | analog phones using Zap? |
| 04:40.05 | GlobbeTotssz | no |
| 04:41.23 | clyrrad | for some reason I have lost all MOH - I get errors that my music files do not exist in any format even though they do.... can any one help? |
| 04:41.28 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 04:41.36 | clyrrad | its only started since I upgraded to the most recent version of asterisk |
| 04:41.43 | clyrrad | but even if I downgrade it still does not work |
| 04:41.50 | lo_tech | well, how are the phones connected to the * server? SIP/Zap/H323? |
| 04:41.53 | clyrrad | can anyone help? |
| 04:42.10 | GlobbeTotssz | on SIP |
| 04:42.26 | GlobbeTotssz | the NEC and * are connected via T1 |
| 04:43.31 | lo_tech | so, add 'exten _409,1,Dial(SIP/whatever)' to the extensions.conf in the same context that the T1 trunk is setup in |
| 04:44.41 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 04:44.52 | GlobbeTotssz | why 409 |
| 04:45.25 | lo_tech | so that the asterisk server knows which device to make 'ringy-lingy' when the 409 extension is dialed. |
| 04:45.58 | GlobbeTotssz | <PROTECTED> |
| 04:46.23 | lo_tech | yep, if you want to be able to dial that extension, the * server needs to know how to get it there |
| 04:46.33 | giesen | how do I treat someone dialing an invalid extension as if nothing happened |
| 04:46.44 | giesen | say for example I have a menu |
| 04:46.47 | giesen | and someone hits the wrong key |
| 04:46.52 | giesen | right now it disconnects the person |
| 04:46.58 | giesen | but I'd rather it just continue playing the menu |
| 04:47.24 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 04:47.27 | lo_tech | giesen: is it giving a timeout on the CLI? |
| 04:47.36 | linagee | giesen: over and over? that might drive someone insane! lol |
| 04:47.47 | giesen | right now it's just falling through |
| 04:47.52 | giesen | I tried putting a noop in there |
| 04:47.55 | giesen | with i |
| 04:48.04 | giesen | but it executes the noop and falls through |
| 04:48.10 | linagee | giesen: if you'd like an operator, please press zero. if you'd like an operator, please press zero. if you'd like an operator, please press zero. if you'd like an operator, please press zero. if you'd like an operator, please press zero. |
| 04:48.16 | giesen | <PROTECTED> |
| 04:48.16 | giesen | <PROTECTED> |
| 04:48.46 | giesen | linagee: right now it just hangs up on them |
| 04:49.16 | linagee | giesen: why not, wait 30 seconds for input. then loop. :-> |
| 04:49.30 | giesen | I guess I could do that |
| 04:49.35 | *** join/#asterisk tasat- (n=tasat-@c-67-180-181-221.hsd1.ca.comcast.net) |
| 04:49.38 | lo_tech | 30 seconds is generous :) |
| 04:49.55 | linagee | lo_tech: i was going to say like 2 minutes, but anyone who is anyone would think the line is dead by then |
| 04:50.12 | lo_tech | 10 second loop with Playback saying "Where's my digits, bish?" |
| 04:50.26 | JT | <PROTECTED> |
| 04:50.42 | linagee | user error. LOL |
| 04:50.44 | [shodan] | what to the phones menu mean by proxy when they say "outbound proxy" ? is it just the addresse of my asterisk box or is that an optional passthrought server like a http proxy ? |
| 04:51.18 | linagee | JT: user incompetant, please replace. user incompetant, please replace. |
| 04:51.22 | linagee | lol |
| 04:51.29 | *** join/#asterisk tabmeister1 (n=troy@CPE00907f17e478-CM014300013422.cpe.net.cable.rogers.com) |
| 04:51.37 | giesen | gah |
| 04:51.42 | tabmeister1 | i keep getting a no authority found error when trying to call an IAX address.. any ideas? |
| 04:51.42 | lo_tech | "Resistance is futile. Resistance is futile" |
| 04:51.53 | giesen | so it just sits there waiting for an extension |
| 04:51.57 | giesen | I like my way better |
| 04:51.59 | tabmeister1 | lo_tech, it depends on what you are trying to resist :P |
| 04:52.04 | giesen | dial in invalid extension |
| 04:52.09 | giesen | it automatically loops |
| 04:52.16 | linagee | lo_tech: please answer more quickly next time. please answer more quickly next time. please answer more quickly next time. :-D |
| 04:52.38 | lo_tech | lol, you dont want to know what I do for IVR scripts :P |
| 04:52.40 | linagee | giesen: can't you do both? |
| 04:53.15 | clyrrad | can someone please help me with this.... |
| 04:53.16 | clyrrad | Aug 31 00:50:04 WARNING[3232]: file.c:512 ast_openstream_full: File /etc/asterisk/ring_tone/RingTone does not exist in any format |
| 04:53.16 | clyrrad | Aug 31 00:50:04 WARNING[3232]: res_musiconhold.c:231 ast_moh_files_next: Unable to open file '/etc/asterisk/ring_tone/RingTone': No such file or directory |
| 04:53.19 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 04:53.27 | linagee | lo_tech: make an IVR that's a real smartass. LOL. |
| 04:53.37 | clyrrad | It just started happening - it can not play any MOH - for the life of me I cant find out why |
| 04:53.44 | clyrrad | I am using native MP3 |
| 04:53.45 | linagee | lo_tech: (for an invalid key) we've detected an idiot at the phone. please call back when you have a clue |
| 04:54.19 | clyrrad | I have also checked the res_musiconhold.so Music On Hold Resource 1 |
| 04:54.24 | clyrrad | its loaded..... |
| 04:54.44 | tasat- | trying to test the load capacity of my system and running into some problems -- my test machine is placing and receiving calls to/from the server. I can do 160+ SIP but there's some gaps/distortion in the audio (strangely no distortion if I make one test call with IAX)... I can only do about 100-110 IAX before Asterisk seems to stall... The CPU usage is very low... only around 6%. Any idea why the audio distortion with the SIP call |
| 04:54.58 | linagee | the new answer in here should be, "have you tried asterisk@home? doesn't it already work on asterisk@home?" :-D |
| 04:55.19 | clyrrad | .... can anyone help me out? |
| 04:55.36 | linagee | clyyrad: have you tried asterisk@home? doesn't it already work on asterisk@home? |
| 04:55.44 | lo_tech | clyrrad: is your musiconhold.conf pointing the sound files to /etc/asterisk/ring_tone? |
| 04:55.51 | clyrrad | linagee I am not using asterisk at home |
| 04:55.59 | clyrrad | lo_tech - yes all the paths are correct |
| 04:56.01 | linagee | no? |
| 04:56.07 | clyrrad | and the files are where they are supposed to be |
| 04:56.13 | giesen | oh |
| 04:56.14 | giesen | another thing |
| 04:56.19 | clyrrad | it almost seems like a permissions issue... |
| 04:56.23 | clyrrad | or |
| 04:56.31 | clyrrad | something wrong with whatever is playing the MOH |
| 04:56.34 | giesen | is there any way I can have the user dial an extension to escape from a queue and go straight to vm? |
| 04:56.36 | clyrrad | but the module is loaded.... |
| 04:56.42 | giesen | so for example |
| 04:56.48 | giesen | they hit 8 while they're in the queue |
| 04:56.55 | giesen | and it takes them to a voicemail box |
| 04:56.55 | linagee | clyrrad: i think my answer was actually pretty good then. :-> |
| 04:57.03 | clyrrad | and the system can play back regular sounds ... like the main menu.... voicemail prompts etc... |
| 04:57.10 | clyrrad | just not MOH |
| 04:57.29 | clyrrad | linagee I can not use asterisk@home |
| 04:57.38 | clyrrad | and that is not a solution |
| 04:57.44 | linagee | no CD burner? |
| 04:57.55 | lo_tech | clyrrad: can you "Playback" one of the files? |
| 04:57.58 | clyrrad | what would using asteriks@home solve? |
| 04:58.10 | clyrrad | yes playback works |
| 04:58.25 | lo_tech | well, permission errors should keep you from doing either playback or MoH |
| 04:58.28 | clyrrad | Playback and Background work |
| 04:58.46 | tasat- | clyrrad: you say it was working before? |
| 04:58.50 | lo_tech | so I'm thinking it's the MoH, but i gave it up for lent |
| 04:59.10 | clyrrad | yes its the MOH.... |
| 04:59.16 | clyrrad | but what could be the fix? |
| 04:59.20 | lo_tech | rather, i think it's MoH and not a file permissions issue |
| 04:59.37 | clyrrad | I would tend to agree with you on that |
| 04:59.42 | clyrrad | I just have no idea how to fix it |
| 04:59.49 | clyrrad | or why it just stopped working.... |
| 05:00.18 | lo_tech | sorry, chief... i swore off using a phone system as a juke-box long ago |
| 05:00.36 | clyrrad | its not a juke-bok |
| 05:00.40 | clyrrad | its on hold music |
| 05:00.51 | lo_tech | and it's different exactly how? |
| 05:01.20 | lo_tech | give em a looping message, "Damn, you're still here?" instead :P |
| 05:01.29 | tasat- | what extension does MOH look for? mp3? |
| 05:01.37 | clyrrad | nah - its something that is missing or needs to be re-installed |
| 05:01.41 | clyrrad | there is no extension |
| 05:01.48 | clyrrad | in the CLI error message |
| 05:01.57 | clyrrad | but the file is a mp3 |
| 05:02.14 | clyrrad | im using the standard mp3's that come with the default asterisk install |
| 05:03.57 | lo_tech | 'moh reload'? |
| 05:04.19 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 05:04.29 | clyrrad | it works with no errors - but still no MOH |
| 05:04.30 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-9-118.cybersurf.com) |
| 05:04.35 | clyrrad | and same error on the CLI |
| 05:05.09 | lo_tech | http://bugs.digium.com/view.php?id=7643 |
| 05:05.24 | clyrrad | Aug 31 00:50:04 WARNING[3232]: res_musiconhold.c:231 ast_moh_files_next: Unable to open file '/etc/asterisk/ring_tone/RingTone': No such file or directory |
| 05:05.52 | clyrrad | yes that is the problem i am having |
| 05:06.11 | lo_tech | read the last entry... |
| 05:07.07 | clyrrad | Fixed in 1.2 as of revision 38825 and trunk in revision 38826. Thanks! |
| 05:07.11 | clyrrad | that entry? |
| 05:07.43 | lo_tech | ya... you said you had the problem after bumping up to .11? |
| 05:07.49 | clyrrad | yes |
| 05:07.58 | file | do you have format_mp3 installed and loaded? if not then Asterisk can not read MP3s |
| 05:07.59 | clyrrad | but now even if i downgrade to 1.2.10 the problem is still there |
| 05:08.14 | clyrrad | file - how can I check that? |
| 05:08.29 | file | show modules in the CLI will show you all loaded modules |
| 05:08.38 | file | if it is not there, then you need to get asterisk-addons 1.2 and install it |
| 05:08.46 | file | (which it probably is not) |
| 05:08.51 | clyrrad | I did show modules like format |
| 05:08.54 | clyrrad | and yes you are correct |
| 05:09.00 | clyrrad | i do not see format_mp3 |
| 05:09.02 | [TK]D-Fender | giesen : Yes you can have a 1-key menu of options that you can allow for people to leave your queue to do whatever it is you want. |
| 05:09.09 | clyrrad | I have everying .... but that |
| 05:09.26 | novafirst | [TK]D-Fender: help, one question |
| 05:10.46 | novafirst | [TK]D-Fender: exten => 1234,2,Macro(sipmacro,1234,${MARK}) how to automaticaly set MARK to SIP/xlite_user1 or SIP/xlite_user2 |
| 05:11.18 | clyrrad | file - you are a life saver!!!!!!!!!!!!!!!! |
| 05:11.25 | clyrrad | that was the problem - thanks bud! |
| 05:11.32 | clyrrad | really apprecate that :) |
| 05:11.38 | novafirst | [TK]D-Fender: is there a system VAR that hold this values? |
| 05:11.54 | file | I would prefer some coke or pepsi |
| 05:11.57 | file | but that's just me |
| 05:12.14 | clyrrad | thanks man I was going nuts..... |
| 05:12.24 | clyrrad | why did that just stop working when I upgraded to 1.2.11? |
| 05:12.28 | [TK]D-Fender | novafirst : Read up on CONSTANTS and where to put them. |
| 05:12.38 | novafirst | ok |
| 05:12.46 | file | you probably wiped out your modules directory because it yelled at you |
| 05:12.57 | clyrrad | ah ha! |
| 05:13.00 | clyrrad | yes you are correct |
| 05:13.03 | clyrrad | i do recall that |
| 05:13.18 | clyrrad | SWEEET thats a good learning lesson - wont forget that |
| 05:13.33 | clyrrad | anytime the modules directory is wipped need to recompile the addons as well |
| 05:13.42 | clyrrad | makes perfect sense - I dont know why I did not think of that |
| 05:14.49 | [shodan] | jeez , how come polycom phones take minutes to boot ? what are they running in there windows embed ?! |
| 05:15.43 | lo_tech | hmm, how could playback work without format? |
| 05:15.54 | tasat- | clyrrad: I ended up trying the echo test app while 160 SIP calls were in progress... got distortion with a SIP phone and through DID... |
| 05:16.34 | rene- | aaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaachu |
| 05:16.52 | *** join/#asterisk tengulre (n=tengulre@222.90.66.4) |
| 05:17.02 | rene- | sorry |
| 05:17.18 | lo_tech | um, that german word that i cant spell |
| 05:17.20 | rene- | [shodan]: even windows start faster |
| 05:17.27 | tasat- | clyrrad: but when I tried with an IAX softphone, it sounded perfect... no idea what the bottleneck is... |
| 05:17.42 | lo_tech | tasat: no translations? |
| 05:17.48 | [TK]D-Fender | [shodan] : Low end CPU. Then again, they are STABLE.. why would you NEED to reboot them? |
| 05:18.06 | tasat- | lo_tech: what kind of translations? |
| 05:18.19 | lo_tech | gsm - 729a, for example |
| 05:18.26 | rene- | codec translations |
| 05:18.29 | tasat- | lo_tech: no, ulaw everything |
| 05:18.40 | tengulre | anybody come from CHINA? |
| 05:18.52 | [shodan] | [TK]D-Fender, well I'm changing settings and everytime I do it's time to get another cola before it's done ;) some with all that caffeinated cola I'm getting even more impatient ;) |
| 05:18.57 | [TK]D-Fender | tengulre : Probably untold millions of Chinese..... |
| 05:19.15 | lo_tech | hmm, cant help you much... we're using servers up to 800 Zap + 200 SIP concurrent with MixMonitor and np |
| 05:19.22 | [TK]D-Fender | [shodan] : Thats because you haven't learned to keep away from the web interface.... |
| 05:19.23 | [shodan] | also if there's a power outtage it kinda sucks to have to wait a couple minutes before making a call :( |
| 05:19.33 | [TK]D-Fender | [shodan] : And its 2 mins max |
| 05:20.06 | JT | if it's a power outage, your PoE switch should be battery/UPS backed ;) |
| 05:20.08 | [shodan] | [TK]D-Fender, I'm using the lcd menu .. but I think there's something wrong because it's taking 3-4-5 minutes |
| 05:20.13 | LoneShadow | any spa3k experts in here ? |
| 05:20.38 | tasat- | lo_tech: no idea why I'd get short gaps in audio and some metallicy distortion with SIP, and not IAX? |
| 05:20.52 | [shodan] | I don't have poe , that used cisco 2900xl was already expensive enough ! |
| 05:21.04 | JT | heh |
| 05:21.15 | [shodan] | beside do 301 even have poe ? |
| 05:21.18 | JT | you can use PoE injectors, you know |
| 05:21.32 | [TK]D-Fender | [shodan] : Thats almost as bad... and definately even more limiting. the only seetings you want on the phone is to set up DHCP and provisioning link. Every other setting should come from a full set of provisioning config files and your phone should be otherwise wiped clean. |
| 05:21.41 | lo_tech | well, it sounds like it's clipping... network is clean of the nasty collisions and such? |
| 05:21.44 | [TK]D-Fender | [shodan] : Yes, with the use of a special cable |
| 05:22.03 | tasat- | lo_tech: to clarify... 160 SIP machine test calls, + 1 human SIP w/distortion (or IAX w/no distortion) |
| 05:22.19 | JT | [TK]D-Fender: what does that mean.. a phone cable that brings out a dc jack for the phone? :) |
| 05:22.46 | [shodan] | I might as well just put the phone on an APC |
| 05:22.55 | tasat- | lo_tech: what sort of collisions? how would I check ? |
| 05:22.58 | JT | depends how many phones i guess |
| 05:23.26 | [shodan] | 5 or 6 when I'll be done but each phone as at least one hub between it and the main switch :\ |
| 05:23.37 | JT | nasty |
| 05:23.38 | [TK]D-Fender | JT : No, that sounds like the IP 501 "semi-injector" cable. The IP 301/501 PoE cable has a small PCB in-line that does the 802.3af negotiation for PoE and then sends it down the line raw. So NO, you can't reasonably make that cable yourself |
| 05:23.48 | lo_tech | well, 'ifconfig' should show you if it's the network |
| 05:23.57 | JT | [TK]D-Fender: ah ok |
| 05:24.30 | [TK]D-Fender | JT / [shodan] : The PoE cable is a bit big and has a nugget in the middle with the PoE circuitry |
| 05:24.31 | [shodan] | it would have been great if they had included a bay for a nicd battery or something |
| 05:24.37 | lo_tech | tasat-: drops/overruns/collisions ... if > 0 then it's something to check |
| 05:24.37 | JT | negotiation? |
| 05:24.42 | JT | does it have a dc-dc convertor? |
| 05:24.47 | rene- | has anybody else has ran into the mixmonitor, queues and chan_local crashing asterisk situation? |
| 05:24.53 | [TK]D-Fender | [shodan] : IP phone needs too much power. Completely unreasonable. |
| 05:25.16 | tasat- | lo_tech: what is it about |
| 05:25.31 | lo_tech | depends on the phone :) early PoE was a monster |
| 05:25.32 | tasat- | SIP/RTP vs. IAX that would cause the noise? |
| 05:25.42 | JT | nicads could supply enough power for some period of time |
| 05:25.43 | [TK]D-Fender | JT : PoE is negociated between the endpoint and the switch. Its not on until the switch KNOWS the device can support it so as not to FRY an innocent PC plugged into it :) |
| 05:25.59 | JT | heh fair enough |
| 05:26.07 | [TK]D-Fender | JT : Yes... MINUTES = "some" :) |
| 05:26.18 | JT | how much power do they draw? |
| 05:26.21 | [shodan] | the wallwart that came with the 301 is rated at 400mw , that's not much |
| 05:26.35 | JT | at what voltage? |
| 05:26.41 | [TK]D-Fender | JT : around 5-12W depending @ 24V typically |
| 05:26.45 | lo_tech | tasat-: the ifconfig will show you if it's a glaring problem with the network... just look to see if the drops, collisions, or overruns are incrementing while you're testing |
| 05:27.14 | [shodan] | 12V |
| 05:27.15 | JT | 5-12W isn't too bad, lots of portable 2-way radios output 4-5W rf, so more goes in obviously |
| 05:27.16 | lo_tech | tasat-: i suspect it's likely not the net, but 1st things 1st |
| 05:27.25 | JT | would last long enough to ride out short outages |
| 05:27.40 | JT | mind you, i'd prefer central lead acid batteries |
| 05:27.55 | [shodan] | a typical cell phone is almost 10 watts , this is 0.4 watts |
| 05:28.21 | [shodan] | I would mind a cell phone battery-sized bay in there |
| 05:28.32 | [TK]D-Fender | JT : With PoE you just shove your whole switch on a UPS. what good is a powered phone without a SWITCH and server to connect to? |
| 05:28.34 | JT | 400mW @ 12V = 4.8W |
| 05:28.40 | lo_tech | tasat-: we're using GB either with jumbo to get that many connections into our servers... if your ifconfig stats are not zero, that's where I'd look... otherwise it's in the * server |
| 05:28.41 | [shodan] | wouldn't |
| 05:28.49 | JT | [TK]D-Fender: well i already mentioned the UPS bit earlier |
| 05:28.57 | JT | obviously ups the switch and server |
| 05:29.02 | JT | and anything else that needs it |
| 05:29.11 | [shodan] | jt this is amps * volts = watts , not watts * volts = watts |
| 05:29.23 | [TK]D-Fender | JT : Math failure, you can't take watt & volts and determine watts as being something different :) |
| 05:29.39 | JT | ah |
| 05:29.43 | JT | i thought 400mA |
| 05:29.47 | JT | chutup |
| 05:29.47 | JT | :P |
| 05:29.54 | [shodan] | heheh |
| 05:30.11 | JT | 400mW is nothing |
| 05:31.18 | [shodan] | oh now I've done it ! "error loading 0004f039adc.cfg" ... rebooting |
| 05:31.32 | [shodan] | this is going to be one long night :\ |
| 05:31.47 | lo_tech | ooooh, you win the prize... in 2 minutes after it reboots |
| 05:32.42 | SwK | amps * volts == watts but only in DC |
| 05:33.09 | [shodan] | I hope this phone got better documentation than that spa-2102 thingy |
| 05:33.17 | tasat- | lo_tech: no errors, drops, or overruns... |
| 05:33.19 | lo_tech | watts / volts = amps, in any state |
| 05:33.25 | *** join/#asterisk cmurphy (n=cmurphy@24-155-147-68.ip.grandenetworks.net) |
| 05:33.26 | JT | lo_tech: not correct |
| 05:33.31 | [shodan] | SwK, well yeah , maybe in a parrallel universe |
| 05:33.34 | SwK | amps * volts == VoltAmps in AC theres a different formula to determine AC Wattage (which eludes me) |
| 05:33.41 | JT | Watts(AC) = VoltAmps * PowerFactor |
| 05:33.50 | SwK | yeah thats it |
| 05:33.53 | SwK | powerfactor |
| 05:33.54 | lo_tech | state, bucko... as in 'not washington, DC' |
| 05:33.54 | [TK]D-Fender | [shodan] : that MAc is 1 char too short... make sure your files are named right.... |
| 05:33.55 | SwK | dug |
| 05:33.59 | SwK | duh heh |
| 05:34.05 | clyrrad | lo_tech - did you see the issue to the MOH issue? |
| 05:34.21 | [shodan] | [TK]D-Fender, what "files" ? ;) yeah , I guess I really need that doc ! |
| 05:34.25 | clyrrad | i mean the solution.... |
| 05:34.30 | lo_tech | yep, tho i dont know how it could do playback on a file that it didnt have format loaded for |
| 05:34.45 | clyrrad | yea it was strange... |
| 05:34.58 | lo_tech | have 2 files with same name and dif extension? |
| 05:34.58 | clyrrad | but re-compile the asteriks addons and it worked right away |
| 05:35.07 | clyrrad | nope - just one |
| 05:35.26 | SwK | [shodan] you were sure to make sure in those polycom config files to make the macaddy.cfg file have all lower case alpha characters right? |
| 05:35.31 | clyrrad | it was just missing the format_mp3 module |
| 05:35.36 | lo_tech | so was 'playback' the wav instead of mp3? only thing in can think of on why it would play without format loaded |
| 05:35.47 | tasat- | lo_tech: what're your server specs? |
| 05:35.48 | clyrrad | perhaps file format does not matter when you envoke the call using MOH? |
| 05:35.52 | clyrrad | instead of Playback? |
| 05:36.19 | [shodan] | SwK, I didn't name any files , all I've done so far was mess with the lcd menus |
| 05:36.33 | clyrrad | unless the SetMusicOnHold and MusicOnHold require that format_mp3 module to be installed? |
| 05:36.38 | clyrrad | its the only thing i can think... |
| 05:36.51 | SwK | [shodan] so you didnt set up a boot server and all that eh? |
| 05:37.19 | lo_tech | tasat-: dual 3.1s with 4Gb ram, U320 SCSI Raid1... overkill, really. but our problem is most often with the IRQs using digiums than with proc power |
| 05:37.30 | [shodan] | nope, and it was kinda working before I started messing around with it ;) |
| 05:37.42 | clyrrad | anyhow - lo_tech thanks for troubleshotting it with me |
| 05:37.48 | clyrrad | learning something new none the less |
| 05:37.53 | [TK]D-Fender | lo_tech : Change your card then. |
| 05:38.11 | clyrrad | hey TKD :) |
| 05:38.14 | SwK | [shodan] go find the sip firmware from polycom's website (its a PITA to find but its there) set up a tftp, ftp, or (if you have a X01 phone) HTTP server to boot from you'll be a lot happier and its easier |
| 05:38.21 | tasat- | lo_tech: and you said 200 SIP and 80 ZAP? |
| 05:38.35 | SwK | I have never gotten the "lcd" or local http configs to actually work right 100% of the time |
| 05:39.02 | lo_tech | TK: not having problems here, jst saying more often than not our problems deal with having to use 'alot of quad pri' instead of using a DS3 card |
| 05:39.07 | *** join/#asterisk tengulre11 (n=tengulre@221.11.5.180) |
| 05:39.11 | [TK]D-Fender | SwK : All of their phones can provision every which way. you just need to upgrade the BR. |
| 05:39.20 | lo_tech | tasat-: 800 Zap w/200 SIP |
| 05:39.39 | [shodan] | hmm I have a http server already setup , I guess I'll try to use that , I don't care to setup a full ftp server very much |
| 05:39.43 | [TK]D-Fender | lo_tech : Yes... not channelized DS3 yet.... |
| 05:39.48 | bhrobinson | Hi, is anyone here an expert with the PAP2? |
| 05:39.50 | SwK | [tk]D-fender: wrong answer... the X00 series phones dont do http at all |
| 05:40.01 | SwK | even with the newest bootroms |
| 05:40.28 | [TK]D-Fender | [shodan] : FTP is remarkably easier, faster, and more secure than HTTP.... |
| 05:40.47 | [TK]D-Fender | SwK : Really? News to me... I should go read up on that... |
| 05:41.03 | lo_tech | TK: yep, so we have to use a bastard setup with tons of quad-pri instead... if it wasn't NFAS we'd use something sensible like HP DL360s with 2 quad-pri each instead of these mongoloids |
| 05:41.07 | [TK]D-Fender | SwK : I'll go over my release notes tomorrow |
| 05:41.57 | [shodan] | well I guess so , but I haven't set one up in ages :\ I wish they'd have used scp ! |
| 05:42.20 | SwK | [tk]d-fender: yep... I wish the 00's would do it... https is actually the best way to go...but you end up supporting them under tftp more then you like because everyone else out there (other manufacturers that is) want to use tftp for configs |
| 05:42.57 | [TK]D-Fender | SwK : I use FTP exclusively. 100% success. |
| 05:43.09 | SwK | lo_tech: you know I have something sitting at my office you might be interested in... |
| 05:43.20 | [shodan] | that's weird in setup->Server Menu->Server Type my options are only FTP and TFTP , no http ?! |
| 05:43.22 | SwK | well leave 2 secs too soon heh |
| 05:43.33 | [shodan] | wait is this even a 301 like I ordered .... |
| 05:43.34 | SwK | [shodan] what model phone i sit? |
| 05:44.24 | [shodan] | guess it's 301 "2201-11301-001" |
| 05:44.37 | SwK | that's a 301... upgrade the bootrom |
| 05:44.56 | [TK]D-Fender | Ok, I'm off, back tomorrow (later today more like...) |
| 05:44.59 | [TK]D-Fender | ciao |
| 05:45.02 | rene- | later |
| 05:45.03 | [shodan] | bye [TK]D-Fender |
| 05:45.08 | SwK | peace |
| 05:45.09 | rene- | D-Fender |
| 05:45.17 | rene- | how much hours of sleep do you get per day |
| 05:45.34 | SwK | what is sleep? heh |
| 05:45.43 | JT | for the weak |
| 05:46.07 | rene- | ;P i like to sleep |
| 05:46.21 | [shodan] | can't get the 3.2.1 bootrom from polycom's eh ? |
| 05:46.24 | rene- | i only sometimes go into hacker binges |
| 05:46.29 | rene- | not everyday |
| 05:49.14 | rene- | just imagine the poor health: bad vision, eating crap all day, sitting in a chair all day, repetitive strain disorders, no exercise, loads of caffeine, not talking much to people in the traditional way, and you want to add sleep disorders to the list, bad habits must go out of the window, there is need that some brilliant hacker or somebody who is very visible start advocating for good habits for the programmers/hackers |
| 05:49.41 | [shodan] | hmm when the phone says "boot server" does it actually mean a BOOTP like for booting diskless clients ? |
| 05:50.21 | [shodan] | bah I don't see talking to people in the traditional way that important ;) |
| 05:50.23 | rene- | [shodan] it could mean dhcp or tftp |
| 05:50.44 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 05:52.58 | [shodan] | this is really a business phone, how many people have a dhcp boot and ftp server in their homes ! |
| 05:53.24 | [shodan] | and not just any dhcp server, a real one that you can give per client special options ! |
| 05:53.25 | Strom_C | also, how many people have Asterisk servers in their homes? :) |
| 05:53.36 | joe | more than you think |
| 05:53.43 | JT | can't most dhcp servers do that? |
| 05:53.46 | Strom_C | it was rhetorical |
| 05:54.11 | JT | oh wait, there are zillions of hardware routers |
| 05:54.11 | *** join/#asterisk Rahail (n=rahail1@209.19.88.243) |
| 05:55.11 | [shodan] | luckily my home is so wired a 24 port switch is not enough to accomodate all the ethernet devices and it just happens that I was playing with bartpe over ethernet ,or else this long night would have just turned into one long week ! |
| 05:55.12 | rene- | [shodan]: those are hardly phones to use in your home. linksys/grandstream would be more appropiate |
| 05:56.01 | JT | appropriate, if you like to settle for second best just because it's your house :) |
| 05:56.11 | rene- | [shodan] other than that, those phones are crap to configure, horrible boot times, repulsive web ui, to get them working i had to play with web configuration, direct configuration on the phone interfase , and tftp/dhcp config |
| 05:56.22 | rene- | and they would still do stupid things |
| 05:56.28 | Rahail | any one know what is the code for Call Waiting |
| 05:56.49 | JT | [shodan]: bartpe? |
| 05:57.06 | [shodan] | Rahail, depends on your pots provider I guess |
| 05:57.12 | [shodan] | JT, nu2.nu/pebuilder |
| 05:57.40 | [shodan] | windows pre-installation environement thingy , it's like knoppix for windows |
| 05:57.44 | Rahail | i am trying to use it extension to extension |
| 05:57.48 | Rahail | i pressd *70 |
| 05:57.51 | Rahail | still dont work |
| 05:57.56 | JT | ah ok |
| 05:58.03 | rene- | the only reason i buy them is because they dont look cheap and are not assembled in china, because in my country chinese telephones pay compensatory taxes, 300%, and those are in my opinion stupid protectionist and outdated laws as if mexico was a prime voip phone manufacturer |
| 05:58.15 | *** part/#asterisk rene- (n=rene1@gea-gye-internet.telconet.net) |
| 05:58.24 | JT | 300% wow |
| 05:58.26 | Strom_C | a bit angry there, eh? |
| 05:58.27 | *** join/#asterisk daysmen3 (n=primus@host86-139-212-246.range86-139.btcentralplus.com) |
| 05:58.55 | [shodan] | it's protecting US imports not mexico exports ;) |
| 06:00.38 | *** part/#asterisk tempest1 (n=asf13@c-68-58-187-78.hsd1.sc.comcast.net) |
| 06:02.09 | *** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135) |
| 06:02.48 | Rahail | man I cant get call waiting |
| 06:02.48 | Rahail | :( |
| 06:02.48 | [shodan] | so, what's the deal with the latest bootrom not being freely available from polycom ? |
| 06:03.30 | Strom_C | [shodan]: beats me |
| 06:03.39 | Strom_C | ive not had great experiences with configuring the polycom phones though |
| 06:06.16 | [shodan] | it's not going well at all for me ! now the phone justs boots in loop , I think the only way I can get it to even finish booting is making one of those configuration files from scratch and upload it through a tftp server . they could have at least included a reset to manufacturer settings option ! |
| 06:06.46 | Strom_C | [shodan]: which phone did you get? |
| 06:07.26 | [shodan] | it's a 301 |
| 06:07.57 | Strom_C | ah |
| 06:08.00 | tengulre11 | HI,alll |
| 06:08.10 | Strom_C | i have an install to do this weekend where i have to configure two dozen of those |
| 06:08.16 | tengulre11 | GOOD AFTERNOON!!!! |
| 06:08.18 | Strom_C | i'm not looking forward to it |
| 06:08.27 | Strom_C | tengulre11: shut up and hello |
| 06:08.38 | tengulre11 | Strom_C, :( |
| 06:10.38 | [shodan] | the polycom phone maybe be great hardware but software-wise they stink :\ |
| 06:11.20 | Strom_C | yeah, polycom is an ancient Algonquin word which means "takes FOREVER to boot" |
| 06:11.20 | [shodan] | I hope their tftp stack is solid , I don't want to try 5 tftp servers :\ |
| 06:11.29 | [shodan] | hahahah |
| 06:12.45 | joe | [shodan]: I have over 60 of them and yes they take forever to boot but are damn solid once they're setup |
| 06:13.11 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 06:14.40 | Strom_C | joe: do you know of a configuration utility that autogenerates configs for these things? I'm dreading the thought of having to wade through seventy completely incomprehensible XML files. |
| 06:15.29 | joe | Strom_C: I'm writting one :/ did them all by hand first round |
| 06:15.34 | [shodan] | are they meant to download their configs from ftp everytime they boot, I mean I'd hate all the phones to go down just because the dhcp or ftp server is down ! |
| 06:15.42 | joe | Strom_C: doesn't take that long |
| 06:15.59 | Strom_C | joe: compared to a cisco phone, it's forever |
| 06:16.23 | joe | [shodan]: depends how you set them up |
| 06:16.51 | joe | Strom_C: true but they're so damn cheap... I got mine for ~110 |
| 06:17.38 | Strom_C | cheap means nothing if it means I have to waste more than the price difference in the amount of time it takes to configure and install |
| 06:17.56 | joe | hehe, true |
| 06:18.09 | Strom_C | I'd rather spend a good chunk more money on something which is relatively easy to set up |
| 06:18.16 | joe | but it really wasn't that bad |
| 06:18.36 | joe | after the first 5 it just made sense and didn't take long at all |
| 06:18.45 | *** join/#asterisk vlt (n=dm@p54B339B1.dip0.t-ipconnect.de) |
| 06:18.56 | Strom_C | joe: are you crazy? Last time I tried to configure one of these things, I had to spend an hour just figuring out where the line breaks in the XML file were supposed to go |
| 06:18.57 | joe | configured in less than a minute and three times that too boot them up |
| 06:19.13 | *** join/#asterisk Aurs (n=Aurs@host-81-191-123-189.bluecom.no) |
| 06:19.20 | [shodan] | they may be great phones , but before I buy another polycom I will buy at least one grandstream and one snom |
| 06:19.30 | [shodan] | yeah it's that first one that is a bitch to setup ;) |
| 06:19.32 | Strom_C | [shodan]: also try a cisco |
| 06:19.55 | joe | Strom_C: didn't use the xml files at all except for the diretories. I did it via the phone menus, old school style |
| 06:20.10 | Strom_C | ugh |
| 06:20.55 | joe | yah, but it was done in less than 2 hours |
| 06:21.05 | Strom_C | id rather spend an hour in vi than an hour at a phone menu |
| 06:21.07 | Strom_C | TWO HOURS?! |
| 06:21.26 | joe | at least half of that was waiting for the damn things |
| 06:21.40 | Aurs | provisioning... |
| 06:21.41 | Strom_C | that's $300 worth of time, you realize |
| 06:21.42 | [shodan] | joe, that's how I killed my 301 ! damn menus ! |
| 06:22.05 | joe | which is less than one cisco phone ;P |
| 06:22.08 | Aurs | polycom sucks to manually configure, cause it has to reboot after every change, and that takes like 5 mins every time |
| 06:22.32 | Strom_C | joe: maybe, but i can configure the cisco phones in my sleep |
| 06:22.57 | joe | <shrug> |
| 06:23.08 | joe | whatever works for yah :) |
| 06:23.33 | [shodan] | the cisco phone are just too expensive for anything less than large entreprises |
| 06:24.06 | Strom_C | yeah, but I've loved mine for almost two years now :) |
| 06:24.36 | joe | I have a 301 and 501 at home and love them too :P |
| 06:25.28 | Strom_C | but yeah, I had to order an IP430 to practice on so that I don't waste six hours of the client's time this weekend |
| 06:25.36 | Strom_C | that concept in and of itself is ridiculous |
| 06:25.40 | Strom_C | oh well |
| 06:25.52 | *** join/#asterisk sumasuma (n=sumkji@61.14.86.23) |
| 06:26.09 | sumasuma | is it possible to have signal and rtp in seperate paths with IAX2 ?? |
| 06:26.17 | *** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 06:26.27 | Strom_C | sumasuma: there is no rtp in iax2 |
| 06:26.38 | linlin | is it possible to tunnel IAX2 through SSH ? |
| 06:27.29 | Strom_C | linlin: I don't see why not |
| 06:27.36 | linlin | well its udp |
| 06:27.40 | JT | well |
| 06:27.47 | JT | it's be a crackhead thing to do |
| 06:27.49 | JT | ssh is tcp |
| 06:27.54 | linlin | yeah |
| 06:27.56 | JT | it'd probably be awful |
| 06:27.59 | linlin | i need to get through a firewall |
| 06:28.06 | linlin | that allows only sttp, https, and ssh |
| 06:28.11 | JT | try some other sort of vpn? |
| 06:28.11 | JT | hrm |
| 06:28.22 | JT | does it do packet inspection for application type too? |
| 06:28.25 | sumasuma | Strom_C: i know there is no rtp |
| 06:28.54 | sumasuma | Strom_C: Is it possible to route Signa and Voice packets in seperate paths in IAX2 ? |
| 06:28.59 | sumasuma | *signal |
| 06:29.07 | Strom_C | sumasuma: why do you want to do that |
| 06:29.46 | sumasuma | Storm_C: I have a calling card system, i'm receiving a call from one party and termination through another party, i just need to monitor how much time the call should go |
| 06:30.24 | Strom_C | my name is not Storm |
| 06:30.45 | linlin | haha |
| 06:30.45 | Strom_C | but anyway, you want to stay in the media path with that one so you can cut the call off when the time is used up |
| 06:30.47 | novafirst | if using macro how to call exten => s,3,Playback(vm-goodbye) after user leaves vm |
| 06:30.55 | Strom_C | so use notransfer=yes in iax.conf and be done with it |
| 06:31.21 | *** join/#asterisk jart (n=user@ool-4356f82f.dyn.optonline.net) |
| 06:31.57 | sumasuma | Strom_C: I don't want to take media to my server, i just need to know the total time, i don't care about the media data |
| 06:33.12 | Strom_C | so don't put notransfer=yes in your iax.conf and asterisk will drop out of the media path if the two endpoints can talk with each other directly |
| 06:33.15 | jart | I <3 asterisk |
| 06:33.35 | sumasuma | Strom_C: A --> B --> C B is my server and I want to know how much time did A and B talked, and also the call is intiated by B |
| 06:33.40 | novafirst | anyone ??? |
| 06:33.55 | tengulre11 | how to dial and hear voice from web? |
| 06:34.19 | Strom_C | sumasuma: you are making little sense |
| 06:34.21 | sumasuma | Strom_C: sorry, how much time A & C talked, mentioned wrong |
| 06:34.29 | jart | stop making sense |
| 06:34.34 | Strom_C | talking heads |
| 06:34.39 | jart | YES |
| 06:34.43 | Strom_C | <3 |
| 06:34.54 | jart | <3 <3 |
| 06:34.59 | sumasuma | Strom_C: got my problem ? |
| 06:35.01 | sumasuma | <3 |
| 06:35.14 | Strom_C | sumasuma: yes, and i gave you the answer already |
| 06:35.53 | sumasuma | Strom_C: i removed the notransfer=yes in iax.conf, rest will do ? |
| 06:36.02 | Strom_C | yes |
| 06:36.32 | sumasuma | Strom_C: Media will go between A & C and I will be notified when the call gets hanged up by anyone of them ? |
| 06:36.42 | Strom_C | i believe so |
| 06:36.56 | sumasuma | Strom_C: thanks, i will check it out and be back |
| 06:37.49 | Strom_C | damnit, I can't find any sort of polycom autoconfig utility |
| 06:38.30 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43-static.72-81-b.business.telecomitalia.it) |
| 06:39.26 | sx-wks | haha... |
| 06:39.39 | sx-wks | the TDM31B showed up at my door |
| 06:39.48 | Strom_C | sweet |
| 06:41.29 | jart | hey, is there an app or function or something to check if an account code is in use? i.e. check all the active channels to see if any has CDR(accountcode) set to X |
| 06:43.12 | jart | hey, is there an app or function or something to check if an account |
| 06:43.12 | jart | <PROTECTED> |
| 06:43.13 | jart | <PROTECTED> |
| 06:43.25 | novafirst | inside macro how to define Playback(vm-goodbye) so that after user leaves the VM it plays Playback(vm-goodbye) ??? help! |
| 06:43.42 | *** join/#asterisk adorah (n=Administ@87.68.158.242.cable.012.net.il) |
| 06:43.57 | Strom_C | novafirst: calm the hell down |
| 06:44.12 | Strom_C | you put the playback() command right after the voicemail() command |
| 06:44.16 | Strom_C | done and done |
| 06:44.43 | novafirst | Calm down ?? I tried that and is not working -> exten => s,3,Playback(vm-goodbye) |
| 06:45.01 | JT | OMG EMERGENCY |
| 06:45.23 | Strom_C | show me the whole macro |
| 06:45.27 | Strom_C | use pastebin |
| 06:45.43 | jart | spencer has just raised the fun level to astcon 3 |
| 06:45.56 | *** join/#asterisk HaMYaI (n=hamyai@ppp-58.8.9.29.revip2.asianet.co.th) |
| 06:46.10 | novafirst | Strom_C: http://pastebin.ca/155678 |
| 06:46.16 | *** join/#asterisk sergee (n=opera@195.94.224.197) |
| 06:46.34 | HaMYaI | what else can we do during MusicOnHold()? |
| 06:46.45 | HaMYaI | can anyone give me some hints? |
| 06:46.51 | tessier | Snom 320 is da bomb |
| 06:46.57 | Strom_C | novafirst: well obviously its never going to get to priority 3, because in priority 2 it goes to a completely different extension within the macro |
| 06:47.05 | tessier | Wish I had used only Snom phones all along. |
| 06:47.11 | jart | tessier: 1995 called, they want their slang back |
| 06:48.05 | tessier | jart: Yo momma called. She's having a sale. |
| 06:48.12 | novafirst | Strom_C: how can I fix ? |
| 06:48.36 | Strom_C | novafirst: follow the execution flow of the macro and put the command in the appropriate place |
| 06:49.00 | jart | don't be diss'n me ya'll |
| 06:50.02 | novafirst | I thought exten => s,2,Goto(s-${DIALSTATUS},1) if status is exit it will play Playback(vm-goodbye) . Now I realy don't have a clue what to do. |
| 06:50.08 | *** join/#asterisk crCernier (n=crochat@adsl-84-227-131-144.adslplus.ch) |
| 06:50.24 | Strom_C | novafirst: well, either learn how extensions.conf works, or hire a consultant |
| 06:50.42 | Strom_C | novafirst: I will be happy to write your macro for you at my standard hourly rate |
| 06:51.36 | Snake-Eyes | Any one have any idea how hard it is to install Voicetronix card for Asterisk is ? |
| 06:51.40 | novafirst | Strom_C: Nice place to advertise your service !!! Get lost ! |
| 06:51.50 | JT | novafirst: you're a fucking rude prick |
| 06:52.04 | JT | novafirst: with an attitude like that, you'll be lucky to get any help around here |
| 06:52.09 | sx-wks | TDM31B installed in box |
| 06:52.34 | novafirst | Anyone that can pay consultant will not spend hours trying to fix and learn them self |
| 06:52.50 | Strom_C | novafirst: read the book |
| 06:52.52 | Strom_C | ~book |
| 06:52.59 | jbot | i heard book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 06:52.59 | JT | and spend hours reading up on how extensions.conf works |
| 06:53.00 | *** join/#asterisk SaTLaN32 (n=satlan32@212.150.142.211) |
| 06:53.00 | novafirst | JT: yup ... Wolfs |
| 06:53.05 | SaTLaN32 | hello |
| 06:53.12 | jart | hello |
| 06:53.13 | SaTLaN32 | need help with E1 card from Rhino |
| 06:53.34 | novafirst | this channel is not open so that you can sell your service .. .go to rentacoder ! |
| 06:53.39 | SaTLaN32 | i have installed and configured the card, zaptel and zapata, but i don't have sync with the switch |
| 06:53.39 | JT | we get all sorts of people in here, some are doing asterisk for large companies |
| 06:53.47 | JT | who can afford a bit of consulting hours |
| 06:54.06 | adorah | <SaTLaN32>r u still fuc**** around with Rhino? the heat in BS must be very strong..:) |
| 06:54.28 | JT | you may have seen it as him pushing his services, but he provided a completely free alternative too |
| 06:54.32 | crCernier | Hello ! |
| 06:54.36 | novafirst | anyone asking any money in this or any channel should be baned. |
| 06:54.41 | novafirst | blah blah |
| 06:54.41 | Snake-Eyes | novafirst, assume you don't want any one to help you |
| 06:54.49 | Snake-Eyes | you should be banned |
| 06:54.51 | Ryushin | Hey all. I think I screwed up. I have a Sangoma A104D card for the incoming PRI. The Polycom IP phones for the phones, but I didn't take into account the two fax lines that have to be hooked up. |
| 06:55.05 | crCernier | I have an AG168V ATA adapter (with "line" and "phone" connectors) |
| 06:55.14 | Ryushin | What card should I be looking at to add 2-4 analog lines for the fax machines? |
| 06:55.17 | JT | and you can always say "no, i'm not interested in your commercial services", instead of swearing your head of being extremely rude telling people how to run a channel when you've only just got in there |
| 06:55.19 | novafirst | nice to see how the wolfs are protecting their crew |
| 06:55.22 | JT | well this is IRC |
| 06:55.29 | jart | novafirst: are you an op? |
| 06:55.29 | JT | it is not a democracy |
| 06:55.32 | JT | wolf? |
| 06:55.38 | JT | what the hell are you on about? |
| 06:55.43 | crCernier | I think we could name "phone" FXS and "line" FXO, no ? |
| 06:55.46 | JT | you are deranged and paranoid |
| 06:55.49 | Snake-Eyes | feels like novafirst is a troll |
| 06:56.00 | adorah | <crCernier>In a way.. |
| 06:56.03 | novafirst | maybe |
| 06:56.09 | crCernier | Does anybody have experience with this AG168V ATA ? |
| 06:56.14 | Strom_C | Ryushin: Digium TDM400P is good for FXS ports ;) |
| 06:56.40 | adorah | <crCernier>fxo doesn't generate but receive voltage/dialtone etc.. |
| 06:56.55 | giesen | is there any way to have asterisk play a prompt for ackcall in a queue, prompting the agent to press # to accept the call? |
| 06:56.58 | novafirst | whole fuc.... day I was listening politics and now I need to listen how someone is trying to make money on giving advise! |
| 06:57.03 | jart | btw, Strom_C, would you mind telling me what your rate is? j/c |
| 06:57.14 | Strom_C | heheheh |
| 06:57.16 | Aurs | omg Strom_C... what were you thinking </irony> |
| 06:57.25 | JT | novafirst: go cry me a river |
| 06:57.33 | jart | novafirst: what do you want? a cookie? |
| 06:57.37 | JT | the world's smallest violin is playing |
| 06:57.42 | JT | a very sad song |
| 06:57.46 | novafirst | kis.. my a... |
| 06:57.48 | crCernier | adorah: Actually, I want to forward incoming calls from PSTN to Asterisk with SIP (or IAX)... is that possible with this ATA ? |
| 06:57.50 | Aurs | haha |
| 06:57.55 | SaTLaN32 | so anyway, |
| 06:58.08 | SaTLaN32 | how do i see the card status in asterisk? |
| 06:58.09 | Ryushin | Strom_C: Wouldn't it be better to use Sangoma since I already own it, or should it get the Digium part? |
| 06:58.30 | Strom_C | Ryushin: beats me; ive never tried running a sangoma board and a digium board in the same PC |
| 06:58.38 | adorah | <crCernier>Try Sipura 3000 or the like with both fxs and fxo |
| 06:58.42 | novafirst | JT: I would love to meet you in person, than you will understand why not to fuc.. with some people :) |
| 06:58.57 | JT | novafirst: blah blah blah whinge whing threat threat |
| 06:59.07 | JT | gotta love irc violence threateners |
| 06:59.10 | adorah | <crCernier>google for guides howto config it.. |
| 06:59.13 | Snake-Eyes | Strom_C, seeing you're up on fxo cards: have you ever used a Voicetronix card ? |
| 06:59.17 | sx-wks | yeehah ... |
| 06:59.19 | Strom_C | Snake-Eyes: no |
| 06:59.33 | novafirst | JT: don't make me look for you :) |
| 06:59.43 | novafirst | you know I can find you |
| 06:59.49 | JT | novafirst: try to shut the hell up, don't make us hear your crap |
| 06:59.53 | JT | bullshit |
| 06:59.54 | Snake-Eyes | damn, trying to find out how painful its going to be to use a Voicetronix card and Astersik :P |
| 07:00.14 | Strom_C | Snake-Eyes: just buy the digium card and save yourself the trouble |
| 07:00.17 | novafirst | I listen your crap about politics for 2 dam hours now is my time . |
| 07:00.32 | JT | i don't recall chatting about politics for 2 hours? |
| 07:00.46 | Snake-Eyes | Strom_C, i would if I was allowed to make that decision :) |
| 07:01.09 | [shodan] | what's all the fuss ? |
| 07:01.14 | crCernier | adorah: I dived in the AG168V documentation, but I didn't see any clear information to make that... also, the firmware is OpenSource... maybe the source code will be simpler to understand than the documentation ;-) |
| 07:02.01 | Strom_C | [shodan]: i jokingly suggested that novafirst might want to consider paying me a full hour to do a two-second macro fix, and now novafirst is having a temper tantrum |
| 07:02.10 | JT | [shodan]: unsure, someone is having a big cry when consultancy was mentioned, like it was forbidden |
| 07:02.45 | adorah | <crCernier>I know that chipset very well but there the pstn is only for lifeline..I'm not sure u can do it like u can with Sipura3000 |
| 07:02.48 | novafirst | blah blah, |
| 07:03.09 | Strom_C | Snake-Eyes: who is making the decision? |
| 07:03.10 | sx-wks | ok. the card appears to work :D |
| 07:03.14 | Strom_C | sx-wks: awesome |
| 07:03.20 | [shodan] | yeah, must that ambiguosity about the word "free" in the english language ;P |
| 07:03.27 | crCernier | adorah: Ok... :-( |
| 07:03.34 | [shodan] | asterisk is free, not free , duh ! ;) |
| 07:03.38 | HaMYaI | hi |
| 07:03.55 | HaMYaI | anyone know how to handle MusicOnHold? |
| 07:04.04 | Strom_C | HaMYaI: what are you trying to do? |
| 07:04.15 | crCernier | adorah: I bought it cause I can upload an IAX2 firmware on it... |
| 07:04.17 | sx-wks | Strom_C: at least, I see a 1 appearing in zttool when I pick up the phone :D |
| 07:04.28 | Strom_C | sx-wks: awesome |
| 07:04.38 | HaMYaI | Strom_C: I tried to start MusicOnHold to play mp3 files |
| 07:04.42 | Strom_C | the tdm400p is a breeze to configure; doesnt surprise me |
| 07:04.47 | novafirst | Strom_C: first of all you didn't jokingly make suggestion .. but that is ok. Alah bless you! |
| 07:04.50 | HaMYaI | Strom_C: and then stop it when I need to |
| 07:04.55 | Snake-Eyes | Strom_C, long story but they want to get rid of the openpbx system and get asterisk, and im trying to figure out how long it will take |
| 07:05.09 | Strom_C | Snake-Eyes: how large is the install? |
| 07:05.16 | HaMYaI | Strom_C: once I started it, it just kept going |
| 07:05.24 | adorah | <crCernier>connecting an ATA to pstn has nothing to do with IAX u can do it with sip based ata as well.. |
| 07:05.37 | Strom_C | novafirst: well that's your problem if you can't detect sarcasm and ridiculousness in a suggestion |
| 07:06.10 | novafirst | whatever ... done with you .. thanks for nothing! |
| 07:06.10 | HaMYaI | Strom_C: MusicOnHold(none), from the wiki, is the way to stop but it just didn't work for me |
| 07:06.19 | Strom_C | HaMYaI: the musiconhold() application, as invoked from the dialplan, just plays music on hold indefinitely |
| 07:06.23 | HaMYaI | Strom_C: I'm using agi for that |
| 07:06.25 | Snake-Eyes | Strom_C, 4 fxs and 12 fxo |
| 07:06.25 | crCernier | adorah: Yes I know... but I prefer IAX2 to connect this ATA to my Asterisk server... |
| 07:06.35 | litage | how do you physically connect analog phones and POTS lines to a TDM2400? the card only has what looks like a scsi or parallel connector... |
| 07:06.54 | Strom_C | litage: it's called an amphenol connector and you use a 66 block or breakout box |
| 07:07.05 | Strom_C | Snake-Eyes: how many stations? |
| 07:07.10 | crCernier | adorah: I also have an AT-320 IP phone, and it's configured with an IAX2 connection on Asterisk |
| 07:07.17 | adorah | <crCernier>I understand that u want to connect a remote pstn to u'r * and worry about nat traversal problems.. |
| 07:07.34 | HaMYaI | Strom_C: with the 'm' in Dial String, MusicOnHold seems to stop when the call is answered |
| 07:07.44 | Snake-Eyes | Strom_C, sry, 4 fxo's |
| 07:07.44 | litage | Strom_C: i find it strange that i couldn't find that information on Digium's site |
| 07:07.50 | crCernier | adorah: mmm, you don't understand ;-) |
| 07:07.53 | Strom_C | HaMYaI: do you understand what the purpose of music on hold is? |
| 07:07.57 | HaMYaI | Strom_C: there should be a way top it |
| 07:08.03 | Snake-Eyes | Strom_C, so 12 stations |
| 07:08.16 | litage | Strom_C: thanks for letting me know what the connector is |
| 07:08.27 | Strom_C | litage: it's assumed that if you're working with that many phone lines, you should already know how to run the wiring |
| 07:08.35 | tasat- | anyone aware of a command or option that will cause asterisk to loop an audio file (without interruption)? |
| 07:08.37 | adorah | <crCernier>using sip in local network is actually better with *..I use IAX only for remote ext. |
| 07:09.00 | Strom_C | Snake-Eyes: I could set that up in a day, maybe two if there were some particularly tricky requirements from the client |
| 07:09.04 | HaMYaI | Strom_C: to play music during the call isn't answered I guess |
| 07:09.09 | Strom_C | tasat-: playback() followed by goto()? |
| 07:09.27 | Strom_C | HaMYaI: figure out what you want to use music on hold for and get back to me |
| 07:09.32 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 07:10.06 | tasat- | Strom_C: I should try that... just figured there would be a short break... |
| 07:10.12 | Strom_C | tasat-: nope |
| 07:10.17 | Snake-Eyes | Strom_C, thanks |
| 07:10.26 | [shodan] | damnit :( why are all tftpd server more painful to use than sticking forks in my eyes ! |
| 07:10.44 | Strom_C | [shodan]: atftpd is a no-brainer under debian |
| 07:10.50 | crCernier | adorah: Ok, so if you want that your family "living remotely from you" can connect as easy as possible on your Asterisk, you'll make an IAX2 connection, right ? |
| 07:10.50 | HaMYaI | Strom_C: I just need a way to play music while I am checking status of a file whether it exists |
| 07:11.00 | *** join/#asterisk tengulre (n=tengulre@221.11.5.180) |
| 07:11.13 | HaMYaI | Strom_C: if the file exists I will stop playing music, this is what I really need |
| 07:11.22 | Strom_C | HaMYaI: that process should only take a fraction of a second |
| 07:11.42 | tengulre | hi,all |
| 07:11.53 | tessier | Anyone know how to make the snom phone continue with dialtone after pressing 9 for an "outside line"? I know how with Cisco but haven't figured out Snom yet... |
| 07:12.01 | [shodan] | just got the weirdest thing with aftpd , I make my /tftpboot folder , owned by nobody:nobody a+rwx , I run atftpd no arguments, it give me the help page ! |
| 07:12.03 | tengulre | how to driver dtmf tone with h323 in asterisk? |
| 07:12.04 | tessier | Gotta be some special thing to put in the dialplan string. |
| 07:12.27 | adorah | <crCernier>I'll do that probably..but some ata's and last x-lite version make NAT traversal easy using a public stun server..however I won't trust a public STUN for commercial usage.. |
| 07:12.32 | [shodan] | no matter what I do I get the help page, no errors , nothing in syslog .. even with --verbose=7 ! |
| 07:12.50 | Strom_C | [shodan]: start atftpd through init.d or equivalent |
| 07:13.41 | [shodan] | hmm .. it's worth a shot , but I'm sure I got the syntax right :\ that's so weird |
| 07:14.39 | *** join/#asterisk af_ (n=af@ip-170-156.sn1.eutelia.it) |
| 07:15.02 | [shodan] | hmm , it worked ... go figure |
| 07:16.40 | *** join/#asterisk beu (i=beu@freenode/developer/gentoo.developer.beu) |
| 07:16.44 | sx-wks | hah. the phone works :D |
| 07:16.51 | Strom_C | sx-wks: awesome |
| 07:18.04 | sx-wks | neat ;D |
| 07:18.53 | sx-wks | I had to *restart* asterisk though, it wouldn't work with only zap reoad |
| 07:19.08 | Strom_C | sx-wks: yeah, sometimes that can happen |
| 07:19.46 | sx-wks | ok, shower and go to wrok |
| 07:22.58 | *** join/#asterisk fdepp (n=fdepp@c2cpc3.camptocamp.com) |
| 07:26.25 | *** join/#asterisk xtr (i=01928375@S0106000c41ed11e1.vf.shawcable.net) |
| 07:28.25 | *** join/#asterisk marta (n=a@host184-5.pool8711.interbusiness.it) |
| 07:28.34 | marta | hello |
| 07:31.24 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 07:31.36 | phearless | WaitExten |
| 07:31.43 | phearless | I got a question about it |
| 07:31.52 | Strom_C | maybe i have an answer |
| 07:31.55 | phearless | does it allow somebody to make "calls" ? |
| 07:32.13 | phearless | if he dial a full phone number instead of an internal extension |
| 07:32.51 | Strom_C | that depends on whether full numbers can be dialed from that context |
| 07:33.12 | [shodan] | so the phone tries to grab a bootrom.ld and a sip.ld , if I give it one will it download it and overwrite it's software with it ? (so if I give it the wrong thing my phone becomes a 150$CAD brick ?) |
| 07:33.45 | phearless | exten => s,1,Answer() ; exten => s,2,Background(tt-monkeys) ; exten => s,3,WaitExten(5|m) ; exten => s,4,Queue(myqueue) |
| 07:33.53 | phearless | for example with something like this |
| 07:35.15 | Aurs | [shodan]: don't give it the wrong files :) |
| 07:35.34 | *** join/#asterisk joelsolanki (i=joelsola@202.160.161.94) |
| 07:35.47 | Aurs | [shodan]: I'm using the same bootrom.ld and sip.ld for polycom 301, 430, 501 and 601 |
| 07:36.25 | [shodan] | k, so it's all the same phone , just with a different number of buttons ;) |
| 07:37.02 | Aurs | they use the same software, yes |
| 07:37.12 | joelsolanki | Hi all. I am trying to configure DIDX + asterisk for incoming number. but somehow i m not able to understand or some problem is there. can any one recommend any link ? |
| 07:38.20 | [shodan] | hmm what's the mcu in these things anyway ? an arm7 ? |
| 07:38.23 | *** join/#asterisk rob (i=RobertSt@freenode/staff/rob) |
| 07:39.47 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 07:41.38 | *** join/#asterisk E-bola (i=bola@rbii-valhalla.mrseb.co.uk) |
| 07:41.40 | Strom_C | no, but there is a "ha ha, you just voided the warranty!" sticker inside |
| 07:42.11 | sx-wks | there. I have an 'internal' context that can dial outside :D |
| 07:42.22 | *** join/#asterisk hads|home (n=hads@mail.nice.net.nz) |
| 07:43.54 | sx-wks | and ppl dialing in ring the internal phone :D |
| 07:44.11 | sx-wks | I'm all set for what I have to build :D |
| 07:44.22 | [shodan] | oh to hell with the warranty then ;) |
| 07:44.22 | novafirst | Strom_C and JT: Jebacu vam mamicu ako vas ikada u zivotu vidim. Ako imate muda pojavite se picke jedne! bye! |
| 07:46.39 | joelsolanki | any body has configured Asterisk + DIDX.org for incoming ? |
| 07:47.26 | [shodan] | novafirst, are you 8 ? |
| 07:48.14 | novafirst | I am fine. |
| 07:48.33 | Ryushin | I don't understand what is a FXO and a FXS port? |
| 07:48.45 | JT | that was either hungarian or serbian |
| 07:49.07 | joelsolanki | :( |
| 07:49.25 | novafirst | serb |
| 07:50.08 | novafirst | JT, will you appologise ? |
| 07:50.30 | JT | err |
| 07:50.31 | [shodan] | "Fuck you mummy whether you ever from zivotu see. Whether imate wise set in yourself PICK jedne" |
| 07:50.39 | JT | you just told me to f*ck my mother in serbian |
| 07:50.40 | JT | so no |
| 07:51.36 | novafirst | JT, I sugest you to appologise, and I will to! |
| 07:51.48 | JT | after saying something like that? |
| 07:51.57 | [shodan] | novafirst, stop wasting everyone's time and go bomb some russian APC |
| 07:52.04 | novafirst | [shodan]: wrong translation |
| 07:52.19 | JT | then provide an english version |
| 07:52.19 | [shodan] | novafirst, I'm sure it's close enough |
| 07:52.33 | novafirst | [shodan]: not realy |
| 07:52.38 | JT | if it was something nice, you would've said it in english |
| 07:53.23 | novafirst | JT, it was not nice at all ... I am asking you to appologise and I am willing to do the same. |
| 07:54.08 | JT | for what? |
| 07:54.29 | novafirst | JT, ok .. that is you choice. |
| 07:54.34 | [shodan] | novafirst, 1, they have done nothing wrong, 2, this is the internet and no one give a flying **** |
| 07:54.53 | JT | well tell me what you believe requires an appology, and i will consider it |
| 07:55.05 | novafirst | [shodan]: stay away from this. |
| 07:55.06 | Qwell | novafirst, JT: enough |
| 07:55.10 | [shodan] | haha that sounds so mobbish ... "that is your choice, you're the one who's going to have to live with it" |
| 07:55.17 | JT | yeah oh well |
| 07:55.19 | Qwell | [shodan]: ^ |
| 07:55.26 | novafirst | JT, you know well everything that was said. |
| 07:56.14 | JT | as per Qwell's request, no further comment |
| 07:56.30 | novafirst | ok |
| 07:58.13 | jart | please please, you're all acting like a bunch of hitlers |
| 07:58.33 | [shodan] | what? we're not even invading France , please ! |
| 08:01.04 | novafirst | I wana thank you all for a great entertainment. |
| 08:01.09 | [shodan] | is there anything wrong with using the same SIP user on two phones at once ? (as in if someone dial this user , both phones will rings and they can also make calls at the same time) |
| 08:01.31 | [shodan] | you should try youtube.com ! |
| 08:01.39 | brimstone | i'm pretty sure it won't work like that [shodan] |
| 08:01.47 | beu | or qdb.us ;) |
| 08:02.34 | novafirst | [shodan]: are you musilim? you name suggest you are?? |
| 08:02.51 | *** mode/#asterisk [+b %novafirst!*@*] by Qwell |
| 08:02.54 | joelsolanki | anybody has configured sipx with asterisk |
| 08:02.58 | *** join/#asterisk tengulre11 (n=tengulre@221.11.5.180) |
| 08:03.00 | joelsolanki | ? |
| 08:03.10 | [shodan] | does the same thing goes for a two line phone ? I have to give them each a different SIP user ? |
| 08:03.23 | Qwell | beu: freenode needs /shun |
| 08:03.51 | brimstone | [shodan], if the phone supports 2 lines, it should be ok, as it will handle 2 different calls |
| 08:04.36 | beu | Qwell: hehe |
| 08:04.43 | beu | Qwell: your wish may just come true ;) |
| 08:05.05 | *** join/#asterisk Ahrimanes (n=michael@81.7.159.2) |
| 08:05.05 | JT | what are the most lines people have seen on a voip phone? |
| 08:05.05 | Qwell | beu: switching to unreal? :P |
| 08:05.05 | JT | i know softphones regularly have something like 6 lines |
| 08:05.14 | beu | Qwell: nah, new ircd in the works ;) |
| 08:05.16 | [shodan] | oh , I'm asking because you can set a different sip user on each line , I thought that effectively the two line phone was like two separate phones from * point of view |
| 08:05.19 | Qwell | fun |
| 08:05.31 | beu | (and this time it won't be a shitty hack!) |
| 08:06.12 | brimstone | [shodan], 2 calls =/= 2 lines, as lines imply seperate registrations, in my head at least |
| 08:06.32 | brimstone | so it maybe able to handle 2 different registrations as well |
| 08:06.38 | brimstone | with up to 1 call on each line |
| 08:06.52 | brimstone | "up to 1" ? |
| 08:07.22 | [shodan] | from the logs , it's trying to register once per line .. hmm |
| 08:07.37 | brimstone | one way audio must be half a call HA! |
| 08:08.19 | *** join/#asterisk darkskiez (n=mbryars@194.247.78.146) |
| 08:09.07 | *** join/#asterisk Muck- (n=Muck@145.253.170.162) |
| 08:09.58 | Muck- | hi :) |
| 08:10.28 | brimstone | hi Muck-! |
| 08:10.36 | tessier | hmmm |
| 08:10.57 | tessier | If a user does not enter anything into the IVR I want them to go into our customer service queue. So why doesn't the timeout extension with a goto do this? |
| 08:10.57 | [shodan] | it's working, oh joy !!! :) |
| 08:11.04 | tessier | exten => t,1,Goto(myqueue,s,1) |
| 08:11.10 | [shodan] | ~ivr |
| 08:11.13 | jbot | rumour has it, ivr is Interactive Voice Response |
| 08:11.28 | tessier | It just auto falls through and I get hung up on. |
| 08:11.36 | [shodan] | hmm should've thought of that before |
| 08:11.36 | tessier | I know this behavior has changed in 1.2 |
| 08:11.39 | [shodan] | ~did |
| 08:11.40 | jbot | somebody said did was Direct Inward Dialing |
| 08:11.40 | brimstone | did you set a timeout? |
| 08:11.41 | tessier | Last asterisk I used was 1.0 or 1.1 or something |
| 08:11.48 | tessier | exten => s,5,Set(TIMEOUT(digit)=3) |
| 08:11.50 | *** part/#asterisk joelsolanki (i=joelsola@202.160.161.94) |
| 08:11.53 | tessier | Yes, I set a timeout |
| 08:12.01 | brimstone | are you using a waitexten? |
| 08:12.04 | brimstone | or how? |
| 08:12.09 | tessier | A waitexten? |
| 08:12.18 | brimstone | after the prompt, how do you get user input? |
| 08:12.36 | tessier | ahh.....No, I am not. I forgot about that. |
| 08:15.47 | *** join/#asterisk r_marvin (n=r_marvin@83.166.220.142) |
| 08:16.35 | r_marvin | any idea if i can somehow set he CLIR flag on chan_h323? i've only seen references to chan_capi so far |
| 08:16.41 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 08:17.32 | Ahrimanes | any ideas when 1.4 will be out ? |
| 08:18.20 | brimstone | isn't that the magic question Ahrimanes ? |
| 08:18.35 | tessier | brimstone: hmm....I think that did work. It now tries to join the queue. But now I seem to have something else messed up: Aug 31 01:18:57 WARNING[21952]: app_queue.c:3221 queue_exec: Unable to join queue 'myqueue' |
| 08:18.51 | brimstone | tessier, do you have such a queue? |
| 08:18.55 | tessier | This used to work before but I think I have renamed and reorganized some things. There is definitely a queue called myqueue defined... |
| 08:19.00 | Ahrimanes | brimstone: i guess, but there must be a roadmap somewhere? |
| 08:19.12 | tessier | It is listed in show queues |
| 08:19.18 | brimstone | Ahrimanes, maybe one of the ops knows, but not i |
| 08:19.36 | Ahrimanes | brimstone: ok |
| 08:19.58 | brimstone | tessier, i'd have to see those bits of your dialplan, but it sounds like it is jumping to [myqueue] |
| 08:20.23 | tessier | brimstone: You mean the context as opposed to the queue? |
| 08:20.48 | brimstone | aye, everything sounds right otherwise |
| 08:21.42 | *** join/#asterisk sxpert (n=sxpert@raph.imag.fr) |
| 08:22.21 | *** part/#asterisk sxpert (n=sxpert@raph.imag.fr) |
| 08:22.25 | tessier | The IVR does a Goto(myqueue,s,1) and I have a context [myqueue] which answers the call and does exten => s,4,Queue(queue) |
| 08:22.30 | tessier | I have a queue named queue |
| 08:23.04 | tessier | Probably not. |
| 08:23.20 | brimstone | try something else, like PotatoPants |
| 08:24.23 | Faithful | Is the headset jack on a SPA941 2.5mm stereo (3 pole) or 4 pole? |
| 08:24.28 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
| 08:24.28 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 08:24.41 | brimstone | wb Qwell |
| 08:24.44 | Qwell | Note to self: alt-q != alt-w |
| 08:24.44 | *** join/#asterisk sxpert (n=sxpert@raph.imag.fr) |
| 08:24.52 | sxpert | lol |
| 08:27.56 | Ahrimanes | Qwell: learning something new everyday eh? |
| 08:31.44 | RoyK | -w |
| 08:32.15 | Ahrimanes | is it emacs or some other arcane evil like that? |
| 08:32.27 | tessier | http://de.pastebin.ca/42170 |
| 08:32.36 | tessier | Looks like only one other person has ever had this problem |
| 08:32.40 | tessier | Weird. I always get the weird ones. |
| 08:32.56 | Ahrimanes | i get that with girls :( |
| 08:33.03 | tessier | Asterisk has always been pretty bad with useful error messages. |
| 08:33.16 | tessier | Just telling me it cannot join the queue isn't very useful. WHY can't it join it? |
| 08:33.48 | Ahrimanes | tessier: feel free to add more debugging code to app_queue.c :) |
| 08:34.23 | *** join/#asterisk ghenry (n=ghenry@82-69-192-46.dsl.in-addr.zen.co.uk) |
| 08:34.51 | RoyK | emacs is evil. vim is god |
| 08:35.24 | Aurs | vim for teh win |
| 08:36.35 | X-Rob | EMACS == Emacs Makes A Computer Slow |
| 08:36.51 | Ahrimanes | X-Rob: :) |
| 08:37.10 | X-Rob | Escape-Meta-Alt-Control-Shift is the other one I know |
| 08:37.54 | X-Rob | http://www.gnu.org/fun/jokes/gnuemacs.acro.exp.html |
| 08:37.58 | X-Rob | w00t google. |
| 08:38.22 | tessier | Eventually Malloc()s All Computer Storage |
| 08:38.31 | tessier | Eighty Megs And Constantly Swapping |
| 08:38.40 | tessier | EMACS May Allow Customized Screwups |
| 08:38.55 | *** join/#asterisk inspired (n=mikael@85.221.7.59) |
| 08:39.03 | *** join/#asterisk HaMYaI (i=HaMYaI@202.8.86.164) |
| 08:40.03 | *** join/#asterisk ma_dzen (n=ma_dzen@217.66.17.141) |
| 08:44.10 | foxmjay | plz , how to use the CHANNEL STATUS agi command in php , i did : fwrite($stdout,"CHANNEL STATUS ZAP/1 \n"); , but it's gives me alwayse the result -1 |
| 08:45.43 | sxpert | that's the status ;D |
| 08:45.50 | foxmjay | :) |
| 08:45.58 | sxpert | besides ZAP/1 adresses a card. |
| 08:45.59 | tessier | ah-HAH! |
| 08:46.09 | sxpert | ZAP/1-1 would address the channel |
| 08:46.28 | tessier | I needed to say member => Agent/1001 in queues.conf |
| 08:49.35 | tessier | Yes! It works perfectly. |
| 08:49.48 | tessier | Tomorrow I demo the system for the boss. Then for the head of customer service. Then it's into production we go. |
| 08:50.14 | sxpert | RoyK: heya |
| 08:50.25 | sxpert | I can now zap myself ;D |
| 08:50.47 | foxmjay | but the status of ZAP/1 is buys , so i must have the number 7 as a result |
| 08:51.06 | sxpert | tessier: be careful... test it again *right* *before* going to the boss, just in case the demo effect wants to rear it's ugly head ;D |
| 08:51.12 | tessier | yeah |
| 08:51.24 | sxpert | foxmjay: that should be ZAP/1-1 or something |
| 08:51.32 | tessier | This is the best phone system I have ever made. |
| 08:51.44 | tessier | Probably my 10th asterisk install. But the first one in year and a half or so. |
| 08:51.56 | *** join/#asterisk vistacdd (n=vistac@211.22.77.218) |
| 08:51.58 | sxpert | tessier: heh |
| 08:52.08 | sxpert | tessier: the rest were toy installs ? |
| 08:52.08 | *** join/#asterisk b0gka (n=sunny@216-70-245-253.static-ip.telepacific.net) |
| 08:52.12 | vistacdd | Hello Everybody ... |
| 08:52.20 | tessier | sxpert: No, a few were production installs |
| 08:52.21 | foxmjay | sxpert: what's meen 1-1 :) |
| 08:52.21 | b0gka | Early morning 2 all |
| 08:52.34 | foxmjay | sxpert: mean :) |
| 08:52.45 | sxpert | foxmjay: the first one is the card, the second one is the channel on the card |
| 08:52.48 | tessier | In fact one has been running for 2 years and in all that time I only got one support call on it a month ago when the mediatrix box that handles the POTS lines somehow got wedged. |
| 08:52.56 | sxpert | foxmjay: just like in extensions.conf |
| 08:53.42 | sxpert | tessier: mediatrix ? |
| 08:53.49 | tessier | yeah |
| 08:53.55 | sxpert | tessier: link ? |
| 08:53.56 | tessier | You can google them. They make nice POTS interface equipment |
| 08:53.58 | b0gka | wondering if anyone has Adtran/Digium 110P experience? |
| 08:54.09 | tessier | I don't touch digiums POTS stuff. |
| 08:54.18 | tessier | I got burned on it and I am still annoyed. |
| 08:54.19 | foxmjay | sxpert: but i have just one card, ok i'll try |
| 08:54.41 | b0gka | well this may be more a cable/wiring question |
| 08:54.48 | b0gka | rather than vendor specific |
| 08:55.07 | sxpert | foxmjay: it doesn't make any difference ;D |
| 08:56.04 | foxmjay | sxpert: :) thanka thanka , it's work now,owwhoo, :D |
| 08:56.05 | b0gka | basically have an integrated T1 which is fed to an Adtran device which splits the data and voice components [dsx-1] |
| 08:56.27 | sxpert | foxmjay: congrats |
| 08:56.41 | b0gka | does any one know if a crossover T1 cable is required to connect dsx-1 from adtran to the digium card? |
| 08:56.51 | sxpert | tessier: how so ? |
| 08:56.53 | *** join/#asterisk Sonderblade (n=mah@static-213.131.147.169.addr.tdcsong.se) |
| 08:57.08 | sxpert | tessier: that tdm31b I got this morning appears to work perfectly |
| 08:57.16 | bhrobinson | I believe that you do, yes |
| 08:57.22 | foxmjay | sxpert: :) |
| 08:57.22 | tessier | sxpert: The things failed a lot, the support was a pain, it cost me a ton of money and put the project way behind |
| 08:57.38 | bhrobinson | b0gka, i think it does... |
| 08:57.43 | tessier | This has been the fastest I have ever rolled out a pbx with lots of functionality. |
| 08:57.44 | b0gka | bhr : thanks for replying |
| 08:58.04 | sxpert | tessier: hah... interesting |
| 08:58.05 | b0gka | bhr : if I simply connect the T1 drop into digium card everything works fine |
| 08:58.24 | sxpert | tessier: what hardware do you use ? |
| 08:58.33 | b0gka | bhr : but dsx-1 [after adtran splits data/voice] does not give a green light on digium |
| 08:58.39 | b0gka | so makes sense |
| 08:58.43 | b0gka | need T1 crossover |
| 08:58.53 | tessier | sxpert: For PSTN? I currently use a Sangoma card with a PRI |
| 08:59.03 | bhrobinson | b0gka, you have configured the zaptel.conf, right? |
| 08:59.11 | vistacdd | Does any know how to detect PHONE busy in extensions.conf? |
| 08:59.19 | bhrobinson | I know that is a stupid... |
| 08:59.20 | b0gka | yes |
| 08:59.25 | b0gka | zaptel works just fine |
| 08:59.26 | tessier | vistacdd: Look at macro-stdexten in the demo |
| 08:59.43 | tessier | er...maybe not in the demo context but in the example extensions.conf |
| 08:59.51 | sxpert | tessier: ha, ok |
| 09:00.06 | b0gka | bhrobinson: I can place outbound calls and get inbound calls if I remove adtran completely :) |
| 09:00.45 | bhrobinson | b0gka, so the adtran has the passthru? is it a 624 or a 608? |
| 09:00.46 | b0gka | but the T1 provider insists we split data/voice using their equipment [Adtran something] which helps them better diagnose/troubleshoot problems |
| 09:00.57 | b0gka | its a 608 |
| 09:01.05 | vistacdd | I read the O'Relly Book, it say, if the phone is busy, the priority will jump to Dial+101? |
| 09:01.32 | bhrobinson | b0gka, I thought the 608 only terminated analog... I think you need the 624 for passthru |
| 09:01.33 | e-ddie | does anyone know of a solution, with Dial(Local/blah...) with video phones? |
| 09:01.34 | vistacdd | I tried it, but it not jump to +101, just hangup it = = |
| 09:01.43 | b0gka | Hrmmmmmmmmmmmmmmmmmm |
| 09:01.55 | e-ddie | like, if i use it with Dial(SIP/...) video comes through, but not with Local/ |
| 09:02.00 | b0gka | it does have the dsx-1 module |
| 09:02.01 | sxpert | tessier: I'll be using digium stuff, we'll see how that goes... won't cost us more than the shit ass dialogic crap with that funky n-soft "Rekoll" software we're currently using (on NT4 because it doesn't want to hear about anything else) |
| 09:02.30 | tessier | sxpert: Running a phone system on any kind of windows just pure insanity as far as I am concerned |
| 09:02.32 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
| 09:02.40 | bhrobinson | b0gka, are you stateside? |
| 09:02.49 | b0gka | stateside/ |
| 09:02.53 | b0gka | not sure what stateside means |
| 09:02.58 | Doctor | USA ra ra ra |
| 09:03.03 | b0gka | oh |
| 09:03.04 | b0gka | hehehe |
| 09:03.06 | b0gka | yes I am in States |
| 09:03.15 | bhrobinson | who is your provider? |
| 09:03.47 | sxpert | tessier: heh.. well. that thing has been able to make the company 50K€ per month so far, so, it ain't so bad. but we could make 100K or so by getting rid of the yearly license fees ;D |
| 09:05.32 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 09:06.12 | Doctor | heh does the license fees get you support? |
| 09:06.30 | sxpert | Doctor: not really... |
| 09:06.50 | sxpert | and the thing is horrible to use, hasn't been patched in 5 years or so... you get the idea ;D |
| 09:07.44 | vistacdd | tessire: could u teach me how to detect phone busy? I have already follow the sample extensions.conf to set a phone number, and set a plus 101 priority in this number, but when this phone is busy, seems not work to play busy notice voice. |
| 09:10.31 | *** join/#asterisk spr1te (i=spr1te@194.187.130.227) |
| 09:11.07 | [shodan] | damn this web ui is atrocious , where do you even put you sip username in that (trying to setup a linksys spa-2102) |
| 09:11.30 | vistacdd | exten => 222,1,Dial(SIP/user1,10) |
| 09:11.31 | vistacdd | exten => 222,2,Playback(vm-nobodyavail) |
| 09:11.31 | vistacdd | exten => 222,3,Hangup() |
| 09:11.31 | vistacdd | exten => 222,102,Playback(tt-allbusy) |
| 09:11.31 | vistacdd | exten => 222,103,Hnagup() |
| 09:11.47 | vistacdd | this is my dial play for extension 222 |
| 09:11.48 | RoyK | ~pb |
| 09:11.58 | jbot | well, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/, or http://www.paste-it.net/ |
| 09:12.14 | vistacdd | when 222 is busy, and I call 222 on another phone |
| 09:12.17 | sxpert | vistacdd: you have a typo at line 103 |
| 09:12.40 | vistacdd | it will play vm-nobydyavail, not tt-allbusy |
| 09:12.59 | sxpert | there's no priority jumping anymore |
| 09:13.04 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 09:13.05 | sxpert | it's deprecated |
| 09:13.38 | sxpert | vistacdd: use the DIALSTATUS variable to know what happened to your call, and use gotoif |
| 09:14.24 | vistacdd | ohh, I see |
| 09:14.34 | vistacdd | thanks for ur advise .... |
| 09:19.25 | *** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 09:21.29 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 09:22.25 | *** part/#asterisk kumamoto (n=eryco@24-178-2-212.dhcp.ftwo.tx.charter.com) |
| 09:24.29 | *** join/#asterisk kumamoto (n=eryco@24-178-2-212.dhcp.ftwo.tx.charter.com) |
| 09:25.13 | MrChimpy | morning |
| 09:25.32 | kumamoto | morning all |
| 09:25.53 | phearless | morning * |
| 09:27.09 | MrChimpy | hmm, asterisk-uk site seems broken |
| 09:27.47 | [shodan] | ok , the relieve is here, time to go to bed |
| 09:28.18 | sxpert | [shodan]: lol |
| 09:33.47 | phearless | [root@ouistiti:~]# /usr/src/zaptel-1.2.7/fxotune -i 4 |
| 09:33.48 | phearless | Tuning module 1 |
| 09:33.55 | phearless | I have to wait how many munutes like this ? |
| 09:35.09 | phearless | minutes* |
| 09:36.30 | X-Rob | phearless, a lot, and it's useless if you're not using zaptel trunk. |
| 09:37.15 | phearless | I use zaptel |
| 09:37.27 | phearless | with a TDM400P + one FXO module |
| 09:37.45 | X-Rob | the zaptel svn trunk, as opposed to 1.2 |
| 09:38.38 | sxpert | X-Rob: what is that supposed to do ? |
| 09:38.48 | X-Rob | fxotune? |
| 09:38.59 | phearless | waht does mean "the zaptel svn trunk, as opposed to 1.2" ?! |
| 09:39.20 | X-Rob | phearless, "svn co http://svn.digium.com/svn/zaptel/trunk /usr/src/zaptel" |
| 09:39.37 | phearless | ah okay |
| 09:39.55 | phearless | I need to remove the echo |
| 09:40.03 | phearless | and I heard that I can use fxotune |
| 09:40.10 | phearless | am I wrong ? |
| 09:40.21 | X-Rob | use zaptel trunk to start with. that has better echo cancellation |
| 09:40.27 | X-Rob | do you have a TDM400 card? |
| 09:40.44 | phearless | a TDM400P + one FXO module |
| 09:40.53 | phearless | zapata.conf:echocancelwhenbridged=yes |
| 09:40.53 | phearless | zapata.conf:echocancel=yes |
| 09:40.53 | phearless | zapata.conf:echotraining=800 |
| 09:40.58 | sxpert | X-Rob: yeah. what is fxotune supposed to do ? |
| 09:41.00 | phearless | I got this in my config' |
| 09:41.11 | phearless | sxpert: http://www.voip-info.org/wiki/view/Asterisk+fxotune |
| 09:41.22 | phearless | sxpert: sort of auto-testing to remove the echo |
| 09:41.23 | X-Rob | phearless, use zaptel trunk |
| 09:41.39 | sxpert | phearless: hah. oddly enough it was in the wiki ;D |
| 09:41.56 | X-Rob | sxpert, it measures the impedance of your phone line and adjusts the hybrid on the fxo card |
| 09:42.08 | phearless | X-Rob: I must use the svn version to remove the echo ? |
| 09:42.28 | X-Rob | phearless, try that first before messing with fxotune. Zaptel-trunk usually fixes most echo problems |
| 09:44.46 | phearless | ok let's go |
| 09:46.08 | *** join/#asterisk X-Gen (n=X-Gen@dsl-145-210-130.telkomadsl.co.za) |
| 09:46.36 | sumasuma | which is the best linux flavour to enjoy like windows ? |
| 09:48.46 | phearless | sumasuma: ask this in #troll or ##linux |
| 09:50.19 | RoyK | sumasuma: not really related to asterisk... |
| 09:52.18 | darkskiez | I'd go with raspberry |
| 09:53.19 | sxpert | phearless: I'd say only #troll would be useful in that case ;D |
| 09:53.26 | RoyK | hm. blueberries in the woods now. |
| 09:53.31 | RoyK | gotta get out there soon |
| 09:54.03 | phearless | X-Rob: I finished the install of zaptel ... I reboot ... |
| 09:54.31 | RoyK | why reboot? |
| 09:54.38 | X-Rob | because he loves windows. |
| 09:56.31 | sxpert | lol |
| 09:56.36 | phearless | because I am not sure about how zaptel works |
| 09:56.52 | phearless | I was expecting a "why reboot?" comment |
| 09:57.12 | *** join/#asterisk Nobbie (n=no@fwb003.fw.is.co.za) |
| 09:57.17 | Nobbie | heya =) |
| 09:58.20 | foxmjay | sxpert: hi its me again :) , are you still there |
| 09:58.50 | sxpert | foxmjay: yeah |
| 09:59.50 | foxmjay | sxpert: how can i get the result in a variable, the fwrite function return just the number of caracters have been written |
| 10:00.08 | sxpert | erh ? |
| 10:00.13 | sxpert | fread ? |
| 10:00.15 | foxmjay | lol |
| 10:00.39 | foxmjay | fread($stdin,$result) ? |
| 10:01.33 | sxpert | dunno what language you're using, but that looks about right |
| 10:02.22 | foxmjay | php , i tried fgets($stdin); but it gives me an other string (agi_request) |
| 10:03.04 | sxpert | well, you'd have to read stuff from stdin I guess. |
| 10:03.15 | sxpert | I don't really know, I haven't done any AGI yet |
| 10:03.39 | foxmjay | sxpert: ok tnx i'll try again, i'll come back again muuhaa :D |
| 10:03.52 | *** join/#asterisk BeBoo (n=BeBoo@213.237.44.34) |
| 10:06.05 | BeBoo | Hey. I have 2 SIP clients connected to an asterisk server, working fine. One of the sip clients supports 2 calls/lines at the same time, but if i'm making a call on that client, and another guy calls me, it only rings on the other client. The sip client is X-Lite - is there a way to make it work with 2 incoming calls at the same time? |
| 10:07.03 | sxpert | guess you can only make one call at a time per client ;D |
| 10:07.17 | sxpert | (after all, there's only 1 sound card in the box ;D |
| 10:09.39 | r_marvin | any idea why ${MATH(${EPOCH}%11)} returns the same value at least three seconds in a row? |
| 10:10.00 | BeBoo | No, it's connected to an ISDN30 line, so making 3 calls at the same time wont be a problem :p |
| 10:10.08 | r_marvin | (i'm using 1.2.10, and i'm trying to emulate $RAND without agi) |
| 10:10.18 | BeBoo | It's just like Asterisk doesn't tell the client thats it's ringing, when already in use |
| 10:14.51 | *** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135) |
| 10:18.04 | *** join/#asterisk lunaphyte (n=lunaphyt@static-71-120-128-10.gdrpmi.dsl-w.verizon.net) |
| 10:18.10 | JunK-Y | r_marvin: why not use rand in ur programming language? |
| 10:19.08 | JunK-Y | or use func_rand.c in trunk and just tweak it to be able to use it in ur version |
| 10:20.32 | r_marvin | JunK-Y: i'm trying to do it in the dialplan |
| 10:20.48 | r_marvin | well, i found different (and better) method |
| 10:21.02 | JunK-Y | just tweak func_rand.c and use RAND() in ur dp? |
| 10:21.04 | r_marvin | but it's still weird |
| 10:21.49 | JunK-Y | ${MATH(${EPOCH}%11)} is really inneficient, since all sim calls per sec will have the same result. |
| 10:22.02 | r_marvin | i'd rather refrain from recompiling asterisk, so i hacked something with globalvars |
| 10:22.08 | JunK-Y | lets say, you could have the same result for 50 calls. |
| 10:22.17 | r_marvin | JunK-Y: i know, and it was acceptable |
| 10:22.28 | MrChimpy | r_marvin: do something with the unique call id |
| 10:22.37 | r_marvin | but caching the result was not acceptable |
| 10:22.44 | MrChimpy | dunno if dialplan can manage doing checksums |
| 10:22.59 | r_marvin | MrChimpy: i considered it, but it was overkill |
| 10:23.00 | MrChimpy | then munge that with seconds |
| 10:23.16 | MrChimpy | seconds/clock |
| 10:23.29 | r_marvin | i just incremented a globalvar |
| 10:23.30 | MrChimpy | i'm not really processing english this morning yet. too early. |
| 10:23.37 | MrChimpy | fair enough |
| 10:23.47 | r_marvin | i just wanted to signal a possible bug |
| 10:24.03 | MrChimpy | isn't there a RAND thing in dialplan at all? |
| 10:24.16 | JunK-Y | MrChimpy: just in trunk |
| 10:24.25 | MrChimpy | ah, I see. that old chestnut ;) |
| 10:24.37 | *** join/#asterisk asterisk_noob (i=user@host-82-135-126-82.customer.m-online.net) |
| 10:24.54 | JunK-Y | actually, its the opposite, its too-new chestnut :) |
| 10:24.59 | MrChimpy | hehe |
| 10:25.17 | MrChimpy | i look forward to 1.4 and everything i've done breaking irrevocably |
| 10:25.54 | JunK-Y | r_marvin: just be aware, that RAND() will be available in 1.4. |
| 10:26.47 | Ahrimanes | hm 1.4 scheduled for the beginning of july.. that's not too far away..? |
| 10:26.51 | sheadow | hi, which command can i usw: i like to ring a telphone, after 10 sec a second telfon should ring together with the first? |
| 10:27.13 | JunK-Y | Ahrimanes: actually, it was past july. |
| 10:27.27 | Nobbie | any suggestions on how to bypass the Authenticate() in a dialplan when a call is transferred to a number which requires Auth() ? |
| 10:27.32 | Ahrimanes | JunK-Y: hehe.. well i just hope it's soon |
| 10:27.55 | JunK-Y | Nobbie: set a var and use a gotoif |
| 10:28.24 | JunK-Y | sheadow: see application dial with options. |
| 10:29.17 | Nobbie | Junk: i can set a var, but how will i identify when it's a transfer as opposed to a new call ? |
| 10:29.27 | BeBoo | Hmm, does Asterisk support two SIP clients connected at the same time? |
| 10:29.39 | Nobbie | BeBoo: you mean on the same extension number ? |
| 10:29.50 | sheadow | ok |
| 10:29.53 | sheadow | thx |
| 10:30.04 | BeBoo | Nobbie, yup |
| 10:30.49 | Nobbie | BeBoo: not as far as i know. outbound calls from the SIP clinet will work, but inbound to the SIP client will only go to the one which is currently registered |
| 10:31.15 | BeBoo | hmm damn :( |
| 10:31.25 | Nobbie | BeBoo: setup a ring group or queue |
| 10:31.39 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 10:32.48 | BeBoo | ok thanks :) just gonna find out how to do that :p |
| 10:34.16 | BeBoo | If using a group, can i make it call both phones at the same time? |
| 10:34.40 | BeBoo | Or will it switch between the two clients specified? |
| 10:36.24 | *** join/#asterisk fnordus (n=dnall@s142-179-111-243.bc.hsia.telus.net) |
| 10:43.54 | *** join/#asterisk Adrian__ (n=foo@zux221-214-183.adsl.green.ch) |
| 10:44.51 | phearless | X-Rob :( |
| 10:45.15 | phearless | I still got an echo with the svn trunk version of zaptel |
| 10:45.16 | JunK-Y | BeBoo: u can do that with queues. |
| 10:52.25 | phearless | hello ! |
| 10:52.26 | *** join/#asterisk CSerpent (i=daemon@62.49.253.91) |
| 10:52.30 | phearless | I got a damn echo. |
| 10:52.32 | CSerpent | morning all |
| 10:53.06 | phearless | this is really annoying (TDM400P+FXOmodule). when I call outside, only me got the echo. |
| 10:53.07 | phearless | this is really annoying (TDM400P+FXOmodule). when I call outside, only me got the echo. |
| 10:53.12 | *** join/#asterisk dorel__ (n=liran@212.199.9.246.static.012.net.il) |
| 10:54.06 | MrChimpy | want no echo? me suggest move out of cave. |
| 10:54.50 | CSerpent | lmao |
| 10:55.07 | CSerpent | you got echo cancellation enabled in zapata.conf? |
| 10:56.45 | phearless | yes CSerpent ! |
| 10:57.02 | phearless | zapata.conf:echocancelwhenbridged=yes |
| 10:57.02 | phearless | zapata.conf:echocancel=yes |
| 10:57.02 | phearless | zapata.conf:echotraining=800 |
| 10:59.27 | BeBoo | JunK-Y, okay thanks :) |
| 10:59.51 | BeBoo | I dont suppose there's a tutorial for this? Haven't been able to locate one |
| 11:01.17 | RoyK | ~book |
| 11:01.24 | jbot | [book] a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 11:01.41 | MrChimpy | the book is great |
| 11:02.12 | MrChimpy | but doesn't cover E1 stuff. I think it assumes you're manly enough to google if you're using carrier lines ;) |
| 11:02.32 | phearless | CSerpent: better idea ? |
| 11:06.34 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 11:06.34 | *** mode/#asterisk [+o anthm] by ChanServ |
| 11:07.04 | *** join/#asterisk yatesy (n=yatesy@unaffiliated/yatesy) |
| 11:09.23 | *** join/#asterisk rene- (n=rene1@gea-gye-internet.telconet.net) |
| 11:09.26 | *** join/#asterisk shortguy_ (i=shortguy@panoramix.divadeli.nl) |
| 11:09.26 | rene- | hello |
| 11:10.09 | rene- | in order to avoid sync issues do i want unmaskirq set to 1 or to 0 |
| 11:10.10 | sheadow | hi |
| 11:10.19 | *** part/#asterisk sheadow (i=user@host-82-135-126-82.customer.m-online.net) |
| 11:11.36 | CSerpent | phearless: what ver of zaptel? |
| 11:14.16 | *** join/#asterisk zotz (n=zotz@24.244.163.225) |
| 11:16.43 | phearless | from svn |
| 11:17.01 | phearless | SVN-trunk-r1374 |
| 11:17.09 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 11:17.29 | puzzled | morning |
| 11:17.55 | phearless | any echo guru around ? |
| 11:18.04 | phearless | good morning puzzled |
| 11:18.46 | puzzled | phearless: iirc asteriskguru.com has some good info |
| 11:19.26 | *** join/#asterisk bmg505 (n=leon@dsl-146-48-253.telkomadsl.co.za) |
| 11:19.31 | phearless | yes but not about echo |
| 11:20.41 | tzafrir | phearless, not exactly a guru. But it is always better to ask your actual Q than to ask for an expert |
| 11:21.06 | phearless | I asked several times my question |
| 11:21.28 | phearless | it is : TDM400P+FXOmodule => when I call outside, I got an echo. |
| 11:21.29 | tzafrir | Have you set the proper opermode? |
| 11:21.33 | phearless | yes |
| 11:21.43 | BeBoo | Is it possible to have a queue without music, just default ringing tone? |
| 11:22.11 | phearless | modprobe.conf |
| 11:22.11 | phearless | options wctdm opermode=UK |
| 11:22.13 | tzafrir | and used fxotune? and actually bother to load the parameters it sets at boot time? |
| 11:23.02 | phearless | [root@ouistiti:~]# /usr/src/zaptel/fxotune -i 4 |
| 11:23.02 | phearless | Tuning module /dev/zap/1 |
| 11:23.16 | phearless | [then I wait...] |
| 11:23.32 | tzafrir | you should look at /etc/fxotune.conf |
| 11:23.44 | tzafrir | (after it's done) |
| 11:23.53 | tzafrir | What version of zaptel, BTW? |
| 11:24.18 | phearless | the svn version |
| 11:24.26 | tzafrir | If not latest, try grabbing fxo.c from latest zaptel (at least 1.2.6, maybe there were some later improvements) |
| 11:24.28 | phearless | dowloaded 10mn ago |
| 11:24.43 | phearless | SVN-trunk-r1374 |
| 11:25.35 | phearless | cat /etc/fxotune.conf |
| 11:25.35 | phearless | 1=8,253,2,244,255,10,244,3,253 |
| 11:25.46 | phearless | ( fxotune has finished) |
| 11:26.31 | phearless | then I do `fxotune -s` ? |
| 11:32.39 | phearless | hum? |
| 11:32.44 | phearless | I will try |
| 11:32.56 | phearless | if I break everything I will blame #asterisk :-] |
| 11:33.24 | phearless | [root@ouistiti:~]# /usr/src/zaptel/fxotune -s |
| 11:33.24 | phearless | fxotune: successfully set echo coeffecients on FXO modules |
| 11:33.27 | phearless | okay. |
| 11:34.00 | *** join/#asterisk niter3 (n=niter3@d57-102-239.home.cgocable.net) |
| 11:42.22 | CSerpent | BeBoo: you could just download an mp3 of a ringing tone and stick that in the queue files? :) |
| 11:44.46 | phearless | SAME ECHO ! |
| 11:49.39 | CSerpent | phearless: you read http://www.voip-info.org/wiki/view/Asterisk+fxotune ? Might just provide an idea |
| 11:50.24 | phearless | this is the first thing that I have read |
| 11:54.23 | *** join/#asterisk oej (n=oej@cC3012269.inet.catch.no) |
| 11:55.29 | *** join/#asterisk L-info (n=Adam@62.69.102.99) |
| 11:56.18 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 11:59.54 | dorel__ | i've defined group=0 for channels => 1-4 (its an analog fxo card) |
| 12:00.03 | dorel__ | in incoming calls i see Zap/1-1 |
| 12:00.34 | dorel__ | so thats ok that the trunk is configured a g0, which contains inside it Zap/1-1, Zap/2-1, Zap/3-1 and Zap4-1 right? |
| 12:00.40 | dorel__ | s/a/s |
| 12:00.45 | dorel__ | s/a/as |
| 12:00.48 | macTijn | dorel: uhuh |
| 12:01.11 | [TK]D-Fender | dorel__: : Zap/1, Zap/2, Zap/3, Zap/4 |
| 12:01.12 | macTijn | you dial over it as Dial(Zap/g0/<number>) |
| 12:02.17 | dorel__ | so it's like i said, the g0 takes care of it |
| 12:02.42 | dorel__ | because they're all defined under group=0 then its enough to specify g0 and the group takes care of non-busy lines to receive/dial out calls |
| 12:04.04 | [TK]D-Fender | dorel__: is something in your setup not working or was that just a "to be sure" question? |
| 12:05.36 | dorel__ | it wasnt working but magically it does now |
| 12:05.43 | dorel__ | people at #freepbx call it zapmagic hehe :) |
| 12:06.50 | niter3 | Hey guys, I got a bit of a issue. I have asterisk behind a NAT. I have all the RTP and local bind port set to the asterisk box. People outside on SIP extensions can call the asterisk box and listen to messages, leave messages in mail boxes and such. However when I was trying to communicate with somebody outside the NAT and myself inside the NAT I could not hear the person and neither ccould the other person hear me. However, a few times o |
| 12:06.50 | niter3 | ut of 10 one of us would be able to hear one each other. Any idea what the hell this is? |
| 12:07.52 | Kyler | I'm starting to think about a system that would put callers into a conference (meetme?) and then initiate calls to other devices (to add to the conference) if they're not already part of the conference. I'm curious if this has already been done. |
| 12:09.02 | [TK]D-Fender | niter3: Set "canreinvite=no" in [general] in sip.conf |
| 12:09.56 | [TK]D-Fender | niter3: Once SIP is established the phones TRY to connect directly for RTP but because the outgoing channel never started an RTP port there is no path opened on the remote router to map to the phone. |
| 12:10.20 | [TK]D-Fender | niter3: So INBOUND calls would start to work except where a new port needed to be negociated. |
| 12:11.22 | [TK]D-Fender | Kyler: Quite possibly. a call-out conference is an interesting, but somewhat rare concept. |
| 12:11.27 | niter3 | I think I understand a bit, but the example in the sip.conf canreinvite=no ; allow RTP voice traffic to bypass Asterisk |
| 12:11.31 | niter3 | That throws me off |
| 12:11.49 | niter3 | RTP voice traffic to bypass Asterisk, wouldn't I want that so it would pass on to outside phones? |
| 12:12.30 | Kyler | [TK]D-Fender: The tricky part (to me) is that I'm thinking of the outbound call as something that's forked from the inbound call handling. |
| 12:12.34 | [TK]D-Fender | niter3: What it means is when RTP bypasses * ithe devices will try to initiate direct contact with each other. Also your phone on the inside may not know to forge the return address, etc.... Reinvites=EVIL |
| 12:13.02 | [TK]D-Fender | Kyler: Didn't entirely follow you there... |
| 12:13.03 | Kyler | Is Originate() the thing to use in this situation? I haven't tried it. |
| 12:13.47 | Kyler | [TK]D-Fender: Well, the inbound call arrives and it needs to go to the conference. Fine. But it *also* needs to trigger an outbound call. That's fairly independent of the inbound call's channel. |
| 12:13.49 | *** join/#asterisk af_ (n=af@ip-170-156.sn1.eutelia.it) |
| 12:14.38 | [TK]D-Fender | Kyler: Well what factor will decide to call out to the other pary? |
| 12:14.58 | [TK]D-Fender | party* |
| 12:14.58 | Kyler | [TK]D-Fender: If the other party isn't in the conference already, the call should be placed. |
| 12:15.19 | *** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com) |
| 12:15.20 | [TK]D-Fender | Kyler: Ok, more like "how is the contact info for the other party known? |
| 12:15.34 | [TK]D-Fender | Kyler: WHO is it calling exactly? |
| 12:15.35 | Kyler | Oh, it can be hardcoded for now. |
| 12:15.42 | [TK]D-Fender | Kyler: a fixed person? |
| 12:15.48 | Kyler | Yup. |
| 12:16.03 | [TK]D-Fender | Kyler: And how many people are expected to call into that conference? |
| 12:16.15 | Kyler | Any number. |
| 12:16.58 | [TK]D-Fender | Kyler: So as each caller comes in it should check to see if the "outside moderator" is logged in or not and attempt to drag him in? |
| 12:17.12 | Kyler | Nope. I want it to be automatic. |
| 12:17.20 | Kyler | Oops. |
| 12:17.31 | Kyler | Yes "*it* should check" |
| 12:17.32 | [TK]D-Fender | Kyler: Well the ARRIVAL of a gues would be the trigger no? |
| 12:17.37 | [TK]D-Fender | guest* |
| 12:17.40 | Kyler | That's correct. |
| 12:18.22 | Kyler | Ooooh...so there's a way to trigger something from the arrival of someone in the conference, isn't there? |
| 12:19.17 | [TK]D-Fender | Kyler: Well what you'd do is do your guest authentication in the dialplan right BEFORE dumping the call into the Meetme. And again the last step before doing that will check to see if the moderator is auth'd in. |
| 12:20.10 | Kyler | O.k., but how do I initiate the call on another channel? |
| 12:20.37 | [TK]D-Fender | Kyler: Go read up on "call" files. "asterisk auto dialout" on the WIKI. |
| 12:20.50 | Kyler | Thanks! |
| 12:21.11 | [TK]D-Fender | Kyler: Drop a file with the 2 ends you want to call in a specific place and it will trigger the call |
| 12:21.18 | *** join/#asterisk ambriento (n=ambrient@200.192.160.100) |
| 12:21.42 | *** join/#asterisk tzafrir (n=tzafrir@local.xorcom.com) |
| 12:21.48 | Kyler | [TK]D-Fender: I've wanted that for other reasons. Big help. |
| 12:22.06 | [TK]D-Fender | Kyler: np |
| 12:22.23 | Kyler | http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out |
| 12:22.24 | [TK]D-Fender | Kyler: : Thats what you'd use for a "wake-up call" system, etc.... |
| 12:22.29 | MrChimpy | it's piece of piss to do |
| 12:22.46 | sxpert | any one could check bug #7825 ? |
| 12:22.46 | [TK]D-Fender | MrChimpy: Not so bad really.... |
| 12:22.52 | MrChimpy | i just impressed my bosses by doing a demo of a callback system in a couple of hours |
| 12:23.11 | mut | we finally broke the avg daily ceiling barrier |
| 12:23.20 | mut | went from 12 peak to 14 peak now |
| 12:23.26 | mut | calls on the pri |
| 12:23.28 | Kyler | [TK]D-Fender: I want to do an automated "phone tree" replacement for a large group. This sounds like a key part of the solution. |
| 12:23.51 | niter3 | [TK]D-Fender: Ok I will test this out after. My brain is done. I'm super ass tierd. |
| 12:23.53 | niter3 | 3 hours of sleep |
| 12:24.01 | mut | took 207 people |
| 12:24.02 | [TK]D-Fender | Kyler: Uh.... thats got challenges seperate from this I'm sure.... |
| 12:24.32 | [TK]D-Fender | mut: How do you measure concurrency? |
| 12:25.04 | mut | mrtg |
| 12:25.13 | mut | just a show channels and it grabs whatever data |
| 12:25.38 | Kyler | Hmmm...looks like the manager API (http://www.voip-info.org/wiki/view/Asterisk+manager+dialout) might be a cleaner way to handle this. I wonder if there's a Python interface already... |
| 12:28.32 | [TK]D-Fender | mut: So you run a script at regular test intervals and just parse out the Zap channels? |
| 12:28.32 | MrChimpy | asterisk-dev has been resolutely unhelpful so far |
| 12:28.40 | mut | yea tk |
| 12:28.50 | [TK]D-Fender | mut: Kludgy... I like it! ;) |
| 12:29.10 | [TK]D-Fender | mut : Exactly what I'd do if asked..... |
| 12:29.12 | mut | heh, i have to have a steady monitoring solution somehow |
| 12:29.24 | MrChimpy | problem is AMI is SLOW |
| 12:29.36 | [TK]D-Fender | mut: Maybe use AMI to track events in real-time? |
| 12:29.48 | [TK]D-Fender | ok maybe AMI ISN'T the best :) |
| 12:29.59 | RoyK | hm. with asterisk apps/funcs, how can i extract '21973521' from '"Operator"<sip:21973521@briiz.no;user=phone>'? |
| 12:30.01 | mut | thats more power usage |
| 12:30.05 | MrChimpy | tkd: difficult to do that and keep an accurate track of real concurrency IMHO |
| 12:30.09 | RoyK | AGI/perl? |
| 12:30.25 | ambriento | royk, cut function? |
| 12:31.02 | RoyK | ambriento: thanks |
| 12:31.15 | mut | yea |
| 12:31.35 | mut | to keep something logged into ami and monitor it seems like more trouble than its worth |
| 12:31.50 | mut | a simple asterisk-r grep seemed easiest and less intense |
| 12:31.57 | *** join/#asterisk tr2x (n=alvar@84-73-197-108.dclient.hispeed.ch) |
| 12:31.57 | mut | worked out so far |
| 12:33.08 | mut | there was always the dialplan method of setting global variables and dbgetting them whenever i wanted stats, and i could make whatever type of counters |
| 12:33.14 | mut | but i never thought into it a whole lot |
| 12:33.17 | *** join/#asterisk swytch (n=root@LNeuilly-152-22-86-193.w193-251.abo.wanadoo.fr) |
| 12:33.24 | [TK]D-Fender | RoyK: I'd say get the position of the "@" and split. then do the same for the ":" |
| 12:33.50 | RoyK | [TK]D-Fender: with CUT? |
| 12:34.51 | swytch | question about sip-sip on * : would passing the sip to sip traffic via a digium isdn card ( just out and in again ) lead to voice quality improvement ?????? |
| 12:35.13 | [TK]D-Fender | RoyK: Yup, cut it is... |
| 12:35.57 | [TK]D-Fender | swytch: Its not SIP-SIP if it passes through a card and adding more hops between ANYTHING can't possibly improve things... |
| 12:36.15 | *** join/#asterisk Dr-Linux|work (n=Linux@202.125.139.198) |
| 12:36.20 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 12:36.54 | [TK]D-Fender | swytch: any added transformation can only really take away from what was sent vs what is received |
| 12:39.38 | *** join/#asterisk tdonahue (n=tdonahue@207.138.151.58) |
| 12:40.15 | [TK]D-Fender | OMG, why are the morons on this page still allowed to put out such poor * examples explained wrong EVERY time?! http://www.voipplanet.com/backgrounders/article.php/3628741 |
| 12:40.58 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 12:41.58 | *** join/#asterisk luckasrg (n=lucasrg@204.97.196.20) |
| 12:44.23 | tdonahue | [TK]D-Fender, because we don't live in a totaliarian state which controls what the media puts out :-D |
| 12:44.23 | tdonahue | at least, that is what the media tells us.... |
| 12:44.23 | [TK]D-Fender | tdonahue: True, but this guy should have received enough slaps to either wake up or shut up..... |
| 12:44.23 | [TK]D-Fender | tdonahue: And the situation there is chronic.... |
| 12:44.45 | luckasrg | hi |
| 12:44.45 | luckasrg | all |
| 12:44.46 | tdonahue | anyone know if - (the minus sign) is a valid telco callerid character? |
| 12:44.46 | mut | damnit someone left my window open here at work all night long |
| 12:44.47 | mut | now there are moths flying all over in the office |
| 12:49.40 | tzanger | where do I download the g729 codec library from? I've just upgraded a system that crashed last night and need the .so so I can re-register the licences |
| 12:50.14 | zoa | it used to be on some ftp |
| 12:50.14 | tzanger | indeed |
| 12:50.17 | zoa | accessible through asterisk.org iirc |
| 12:50.17 | sxpert | tzanger: they are on digium's ftp |
| 12:50.25 | tzanger | ahh |
| 12:50.31 | tzanger | ftp.digium.com != ftp.asterisk.org |
| 12:50.42 | sxpert | obviously |
| 12:51.16 | swytch | [TK]D-Fender: we have observed otherwise. |
| 12:51.37 | [TK]D-Fender | swytch: Then something very wrong is happening... |
| 12:52.12 | swytch | [TK]D-Fender: we have tries passing the rtp thru a third party rtpproxy and it seems like some jitter buffer or other functionality is smoothing the voice. |
| 12:52.22 | swytch | /tries/tried/ |
| 12:52.46 | swytch | [TK]D-Fender: i dont understand how either. |
| 12:52.53 | *** join/#asterisk Meaty` (n=meaty3@office.abi.ca) |
| 12:53.27 | key2 | if I have a STM link with optic fiber, can I use asterisk on it ? |
| 12:53.34 | swytch | [TK]D-Fender: I made a test with my boss. And I called him on the third party rtpproxy and the asterisk at random. He knew each time what was used. |
| 12:53.37 | *** join/#asterisk muppetmaster (n=jasongoe@81.184.73.169) |
| 12:54.19 | [TK]D-Fender | swytch: Bad network... hrm... what sip phones? |
| 12:54.41 | key2 | [TK]D-Fender: do you know ? |
| 12:54.54 | *** join/#asterisk roving_prole (n=Harper@72-254-127-12.client.stsn.net) |
| 12:55.16 | swytch | [TK]D-Fender: I am thinking about trying to have for example an ISDN card with the octo-thing chip and just pass the rtp thu it out and in imediately. Its crazy if it shows this is what to do. |
| 12:55.27 | swytch | [TK]D-Fender: we have audiocodes 118x |
| 12:55.33 | [TK]D-Fender | key2: Don't know STM.... |
| 12:55.46 | key2 | [TK]D-Fender: well SS7 ISUP |
| 12:56.03 | swytch | [TK]D-Fender: i mean 118 FXS sip gateways |
| 12:56.11 | [TK]D-Fender | key2: No idea..... that stuff is "out of my league" |
| 12:57.17 | mut | anyone here know how to get a badge to get into SBC/ATT central offices? |
| 12:57.18 | swytch | [TK]D-Fender: the audiocodes are on the callshops side, and we pass outgoing traffic thru a MERA |
| 12:57.44 | mut | we got denied a badge going through our current clec, so now we gotta try with our own but we dunno who our sbc rep is yet |
| 12:58.17 | sxpert | mut: get a lockpicking kit ;D |
| 12:58.20 | mut | heh |
| 12:58.35 | mut | well i think they know we're going to be switching so thats why they deniedit |
| 12:58.43 | swytch | [TK]D-Fender: the mera always transcodes, and the audiocodes contains a jitterbuffer, so i dont think its a jitter problem. But what else could it be. We have every kind of walky talky sound problems. |
| 12:58.52 | sxpert | take the key off the back of one of those ATT/SBC trucks |
| 12:59.11 | *** join/#asterisk charlatan (n=chatzill@202.61.170.62) |
| 12:59.18 | charlatan | Hi |
| 12:59.40 | mut | :P |
| 13:00.06 | luckasrg | hi i have a problem when i call form a sip exten IAD to the x100p trunk i get heard the voice like a roboot |
| 13:00.23 | mut | no clec in here got any contacts for that kinda thing? |
| 13:00.35 | charlatan | Anyone know of a comercial asterisk client that will let you park, put on hold, presence ? |
| 13:00.51 | luckasrg | but in calls the other way around and the ivr sounds fine |
| 13:01.03 | charlatan | Guess I'm looking for teh RolsRoyce client of the Asterisk world |
| 13:01.20 | mut | client being? |
| 13:01.34 | dorel__ | is it valid to define in zapata.conf like this: channels => 1-4,7-8 ? |
| 13:01.37 | charlatan | softphone that will let me control hardphone |
| 13:03.00 | *** join/#asterisk murf (n=steve_mu@216.166.159.235) |
| 13:03.26 | *** join/#asterisk muppetmaster (n=jasongoe@81.184.73.169) |
| 13:04.27 | charlatan | Can anyone suggest a full featured client for asterisk I don't mind paying for it ? |
| 13:05.06 | [TK]D-Fender | charlatan: Presume you mean "softphone"? |
| 13:05.14 | charlatan | yes sorry |
| 13:05.34 | [TK]D-Fender | charlatan: eyemBeam for Windows.... |
| 13:05.49 | charlatan | We have a system called SoX atm it had a very nice softphone for windows and I need a replacement |
| 13:06.08 | [TK]D-Fender | charlatan: Really can't use a hardphone for your needs? |
| 13:06.22 | charlatan | No they demand a softphoen as well |
| 13:06.44 | swytch | [TK]D-Fender: im sorry to yell again, but do you have a hint about what could be the cause (and what could remove it) to our walky-talky sound problems that are reduced with a 3rdparty magical rtpproxy. ?? |
| 13:06.45 | phearless | damned echo !!!! |
| 13:06.51 | charlatan | Wanted something where u can drag and click calls, park, put on hold. See status of other phones etc |
| 13:06.56 | phearless | sorry for the echo |
| 13:06.57 | swytch | damned echo!!! |
| 13:07.09 | phearless | I can hear myself |
| 13:07.17 | swytch | seams like theres echo in the swytch.. |
| 13:07.42 | swytch | phalacee-away: try stuffing it with feathers. |
| 13:07.52 | [TK]D-Fender | swytch: Not offhand... |
| 13:08.18 | swytch | [TK]D-Fender: "offhand" what does that mean? |
| 13:08.43 | swytch | [TK]D-Fender: screw on your hands |
| 13:09.13 | luckasrg | my asterisk was working fine and yesterday i started hearing this robotic voice |
| 13:09.21 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 13:09.35 | MrChimpy | luck: in your mind? |
| 13:09.37 | phearless | I do not understand this echo... |
| 13:09.48 | darkskiez | this echo... |
| 13:09.50 | luckasrg | did someting got corrupted some how in the install |
| 13:09.51 | darkskiez | cho... |
| 13:10.13 | mut | any clec in the house?! |
| 13:10.13 | MrChimpy | yes. corrupted installs usually result in "robot voices" |
| 13:10.29 | MrChimpy | what does the magic robot say? |
| 13:10.44 | darkskiez | luckasrg: definately using a decent codec? |
| 13:10.46 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:10.46 | mut | ALL YOUR BASE ARE BELONG TO US |
| 13:10.59 | [TK]D-Fender | swytch: Means "no" sorry.... |
| 13:11.04 | tzanger | is the g729 codec not available on trunk yet (new loader and all) |
| 13:11.04 | MrChimpy | you could either be talking about audio corruption or you could be running some app that's reading error messages out with festival |
| 13:11.27 | *** join/#asterisk DrukenHME (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 13:11.30 | [TK]D-Fender | mut: Make your time |
| 13:11.40 | DrukenHME | morning peoples |
| 13:11.55 | mut | time.. ugh.. |
| 13:12.13 | tzanger | morning |
| 13:12.23 | DrukenHME | [TK]D-Fender, any idea what unit the timeout= is in queues.conf? minutes or seconds? |
| 13:12.33 | mut | morn tzafrir |
| 13:12.36 | mut | tzanger too |
| 13:12.40 | [TK]D-Fender | DrukenHME: seconds |
| 13:12.43 | DrukenHME | damn nick complette |
| 13:12.51 | DrukenHME | [TK]D-Fender: k, thanks |
| 13:12.58 | tzanger | heh |
| 13:13.02 | luckasrg | yes i am using g711u since is in my lan |
| 13:13.19 | luckasrg | i do not need to compress much |
| 13:13.48 | charlatan | kk can people just tell me there fav softphones for use with asterisk ? |
| 13:14.19 | zoa | idefisk (shameless plug, i'm making that phone) |
| 13:14.19 | luckasrg | no the problem only happends when i make call from sip to x100p trunk |
| 13:14.33 | *** join/#asterisk spr1te (n=spr1te@213.227.193.75) |
| 13:14.35 | *** join/#asterisk muppetmaster (n=jasongoe@81.184.73.169) |
| 13:14.45 | zoa | hey ho muppet |
| 13:14.55 | luckasrg | i am running this on a dual core with kernel 2.6 |
| 13:14.57 | *** join/#asterisk mercestes (n=merceste@216.54.143.2) |
| 13:15.14 | luckasrg | cluod this be related to clocking problems? |
| 13:15.15 | swytch | [TK]D-Fender: another weird thing is that even tho this MERA transcode everything to G729b on the "other" side, we have far less problems when passing it g711 via the asterisk than when passing it g711b from the audiocodes. |
| 13:15.28 | DrukenHME | when the drops out of the queue does it goto +101 or +1 ? |
| 13:15.29 | zoa | Lukasrg: i didnt follow the complete conversation |
| 13:15.33 | *** join/#asterisk momelod (n=momelod@bas5-toronto12-1168028839.dsl.bell.ca) |
| 13:15.39 | momelod | good morning |
| 13:15.41 | zoa | but if your sound is really garbled, your x100p might be broken |
| 13:16.01 | zoa | unless it also happens on pure ip calls |
| 13:16.14 | [TK]D-Fender | Druken: the "timeout" in queues.conf determines the ring time to agent, not the lifespan of a caller in the queue |
| 13:16.26 | luckasrg | but when the call comes from the x100p to the sip it sounds fine |
| 13:16.27 | swytch | [TK]D-Fender: sorry, we have far more problems when passing g729b thu * : 711u works best. |
| 13:16.40 | [TK]D-Fender | Druken : You would be looking for the timeout parm in the Queue application itself as you call it from your dialplan. |
| 13:16.58 | momelod | having an odd problem.. sometimes during a call asterisk will just drop a call mid conversation.. my setup is sip<->asterisk<->zap<->pstn |
| 13:17.15 | momelod | nothing really in the console to describe what happend.. |
| 13:17.16 | [TK]D-Fender | swytch: * doesn't transcode b that I''ve heard of unless its a well buried subset....\ |
| 13:17.49 | luckasrg | and when i plece the call from a softphone it sounds fine |
| 13:18.01 | *** join/#asterisk aixa (n=Miranda@www.crediweb.lv) |
| 13:18.02 | DrukenHME | [TK]D-Fender: ahh... excelent |
| 13:18.03 | *** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk) |
| 13:18.35 | aixa | may I ask a newb question regarding extension patterns and may expect not being banned from here? |
| 13:19.09 | momelod | aixa: the worst that can happen is some1 will tell u to rtfm |
| 13:19.19 | momelod | it wont hurt.. promise |
| 13:19.33 | mitcheloc | or they could start with rtfctr |
| 13:19.57 | momelod | channel topic retard? |
| 13:19.57 | aixa | I am trying to achieve following: the extension should match with a dialed number which would start with *23 then folowed by a number of number and then finish with * again |
| 13:20.24 | aixa | so i thought that _*23X.* should be appropriate |
| 13:20.32 | mitcheloc | haha... i thought it used to say "ask, don't ask to ask"... guess not |
| 13:20.51 | zoa | luckasrg: so it snot the x100p |
| 13:20.54 | [TK]D-Fender | aixa: You can't have a variable portion terminated by a fixed portion, sorry. |
| 13:20.54 | luckasrg | it only happends on calls sip<->aster<->zap<->pstn |
| 13:21.03 | aixa | okay |
| 13:21.16 | aixa | so will have to look for workaround |
| 13:21.23 | luckasrg | probabli not |
| 13:21.38 | aixa | wanted to have a call forward service where line doesnt get picked up |
| 13:21.46 | luckasrg | does it matter if is a clon or not? |
| 13:21.47 | [TK]D-Fender | aixa: best you can do is "exten => _*23X.,1,Whatever()" and in there check to see if the last one is a "*" |
| 13:22.14 | aixa | like waitForDigit? |
| 13:22.21 | luckasrg | or a problem with the sm dual procesor kernel |
| 13:22.30 | *** join/#asterisk Filar (n=none@sw.m5net.com) |
| 13:22.37 | momelod | has anyone ever exerienced zap<->sip bridged calls just dropping w/o warning? |
| 13:22.42 | aixa | oh the miracles of SNMP kernel ;))) |
| 13:22.48 | mitcheloc | [TK]D-Fender: why not _*23XXXXX* ? |
| 13:23.03 | niter3 | AHAHHA i'm so rude today |
| 13:23.04 | [TK]D-Fender | aixa: No, won't make it any better... |
| 13:23.05 | aixa | I dont know the length of number which will be used for transfer |
| 13:23.15 | niter3 | so fucking pissy this morning, and i DON'T CARE! |
| 13:23.20 | [TK]D-Fender | mitcheloc: Not with ".", "X" should work. |
| 13:23.21 | niter3 | It's to damn fucking early. |
| 13:23.23 | aixa | it might be internal (3 numbers) |
| 13:23.34 | aixa | it might me cell (8 numbers) |
| 13:23.42 | aixa | it might be landline (7 numbers) |
| 13:23.50 | DrukenHME | can't we use a ring instead of moh for the queues anymore? |
| 13:23.50 | swytch | [TK]D-Fender: i see that you are helping half of the guys here. thanks. do you know of anyone else here that have used audiocodes and/or MERA?? |
| 13:23.54 | mitcheloc | aixa, make a seperate rule for each one |
| 13:23.55 | aixa | or any international number of an unknown length |
| 13:24.22 | [TK]D-Fender | swytch: Not with the kind of answers you're looking for.... |
| 13:24.26 | DrukenHME | i can buy audiocodes, never used it tho... |
| 13:24.45 | [TK]D-Fender | aixa: why the trailing * at all? |
| 13:25.00 | aixa | to know that the number has ended |
| 13:25.10 | aixa | uses want to have immediateconfirmation |
| 13:25.13 | aixa | users |
| 13:25.21 | aixa | sorry for mistypes |
| 13:25.59 | *** join/#asterisk muppetmaster (n=jasongoe@81.184.73.169) |
| 13:26.00 | DrukenHME | [TK]D-Fender: did they remove the ring instead of moh on the queues? |
| 13:26.07 | aixa | okay, perhaps I should make a simple application with agi then - prompting for number followed by pund key |
| 13:26.16 | swytch | [TK]D-Fender: what kind of answers am i looking for (not to be rude, just interrested in what you think) |
| 13:26.32 | aixa | 1.2.9 asterisk an i can use ring on queues |
| 13:26.53 | aixa | r parameter for Queue command if io am not terribly wrong |
| 13:27.20 | [TK]D-Fender | Druken : Not that I know of and I see no reason why they would... |
| 13:27.20 | swytch | DrukenHME: why can you buy but _not_ use audiocodes?? |
| 13:27.50 | DrukenHME | that isn't what i said, i said i can buy them, but have not used them... |
| 13:28.17 | swytch | DrukenHME: ok ok ok.. /-8 |
| 13:28.53 | aixa | you all seem like a friendly bunch over here |
| 13:29.42 | swytch | aixa: are you hetero (now im gonna be kicked) |
| 13:29.47 | luckasrg | zoa have you bumped in to the same problem before? |
| 13:30.18 | aixa | monogamic hetero ("friendly bunch" sounds gay?) |
| 13:30.20 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 13:30.31 | swytch | aixa: that was my koje, yeah |
| 13:30.40 | swytch | /koje/joke/ |
| 13:30.50 | swytch | (little/big endian problem) |
| 13:31.08 | niter3 | so you guys see that telemarker thing |
| 13:31.12 | *** join/#asterisk sysfault (n=exalted@p3m/member/sysfault) |
| 13:31.18 | *** part/#asterisk sysfault (n=exalted@p3m/member/sysfault) |
| 13:31.22 | niter3 | a guy made an extension with a recorded voice tht he sends telemarkers to |
| 13:31.26 | niter3 | halrious. |
| 13:31.33 | swytch | niter3: telemarker or telemarketer? |
| 13:31.42 | niter3 | I'd set that up to, but uhh I don't have any telemarketers calling |
| 13:31.47 | niter3 | 3 hours of sleep |
| 13:31.51 | niter3 | me tierd |
| 13:32.06 | niter3 | At the damn office today so damn tierd. |
| 13:32.26 | ambriento | why you guys work so hard? |
| 13:32.33 | sxpert | niter3: there was that guy that was so fed up with his bank that he set up an asterisk box with a t1 and used that to hog their pbx by putting them on hold ;D |
| 13:32.38 | niter3 | Yah he mad e prerecorded person saying I'm really interested tell me more blah blah |
| 13:32.45 | sxpert | niter3: rotfl |
| 13:32.47 | swytch | ambriento: cause we cant get that sound quality realy nice... |
| 13:32.59 | zoa | lukasrg: not really in the last 2 year |
| 13:33.05 | niter3 | sxpert: how did that work |
| 13:33.07 | zoa | didnt read the complete conversation though |
| 13:33.08 | aixa | swytch: :) "back to the pile, back to the pile" :P |
| 13:33.09 | niter3 | when his bank would call? |
| 13:33.09 | zoa | no time |
| 13:33.13 | Aurs | asterisk-sounds has some nice prompts |
| 13:33.24 | Aurs | spam.gsm is a good one |
| 13:33.26 | sxpert | niter3: the dude would just dial the bank and fill all the slots on *their* t1 with his own ;D |
| 13:33.41 | sxpert | then the thing would put the bank ppl on hold ;D |
| 13:33.46 | swytch | aixa: sorry what does that mean (aparently me english lacks some expressions) |
| 13:33.48 | niter3 | ahahh fuckers.. |
| 13:34.00 | niter3 | Did he get fined? |
| 13:34.05 | sxpert | nope ;D |
| 13:34.20 | sxpert | he didn't commit illegal entry in the system ;D |
| 13:34.37 | sxpert | and he was after all paying for all the calls |
| 13:34.41 | niter3 | ahah that's funny. |
| 13:34.43 | sxpert | so the telco didn't give a shit |
| 13:34.58 | niter3 | hrm.. i gotta think of something to do myself |
| 13:35.02 | niter3 | Would be funny |
| 13:35.04 | sxpert | lol |
| 13:35.09 | aixa | swytch: quote from the south park |
| 13:35.37 | swytch | aixa: havent seen that episode then. would you give a complete resumee?... |
| 13:35.58 | aixa | you dont get banned for offtopic here? |
| 13:36.17 | zoa | noprx |
| 13:36.18 | zoa | nopez |
| 13:36.19 | swytch | aixa: write some megs of pastebin then |
| 13:36.28 | aixa | :)) okay |
| 13:36.30 | zoa | you can talk about IKEA, dogs and penises all you want here :) |
| 13:36.42 | coppice | you get banned for on topic here |
| 13:36.46 | zoa | http://digg.com/offbeat_news/IKEA_insists_photo_of_dog_does_not_show_human_like_penis |
| 13:36.48 | swytch | aixa: no no no i was just kidding! dont do it. |
| 13:36.50 | aixa | so the stroy was that: guys from future came back to present |
| 13:36.56 | zoa | but dont paste anything |
| 13:37.00 | mut | any clec in the house? |
| 13:37.09 | zoa | be nice or i will release coppice! |
| 13:37.10 | aixa | coppice :)) |
| 13:37.10 | zoa | :p |
| 13:37.18 | *** join/#asterisk muppetmaster (n=jasongoe@81.184.73.169) |
| 13:37.59 | MrChimpy | there's a place in the UK called Argos - they have a catalogue. One year it comes out with kid sat next to a nice cute puppy. |
| 13:38.24 | MrChimpy | they presumably didn't notice the puppy had a sizeable pink boner |
| 13:38.34 | aixa | swytch: : okay heres the episode if ur interested (http://www.youtube.com/watch?v=mvUxKqE666A) |
| 13:39.06 | swytch | zoa: its not the dogs penis. its the IKEA butplug of the woman |
| 13:39.20 | coppice | Argos didn't notice a deficiency in something they supplied to the public? never :-) |
| 13:40.39 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 13:41.02 | mut | ok nvm |
| 13:41.58 | bXi | does one of you know how to fix /usr/lib/asterisk/modules/app_trunkisavail.so: undefined s |
| 13:42.00 | bXi | ymbol: zap_channel_count |
| 13:45.55 | *** join/#asterisk n0n0x (n=n0n0x@201.216.229.213) |
| 13:46.24 | BeBoo | Do i need to load any module to get the 'dial' command in CLI mode to work? |
| 13:46.38 | mut | found out why they denied by badge |
| 13:46.38 | n0n0x | yep, oss module |
| 13:46.43 | mut | the picture i sent wasn't large enough |
| 13:46.45 | BeBoo | okay thanks |
| 13:46.50 | mut | fookers |
| 13:47.09 | BeBoo | Hmm, shouldn't a reload be enough? |
| 13:47.25 | *** join/#asterisk muppetmaster (n=jasongoe@81.184.73.169) |
| 13:48.15 | luckasrg | what will be a good asterisk set up |
| 13:48.27 | luckasrg | as far as kernel, version etc |
| 13:48.59 | mut | god i love customers sometimes |
| 13:49.16 | Dr-Linux|work | how can i debug???? >> Aug 31 19:00:47 WARNING[24362]: res_config_mysql.c:135 realtime_mysql: MySQL RealTime: Failed to query database. Check debug for more info. |
| 13:49.23 | mut | i send a support ticket to our tech support to get ahold of a customer because they were spamming, this was 2 weeks ago |
| 13:49.29 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
| 13:49.44 | mut | i see about 15 seperate notes saying they called her |
| 13:49.49 | n0n0x | BeBoo, i don't think so |
| 13:49.57 | mut | and left messages* |
| 13:50.14 | mut | so i finally got tired of it and disabled the lady, 2 hrs later we get a call |
| 13:50.16 | n0n0x | Somebody knows why i don't have the AGENT function?? |
| 13:50.26 | mut | "omg you shut off my phone, this is for my business turn it back on" |
| 13:50.40 | mut | total BOFH moment |
| 13:51.16 | niter3 | hrm... Anyone got a site for good ideas for asterisk |
| 13:51.20 | niter3 | additonal modules |
| 13:51.20 | niter3 | etc |
| 13:51.22 | niter3 | ? |
| 13:51.27 | *** join/#asterisk darkskiez (n=mbryars@194.247.78.146) |
| 13:51.30 | mut | she tells me she is going to bring in her pc's to get them cleaned up, no pc comes in, disable it first thing again this morning |
| 13:51.41 | mut | calls in again, now its shut off until she brings the blasted pc in |
| 13:51.59 | mut | niter3: good ideas? |
| 13:52.02 | mut | as in stuff to create? |
| 13:52.07 | mut | or stuff created? |
| 13:52.11 | niter3 | addons.. |
| 13:52.13 | niter3 | created |
| 13:52.31 | mut | wiki has all kinda stuff |
| 13:52.37 | mut | i dunno otherwise, i tend to make whatever i need |
| 13:52.55 | luckasrg | how can i check if my x100p is fine? |
| 13:53.03 | luckasrg | what est can i run |
| 13:53.13 | Dr-Linux|work | anybody and clue on my problem? |
| 13:53.16 | BeBoo | n0n0x, doesn't seem to work :/ |
| 13:53.58 | BeBoo | Anyway, when i have a "DBdel(FEAT/${CALLERIDNUM}/timeout)", where is that information stored? |
| 13:56.10 | *** join/#asterisk sb_mx (n=sb_mx@dsl-200-67-149-101.prod-empresarial.com.mx) |
| 13:57.24 | n0n0x | Someone has an AGI scripts that tells the agent if it's logged in or not? or knows any way how to do this? |
| 13:57.42 | key2 | someone knows if with a EAGI, i could play a raw audio buffer, or can I just get an audio buffer |
| 13:58.42 | *** join/#asterisk muppetmaster (n=jasongoe@81.184.73.169) |
| 14:00.11 | [TK]D-Fender | luckasrg: You MIGHT want to consider actually try placing and receiving calls with it... :) |
| 14:00.33 | [TK]D-Fender | BeBoo: In the AstDB. Look up "asterisk database" on the WIKI for more info. |
| 14:00.42 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 14:00.53 | Dr-Linux|work | [TK]D-Fender, do you know how can i debug realtime ? |
| 14:01.43 | [TK]D-Fender | Dr-Linux|work: You keep asking and the answer isn't changing : I nver used real-time. First thing you might want to prove is taht your connection info is right. Them see if MySQL has anything to say about the ATTEMPT itself |
| 14:01.57 | [TK]D-Fender | Dr-Linux|work: And If I knew I'd have said something else already. |
| 14:02.58 | Dr-Linux|work | [TK]D-Fender, thanks sir, i'm searching for my problem, my db is connected properly .. but still this happend.. anyway .. looking into it |
| 14:03.00 | *** join/#asterisk nortex (n=breeves@snapper.titanspecialties.com) |
| 14:03.33 | [TK]D-Fender | Dr-Linux|work: Use an outside client and try connecting with what you have in your realtime config setup |
| 14:03.46 | BeBoo | Thanks, just found it. |
| 14:04.34 | BeBoo | I do have a problem with it. When i run the DBdel(FEAT/${CALLERIDNUM}/timeout) in the extension, it still goes to the voicemail after 30 seconds |
| 14:05.29 | *** part/#asterisk luckasrg (n=lucasrg@204.97.196.20) |
| 14:05.32 | [TK]D-Fender | BeBoo: Your dialplan does what you tell it to... where does it even CHECK for that in it? The value itself has no inherent meaning unless you implement it yourself. |
| 14:05.47 | [TK]D-Fender | BeBoo: Try pastebin-ing your dialplan section..... |
| 14:05.49 | *** join/#asterisk Deeewayne (n=dwayne@ool-44c0d56e.dyn.optonline.net) |
| 14:06.00 | [TK]D-Fender | (me braces for what is to be the blatantly obvious outcome) |
| 14:06.07 | BeBoo | The output from show dialplan? |
| 14:06.39 | [TK]D-Fender | BeBoo: no, straight copy & paste from your dialplan, not the * CLI interpretation. |
| 14:06.41 | [TK]D-Fender | ~pb |
| 14:06.45 | jbot | rumour has it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/, or http://www.paste-it.net/ |
| 14:06.51 | [TK]D-Fender | Use pastebin.ca |
| 14:07.05 | macTijn | use paste-it.net :) |
| 14:07.49 | Dr-Linux|work | [TK]D-Fender, everything worked for me using realtime, but queue is having problem |
| 14:08.50 | sxpert | how would I use realtime with postgresql ? |
| 14:09.04 | BeBoo | Sorry, i'm not sure what information you want then - i'm pretty new to asterisk |
| 14:09.31 | [TK]D-Fender | BeBoo: extensions.conf..... |
| 14:09.33 | rpm | sxpert: you use unixodbc. its quite easy to setup, look on voip-info.org for the schema |
| 14:09.40 | BeBoo | The whole file? |
| 14:09.48 | sxpert | rpm: hah. ok |
| 14:10.00 | *** join/#asterisk muppetmaster (n=jasongoe@81.184.73.169) |
| 14:10.06 | sxpert | rpm: the thing is I want to share the blacklist on multiple boxes ;D |
| 14:10.12 | [TK]D-Fender | BeBoo: TRhe part that USES your AstDB.... |
| 14:10.19 | *** join/#asterisk BjornRobertsson (n=bjornr@213-213-148-71.xdsl.is) |
| 14:10.44 | rpm | sxpert: in asterisk svn there is something called func_odbc, which you could match against your blacklist. |
| 14:10.58 | sxpert | hah |
| 14:11.01 | BjornRobertsson | Anyone have experience with Swyx phones and asterisk? (410 economy) |
| 14:11.12 | sxpert | but I can't really use svn on a production server now can I ? |
| 14:11.34 | mut | why not |
| 14:11.48 | mut | i'm using a cvs from september of last year |
| 14:11.55 | mut | been running like 8 months straight now |
| 14:11.59 | rpm | sxpert: i do. it works fine being that it is only a copy of 1.2 to a new branch, which 1.2 is stable enough, just don't enable anything in trunk which you don't know what it does. |
| 14:12.18 | sxpert | hmm right |
| 14:13.07 | mut | not that i recommend still running cvs from last year |
| 14:13.14 | mut | but it works |
| 14:13.21 | BeBoo | [TK]D-Fender, the part that uses the AstDB, should it be using some dbget() commands? |
| 14:13.34 | sxpert | mut: heh |
| 14:13.51 | BeBoo | I'm comparing a working installation with a non-working, and i cannot see anywhere it uses the dbget() or anything matching timeout |
| 14:13.59 | [TK]D-Fender | BeBoo: Its YOUR dialplan...you should know what you are trying to do and where.... |
| 14:14.43 | *** join/#asterisk Ciber311 (n=Ciber311@user-1087e94.cable.mindspring.com) |
| 14:14.46 | rpm | GotoIf( $[ ${ODBC_ANTI_GIRLFRIEND(${CALLERIDNUM}) = $CALLERIDNUM ]?allow:block) |
| 14:16.00 | *** join/#asterisk ClonMaste (i=ClonMast@200.91.70.212) |
| 14:16.05 | mut | you're a bit backward on that if |
| 14:16.08 | BeBoo | Well i really dont - i'm just helping a guy out, where the old technican has left, i'm just trying to figure out why it isn't working. I've been searching extensions.conf for timeout/dbget/feat, without finding anything on the working installation |
| 14:18.20 | ClonMaste | Hi, Are digium cards with echo cancellation worth the extra money over a "regular" one? |
| 14:20.02 | niter3 | hey guys when calling an extensions and the voice mail picks up, the recorded message i left for the unavailable or busy message does not get played. |
| 14:20.09 | niter3 | Anyone see this before? |
| 14:20.23 | mitcheloc | ClonMaste: yes |
| 14:21.08 | zoa | clon: the new ones yes, the old ones no |
| 14:21.16 | [TK]D-Fender | BeBoo: If you can't find it in a text search there is something very wrong with what you're doing..... |
| 14:21.24 | *** join/#asterisk muppetmaster (n=jasongoe@81.184.73.169) |
| 14:21.57 | [TK]D-Fender | BeBoo: Why don't you just describe how this box was set up (I'm already about 5-9's sure how but I like the game....) |
| 14:22.05 | zoa | the ones with the purple daughterboard is what you want |
| 14:22.36 | ClonMaste | Thanks mitcheloc! Yes zoa... |
| 14:22.39 | [TK]D-Fender | niter3: Then you aren't calling VoiceMail with the right parameters. |
| 14:22.56 | BeBoo | I really dont know, thats the old technican who did the initial configuration. He has now left, and wont help with it anymore |
| 14:23.35 | niter3 | I take it i need to add to the extensions $u |
| 14:23.36 | niter3 | ? |
| 14:23.44 | niter3 | when it pulls up voicemail |
| 14:24.34 | ClonMaste | We are tired of "upgrading" our existing Siemens PBX, and looking to integrate it with an Asterisk box.... |
| 14:24.38 | [TK]D-Fender | BeBoo: Then fine, pastebin the entire extensions.conf. |
| 14:25.08 | [TK]D-Fender | niter3: Paste exactly how you call it now. |
| 14:25.37 | niter3 | ok i just changed it |
| 14:25.41 | niter3 | opps. I'll show you what I have |
| 14:26.07 | niter3 | exten => 1,2,Voicemail(su${222@main}) |
| 14:26.11 | niter3 | will this work |
| 14:26.12 | niter3 | ? |
| 14:26.29 | [TK]D-Fender | niter3: No.... {} = bad |
| 14:26.44 | [TK]D-Fender | niter3: $ too |
| 14:27.00 | [TK]D-Fender | exten => 1,2,Voicemail(su222@main) |
| 14:27.38 | niter3 | Ok this is what I have |
| 14:27.53 | *** join/#asterisk ygguh (n=Thomas@nj-nothing-00.goamerica.net) |
| 14:28.45 | *** join/#asterisk h3x (n=h3xor@64.192.116.17) |
| 14:28.46 | niter3 | exten => 1,1,Dial(SIP/chris,30) - exten => 1,2,Voicemail(su222@main) - exten => 1,3,PlayBack(vm-goodbye) - exten => 1,4,Wait(20 - exten => 1,5,HangUp() |
| 14:28.55 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 14:29.05 | BeBoo | ok [TK]D-Fender just a moment |
| 14:29.17 | ygguh | how do I send arguments and then receive the results to an AGI script? |
| 14:30.28 | [TK]D-Fender | niter3: Looks ok, though ditch the wait... time wastted for nothing. |
| 14:32.00 | Splas | Can anyone tell me how far off 1.4 currently is? |
| 14:32.30 | [TK]D-Fender | SplasPood: About 50 yards |
| 14:32.38 | SplasPood | [TK]D-Fender: :P |
| 14:32.42 | *** join/#asterisk muppetmaster (n=jasongoe@81.184.73.169) |
| 14:32.54 | SplasPood | [TK]D-Fender: Just wondering... I had heard end of august batted around before.. |
| 14:32.59 | niter3 | what does the s do? |
| 14:33.11 | rpm | what is VLDTMF? variable length dtmf tones? |
| 14:33.12 | SplasPood | niter3: no instructions |
| 14:33.18 | [TK]D-Fender | SplasPood: And I've heard July... so I guess nobody is happy :) |
| 14:33.51 | niter3 | i see, |
| 14:34.24 | ClonMaste | Are TE207P cards the good ones? |
| 14:34.26 | niter3 | ok cool.. Now I have to figure out if somebody is on the phone it will it just go to su or will it play the busy one? |
| 14:35.08 | rpm | niter3: use the n+101 priority then |
| 14:35.33 | rpm | or hints |
| 14:35.50 | niter3 | I am not familiar with n+101 priority |
| 14:36.08 | dsfr | ClonMaste: the TE207P is the dual span 5v card that includes the new carrier grade VPM450M echo-cancellation module. |
| 14:36.28 | rpm | if you do a "show application Dial" it will show you how to use it. |
| 14:37.02 | niter3 | k thanks |
| 14:37.37 | ClonMaste | Thanks dsfr! Are these overkill for PBX integration in an office? |
| 14:39.12 | *** join/#asterisk DrukenHME (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 14:39.42 | ghenry | is this still valid for 1.2.x default => mp3:/var/lib/asterisk/mohmp3-empty,http://www.waixwave.com:8000/ |
| 14:39.52 | ghenry | trying to use an audio stream for moh |
| 14:41.14 | ghenry | nevermind, musiconhold.conf.sample has the answers ;-) |
| 14:41.15 | ghenry | sorry all |
| 14:42.12 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 14:43.00 | ghenry | ah, streamplayer, nice |
| 14:43.59 | *** join/#asterisk muppetmaster (n=jasongoe@81.184.73.169) |
| 14:44.06 | ClonMaste | we just need one E1 interface to our current Siemens PBX, to handle with asterisk the new extensions that we need to grow... |
| 14:44.57 | ClonMaste | don't know if all this "carrier grade echo-cancellation module" is needed for this... :-( |
| 14:45.03 | Assid | ~fxofxs |
| 14:45.13 | jbot | it has been said that 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. |
| 14:45.13 | *** join/#asterisk dhill (i=dhill@fog.mindcry.org) |
| 14:45.22 | dhill | has anyone gotten a mediatrix to work with asterisk? I am getting Forbiddens on INVITE's |
| 14:47.10 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 14:47.13 | *** join/#asterisk rpsgb (n=rorytara@gateway.rpsgb.org) |
| 14:47.57 | rpsgb | is anybody about? |
| 14:50.56 | mitcheloc | only 260 people ;0 |
| 14:51.32 | jhb | hi *. I try to use realtime, and I can see a user when doing sip show user 2000 load, but registering does not work - I get " Username/auth name mismatch", and no query appears in the mysql.log. Ideas? |
| 14:51.40 | *** join/#asterisk agallo (n=Antonio@host63-216-static.34-88-b.business.telecomitalia.it) |
| 14:52.37 | agallo | Hello, anyone knows a phone where we can control ring tones from Asterisk? so i can set different ring tones for it depending if the call is from a local phone or from an external trunk? |
| 14:54.12 | stoffell_h | agallo: polycom can do that, i also think snom can.. it's called "distinctive ringtone" |
| 14:54.16 | jhb | also, if I use the same user entry in sip.conf it works... |
| 14:54.26 | *** join/#asterisk DrukenHME (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 14:55.14 | agallo | stoffell_h, ty atm i'm full of GXP-2000 it seems to be customizable thru Alter-Info but only work for the 1st custom ring tones... also ring tones not easily changeable onto that phones; if Snom support it i will be very happy :) |
| 14:55.38 | *** join/#asterisk brif8 (n=brif8@186-15.9-67.tampabay.res.rr.com) |
| 14:56.02 | sxpert | the unixODBC docs are crap |
| 14:56.14 | stoffell_h | agallo: indeed, gxp's can't do it like they should.. (yet) |
| 14:56.18 | *** join/#asterisk muppetmaster (n=jasongoe@81.184.73.169) |
| 14:56.47 | brif8 | I have a remote IP phone. I can make outbound calls via the * server. But I can't receive inbound calls ? sip show peers has the peer as UNREACHABLE, yet I can still make outbound calls |
| 14:57.21 | *** part/#asterisk anthonyl (n=anthonyl@c-71-57-43-221.hsd1.il.comcast.net) |
| 14:58.07 | agallo | stoffell_h, perhaps do you know how to do it in DialPlan? if i use Set(ALERT_INFO... does not work: i've to put 3 lines a) SET(ALERT_INFO b) Set(_ALERT_INFO and c) Set(__ALERT_INFO very funny :))) |
| 14:58.30 | brif8 | I can pastebin sip debug if that will help ? |
| 14:59.03 | stoffell_h | agallo: no idea, it is with the set thing.. there's some doc on voip-info that explains it, i've tried it once, a few months ago, on polycom it works. gxp: no way.. |
| 14:59.26 | brif8 | what is also odd is that SIP show peer 090 has ToHost incorrect and yet Addr->IP is correct ? |
| 14:59.28 | niter3 | hey guys.. will this work exten => 222,2,Voicemail(su222@default) - exten => 222,dial+101,Voicemail(sb222@default) |
| 14:59.40 | niter3 | exten => 222,2 |
| 14:59.41 | niter3 | sorry |
| 14:59.51 | niter3 | no i take that back |
| 15:00.11 | niter3 | if they get a busy signal calling will it jump to the sb one? |
| 15:02.31 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 15:05.27 | sxpert | ah, there... |
| 15:07.44 | *** join/#asterisk muppetmaster (n=jasongoe@81.184.73.169) |
| 15:08.07 | vlt | Hello. What could be a reason for "unable to join queue 'foo'" on only one extension (incoming CAPI). Other extensions (SIP) work fine. |
| 15:08.39 | n0n0x | hi, in which version AGENT function is implemented? |
| 15:12.19 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 15:15.41 | *** join/#asterisk teknoprep (n=teknopre@unaffiliated/teknoprep) |
| 15:16.05 | brif8 | <PROTECTED> |
| 15:17.17 | niter3 | my boss is an asshole |
| 15:17.19 | niter3 | whoa.. |
| 15:17.20 | Filar | rofl |
| 15:17.21 | niter3 | asshole |
| 15:17.25 | brif8 | If I set the host=ip_address not dynamic I get Username/auth name mismatch (as above). But in the sip.conf and SIP<macAddress>.cnf file I have the same username and password ? phone is a cisco 7960. Why would * report that the name is a mis match ? |
| 15:17.27 | Filar | dude |
| 15:17.59 | niter3 | ? |
| 15:18.31 | *** join/#asterisk eKo1 (n=eKo1@190.4.7.90) |
| 15:18.39 | *** join/#asterisk DarKnesS_WolF (n=wolf@212.12.250.34) |
| 15:19.10 | *** join/#asterisk _deg_ (n=deg@200.163.193.247) |
| 15:19.21 | brif8 | even sip show users has the correct username and secret/password ? |
| 15:19.56 | *** join/#asterisk muppetmaster (n=jasongoe@81.184.73.169) |
| 15:21.25 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 15:21.39 | *** join/#asterisk c4t3l (n=c4t3l@72.16.250.149) |
| 15:22.14 | *** join/#asterisk FlatFoot (n=simon@80.88.192.113) |
| 15:23.16 | vlt | Where can I tell asterisk to take a "#" from analog phone as "Enter/number finished"? |
| 15:23.26 | *** join/#asterisk MRH2 (n=Mr_happy@host-83-146-30-242.bulldogdsl.com) |
| 15:23.57 | bkw_ | you can't |
| 15:24.29 | MRH2 | hi - I am getting crashes with asterisk 1.2.x - can you folk confirm for a bug report it is make valgrind, attach backtrace and attache debug log |
| 15:24.59 | *** join/#asterisk DrukenHME (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 15:25.15 | vlt | What is extension "h"? |
| 15:25.21 | vlt | Hungup? |
| 15:26.39 | mut | genius! |
| 15:26.40 | MRH2 | and which category do I file it if I don't know the cause? |
| 15:26.55 | *** join/#asterisk jtexter3 (n=jtexter3@COX-66-210-197-34-static.coxinet.net) |
| 15:27.09 | jtexter3 | Anybody here have any experience with ChanSpy? |
| 15:27.26 | brif8 | Please, is anyone using a cisco 7960 with double nat ? |
| 15:27.51 | sxpert | double nat ??? one nat is crappy enough |
| 15:28.06 | MRH2 | just core asterisk? |
| 15:30.29 | brif8 | expert: I know, . I'm sure my problem with the 7960. Because I have seen it working on a siptura |
| 15:31.14 | *** join/#asterisk muppetmaster (n=jasongoe@81.184.73.169) |
| 15:32.46 | *** join/#asterisk Deeewayne (n=dwayne@ool-44c0d56e.dyn.optonline.net) |
| 15:33.41 | brif8 | in looking at sip debugs statements comparing phones that one is NAT and the other is not I see the tag user=phone is missing from the natted phone |
| 15:33.47 | brif8 | is there anyway to add this ? |
| 15:34.33 | Dr-Linux|work | anybody fimiliar with Realtime? |
| 15:35.45 | eKo1 | brif8: add it where? |
| 15:36.21 | jtexter3 | If I use ChanSpy to spy on a SIP extension, and the person on that extension speaks, I don't get anything. Is that normal? |
| 15:36.32 | brif8 | eKo1: to the sip register for this single phone ? |
| 15:36.45 | *** join/#asterisk brijn (n=bas@204.244.176.116.net-conex.com) |
| 15:36.48 | jtexter3 | If I play a file on the channel being spyed on, the spy can hear it, but not if the person speaks |
| 15:37.48 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 15:38.04 | brif8 | OPTIONS sip:600@192.168.2.232:5060;user=phone SIP/2.0 AND OPTIONS sip:090@67.9.15.186:5060 SIP/2.0 the first is a 7960 on the same LAN, while the second is a 7960 outside the LAN |
| 15:38.06 | eKo1 | brif8: do you know what that tag actually means? |
| 15:38.17 | sxpert | it's me or app_meetme doesn't compile anymore ? |
| 15:38.21 | brif8 | why is there no tag on the second one. it's the same type of phone |
| 15:38.23 | sxpert | (in trunk that is) |
| 15:38.28 | brif8 | eKo1: no |
| 15:38.53 | *** join/#asterisk cilly (i=pluto@unaffiliated/cilly) |
| 15:39.12 | eKo1 | Then you're guessing that adding that will fix your NAT problem. |
| 15:40.33 | brif8 | eKo1: yes I'm guessing. but I'm also looking / trying to learn why this is not working? I noticed this difference |
| 15:40.41 | vlt | How can I wait for the caller dialling an extension without Background()ing a sound file? |
| 15:40.43 | brif8 | eKo1: do you have any ideas ? |
| 15:41.02 | tzanger | hmm interesting |
| 15:41.19 | tzanger | I'm getting double-dtmf on zap channels (t100p to channel bank), only the first digit |
| 15:42.16 | *** join/#asterisk muppetmaster (n=jasongoe@81.184.73.169) |
| 15:42.26 | eKo1 | brif8: not really. |
| 15:44.57 | coppice | tzanger: do you have echo cancellation? |
| 15:45.09 | L-info | brif8, it wont solve your problem, but entering 'usereqphone=yes' in the general or peer context in your sip.conf will ensure its presence in your sip URI's |
| 15:45.24 | brif8 | [TK]D-Fender: don't you have double Nat expertise, and could help |
| 15:45.30 | tzanger | coppice: yes. I actually noticed relaxdtmf was in there, I've killed it and waiting for a restart |
| 15:45.39 | brif8 | L-info: ok any ideas what might work ? |
| 15:45.54 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 15:45.54 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:46.32 | coppice | tzanger: the DTMF decoder in * has little immunity to dial tone. it was developed on the assumption there would be echo cancellation to eliminate the dial tone. |
| 15:46.35 | L-info | brif8, get rid of the double NAT situation your server is having to work around would be my first suggestion |
| 15:46.53 | *** join/#asterisk marv[work] (n=timr@64.89.118.139) |
| 15:47.13 | coppice | tzanger: I have added a selectable dialtone filter to the DTMF decoder in spandsp, to make things more flexible. |
| 15:47.21 | tzanger | hmm, alright |
| 15:47.24 | tzanger | interesting |
| 15:47.47 | brif8 | L-info: yeah that would be nice. Let me ask if the phone is on DMZ, does that still count as NAT from an */voip point of view ? |
| 15:48.10 | brif8 | also I added usereqphone=yes and it does not show up in the OPTIONS line |
| 15:48.10 | L-info | yep |
| 15:48.20 | coppice | considering how unreliable the echo cancellers still are, I'm surprised dial tone on the first digit isn't a bigger source of problems |
| 15:48.49 | tzanger | this is the first I've run across it |
| 15:49.47 | sxpert | hah. no, found it... the stupid /usr/src/linux link had changed ;D |
| 15:49.59 | sxpert | duh |
| 15:50.24 | sxpert | the dang configure should use the output of `uname -a` |
| 15:50.52 | brif8 | L-info: then I can't loose the Nat. I'm on a Satellite Modem. Which gives me a single static IP and the Asterisk box is behind a firewall |
| 15:51.47 | brif8 | L-info what I really make no sense is that phone shows as UNREACHABLE, yet I can make outbound calls from the phone and they go through the * server fin no problems |
| 15:52.21 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 15:52.38 | *** join/#asterisk DrukenHME (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 15:53.36 | *** join/#asterisk muppetmaster (n=jasongoe@169.red-81-184-73.user.auna.net) |
| 15:53.45 | eKo1 | I'm trying to compile a module. I #include <asterisk/options.h> because I need the VERBOSE_* #define's but gcc is telling me there are syntax errors in it. WTF? |
| 15:53.58 | jhb | Hi *, I have my sipusers in a mysql, and now I can register. Now I wonder how I make them available as extensions. I have the feeling that the switch => Realtime is for that, but I can't figure it out. Any help? |
| 15:54.46 | [TK]D-Fender | bri8 : Your phone isn't responing within the quailfy timeout and then is considered unreachable..... you need to either give it a fixed IP, change the qualify timeout and/or change the rergistration dely on the phone. |
| 15:55.10 | file | [TK]D-Fender: ! ! ! |
| 15:55.27 | [TK]D-Fender | file : I don't want relationship! |
| 15:55.38 | file | [TK]D-Fender: wasabi? |
| 15:55.54 | *** join/#asterisk bmg505 (n=leon@dsl-146-22-87.telkomadsl.co.za) |
| 15:56.47 | Strom_C | hi |
| 15:56.51 | *** join/#asterisk SplasPood (n=jwb@gate.lga2.us.voxel.net) |
| 15:56.51 | *** join/#asterisk joerns (i=duckbert@dslb-082-083-151-039.pools.arcor-ip.net) |
| 15:57.00 | vlt | When "X" is [0-9], "N" is [1-9], "Z" is [2-9] and "." is one _or more_ following chars, what stands for one single char in exten patterns? |
| 15:57.09 | *** join/#asterisk Antax (i=StanG@nsc69.38.12-158.newsouth.net) |
| 15:57.20 | Strom_C | vlt: Z is 1-9, N is 2-9 |
| 15:57.26 | Antax | greetings |
| 15:57.43 | Strom_C | vlt: and all of those stand for a single digit except . |
| 15:58.16 | sxpert | you can also use [3-5] |
| 15:58.16 | sxpert | for instance |
| 15:58.34 | Strom_C | [TK]D-Fender: do you know of a polycom autoconfiguration wizard or somesuch? this XML file is going to take an hour to configure |
| 15:58.37 | [TK]D-Fender | sxpert: I suspect he's thinking alpha-numeric... |
| 15:58.41 | sxpert | hah |
| 15:58.54 | Lyfe | anyone have any ideas how i can have a span which has the first 11 channels active, and the other 12 are in lockup state, with the d channel seeming to work as well? (i'm getting things into a pastebin link right now, but, i don't know that it'll help much) |
| 15:59.03 | [TK]D-Fender | Strom_C: Should take 10 minutes from scrath for sip.cfg and your phone template.... |
| 15:59.14 | [TK]D-Fender | Strom_C: And its time well spent regardless |
| 15:59.24 | vlt | [TK]D-Fender: You're right. So I have to use [0-9*#] for normal phones? |
| 15:59.25 | Strom_C | [TK]D-Fender: I'm looking at this sample file. It's a mess. |
| 15:59.34 | Antax | any1 want to answer some newbie questions? :) |
| 15:59.38 | joerns | cheers everybody. I'd like to use asterisk not for VoIP but for sending other data streams, say ssh. can somebody point me to documentation that would explain how to send whatever data using asterisk? |
| 15:59.42 | brif8 | [TK]D-Fender: if I give is a fixed IP and not =dynamic, then I get a Username/Auth Mismatch. Which is not true. Both the username, secret in sip.conf agrees with the setting in SIP<macaddress>.cnf |
| 15:59.55 | [TK]D-Fender | Strom_C: : the one that comes with the firmware? |
| 15:59.59 | Strom_C | [TK]D-Fender: yes |
| 16:00.25 | *** join/#asterisk anthonyl (n=anthonyl@c-71-57-43-221.hsd1.il.comcast.net) |
| 16:00.32 | Antax | how much hardware would i be looking at for a small office setup? |
| 16:00.40 | [TK]D-Fender | then its the IP being reported by the phone.... public/private issue |
| 16:00.47 | Antax | the pc and a TDM40B? |
| 16:00.55 | Strom_C | Antax: depends on your requirements |
| 16:01.06 | *** join/#asterisk Fraeggl (n=Fraeggl@rkom.r-kom.de) |
| 16:01.07 | Antax | ive got 6 analog phones |
| 16:01.21 | [TK]D-Fender | Antax: What/how many phones, what/how many lines, etc... |
| 16:01.23 | Antax | nwo with 2 lines |
| 16:01.31 | Antax | now |
| 16:01.31 | FlatFoot | afternoon all |
| 16:02.02 | Antax | but the only isp connection availabel is dsl |
| 16:02.09 | Antax | will it work through dsl? |
| 16:02.25 | Strom_C | Antax: TDM2400P will take care of your telephone sets and your telephone lines |
| 16:02.28 | [TK]D-Fender | Antax: So you just want to replace your current PBX (not even thinking about the outside world here...) |
| 16:02.55 | Antax | right now i dont have a pbx just 2 copper lines pulled into the building |
| 16:03.05 | sxpert | grrr why doesn't app_meetme compile in trunk ? I have trunk zaptel too as I should... |
| 16:03.11 | Antax | with 2 line phones |
| 16:03.18 | [TK]D-Fender | Antax: So all your phones support 2 lines right now? |
| 16:03.24 | Antax | right |
| 16:03.28 | rpm | sxpert: did you install zaptel? it needs to be able to find the zaptel.h file for dependencies |
| 16:03.41 | *** join/#asterisk joelsolanki (i=joelsola@202.160.161.94) |
| 16:03.42 | file | you need the newest zaptel to build trunk right now |
| 16:03.43 | rpm | you ran make menuselect right? |
| 16:03.46 | sxpert | rpm: yeah I installed it, reloaded the ztdummy from the new one |
| 16:03.52 | sxpert | rpm: yeah. it's selected |
| 16:03.54 | [TK]D-Fender | Antax: You looking to keep your phonesor are you willing to replace them? |
| 16:04.04 | Antax | depends on the cost |
| 16:04.05 | rpm | sxpert: is it giving a compiler error or completely skipping it? |
| 16:04.07 | sxpert | file: that's what I goes |
| 16:04.19 | sxpert | rpm: it visibly skips it |
| 16:04.19 | brif8 | [TK]D-Fender: correct it shows "Registration from '<sip:090@ASTERIK_INT_IP;user=phone>' failed for 'REM_PHONE_PUBLIC_IP' - Username/auth name mismatch" any ideas how to correct |
| 16:04.22 | [TK]D-Fender | Antax: Where are you located? |
| 16:04.26 | Antax | TN |
| 16:04.30 | joelsolanki | Hi all. has anybody used didx.org with asterisk. I m trying to make them work but it is not working. |
| 16:04.34 | file | sxpert: do ./configure again in asterisk |
| 16:04.37 | Lyfe | to go with my question, http://pastebin.com/780696 .. i have a pri span that the first 11 channels come up apparently fine, and the rest of the 23 b-chans do not, and we're rather befuddled. |
| 16:04.38 | sxpert | just did |
| 16:04.41 | sxpert | after make clean |
| 16:05.04 | *** join/#asterisk hmmhesays (n=hmmhesay@24-117-135-28.cpe.cableone.net) |
| 16:05.07 | [TK]D-Fender | brif8: PB your phone definition. |
| 16:05.14 | Antax | USA, TN, Tennessee Ridge |
| 16:05.20 | Antax | its a small town |
| 16:05.23 | hmmhesays | ok, do I need res_crypto for anything? i'm trying to slim this asterisk down and it is huge |
| 16:05.24 | file | and did it say - checking for ZT_TONE_DTMF_BASE in zaptel.h... yes |
| 16:05.57 | joelsolanki | Can anybody help me out with didx + asterisk ? |
| 16:06.51 | eKo1 | hmmhesays: chan_iax.so using res_crypto |
| 16:07.17 | brif8 | [TK]D-Fender: http://pastebin.com/780701 |
| 16:07.42 | sxpert | file: I do get "checking for ZT_TONE_DTMF_BASE in zaptel.h... yes" |
| 16:07.59 | file | cool, then both trunk and zaptel are in sync fine... |
| 16:08.04 | Boggi | what's a good sip client for linux? |
| 16:08.22 | vlt | Boggi: asterisk? |
| 16:08.29 | sxpert | ok. it's selected there |
| 16:08.35 | Boggi | vlt, client ? softphone? |
| 16:08.42 | [TK]D-Fender | brif8: Did you used to have "qualify=yes (or some #?)) |
| 16:09.22 | *** join/#asterisk fdepp (n=fdepp@tsf-wpa-1-174.epfl.ch) |
| 16:09.28 | vlt | Boggi: was joking (btw asterisk is indeed a good sip client). Did you try kphone? |
| 16:09.39 | brif8 | [TK]D-Fender: yes it was =yes not I'm trying 8000 still has unreachable |
| 16:09.45 | file | sxpert: I just did a clean trunk build, and meetme built fine |
| 16:09.49 | sxpert | hmm |
| 16:09.51 | joelsolanki | anybdy using didx + asterisk ? |
| 16:09.56 | sxpert | mebbe I had crap laying around |
| 16:10.44 | brif8 | not->now |
| 16:10.51 | [TK]D-Fender | brif8: Ok, I'm not sure what to say in your case... |
| 16:11.08 | [TK]D-Fender | joelsolanki: We heard you the first time, don't spam it.... |
| 16:11.10 | *** join/#asterisk WEH (n=bob@pool-71-250-60-139.nwrknj.east.verizon.net) |
| 16:11.15 | WEH | hey |
| 16:11.24 | brif8 | [Tk]D-Fender: any suggestions, links or things that may be incorrect ? |
| 16:11.27 | sxpert | file: ah. it just worked... guess svn trees fsck themselves over time ;( |
| 16:11.33 | WEH | what would i need to do to enable the manager api? |
| 16:11.43 | FlatFoot | not exactly the right channel but i have been searching for a good USB phone to use with * any suggestions ???? |
| 16:11.43 | WEH | other than editing manager.conf |
| 16:12.05 | *** part/#asterisk jtexter3 (n=jtexter3@COX-66-210-197-34-static.coxinet.net) |
| 16:15.45 | sxpert | file: why the h*** does make install rebuilds everything ? |
| 16:16.09 | *** part/#asterisk WEH (n=bob@pool-71-250-60-139.nwrknj.east.verizon.net) |
| 16:16.10 | file | it depends. |
| 16:16.34 | MRH2 | is make make don-optimize the same as "make valgrind" |
| 16:20.20 | *** join/#asterisk QbY (n=Kelvin@cm-64-221-172-232.dhcp.southerncoastalcable.net) |
| 16:20.44 | QbY | Has anyone experienced trouble with inbound IAX calls, Queues, and Version 1.2.11? |
| 16:22.00 | joelsolanki | Does this forum help for asterisk + didx ? |
| 16:23.10 | *** join/#asterisk l-fy (n=pchitesc@yate/developer/l-fy) |
| 16:23.11 | l-fy | hello |
| 16:23.24 | l-fy | can i use asterisk to record passive 1 t1 using a 2 t1 card? |
| 16:23.29 | hmmhesays | do I need chan_modem for anything if i'm just using sip? |
| 16:23.57 | eKo1 | hmmhesays: no |
| 16:25.49 | *** join/#asterisk mountainm2k (n=mountain@216.87.64.218) |
| 16:27.33 | [TK]D-Fender | l-fy: Sure, just take a call in on 1 port and spit it out the other end. Thats it. |
| 16:28.00 | *** join/#asterisk WEH (n=bob@pool-71-250-60-139.nwrknj.east.verizon.net) |
| 16:28.12 | l-fy | [TK]D-Fender > passive |
| 16:28.20 | l-fy | it means that asterisk is not in the middle |
| 16:28.41 | l-fy | it just listening for the tx from both sides and than connects them |
| 16:28.47 | [TK]D-Fender | l-fy: Well I guess my description was more like "transparent"......... but for what you just added... NO |
| 16:29.24 | [TK]D-Fender | l-fy: * is really so much in the way as "yeah if it goes down, traffic =toast" |
| 16:29.24 | WEH | how would i configure * so that a script can call two phones and bill as if one called the other? |
| 16:29.48 | l-fy | the point is not to lose traffic |
| 16:29.48 | eKo1 | WEH: bill? |
| 16:30.37 | WEH | let me be more specific |
| 16:30.44 | mountainm2k | Hey, so I have an extension that does Answer() and then MusicOnHold() -- great for transfering sales guys to... |
| 16:30.53 | mountainm2k | Except it doesn't actually play the music... |
| 16:31.11 | WEH | eKo1: letting the server be B and the clients A and C: B -> A, B -> C, cdr shows A -> C |
| 16:32.37 | WEH | being triggered from a script |
| 16:38.14 | *** join/#asterisk swytch (n=root@LNeuilly-152-22-86-193.w193-251.abo.wanadoo.fr) |
| 16:39.39 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 16:39.41 | swytch | anyone have a comment on the following? I will try to mount a Digium card just to pass the call out of it and in again. All this to see if it improves the voice quality of my sip-to-sip calls. |
| 16:40.14 | mountainm2k | what's the voice quality issue on sip-to-sip? |
| 16:40.18 | mountainm2k | are they on your LAN? |
| 16:41.23 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 16:41.58 | swytch | mountainm2k: calls are coming from callshops on adsl (50ms). they leave * to hit a mera just besides it that transcodes and sends (100ms) over inet to various pstn. |
| 16:44.47 | *** join/#asterisk daysmen3 (n=primus@host86-139-212-246.range86-139.btcentralplus.com) |
| 16:45.36 | Antax | do i need an SPA-2002 for ea phone? |
| 16:45.52 | Qwell | ea phone? |
| 16:45.58 | Antax | each phone |
| 16:46.18 | mountainm2k | swytch: SIP over the internet is really hit or miss... Heh, you're at the mercy of all the ISP's in between... But I don't think going out and back in an analog board is going to help you |
| 16:46.18 | Antax | im thinking yes but just want to verify |
| 16:46.26 | mountainm2k | swytch: But I've been wrong before... |
| 16:46.32 | WEH | letting the asterisk box / script server be B and the clients A and C, how can i get B to initiate calls to A and C in such a way that the cdr is written as A-> C? :: B -> A, B -> C ==> cdr shows A -> C |
| 16:47.17 | swytch | mountainm2k: it goes analogphone--->augiocodes118fxs-(711u/20)50ms-->asterisk-(711u/20)-->mera-(729b/40)100ms-->pstn and it gets walky talky style |
| 16:48.55 | swytch | mountainm2k: but this works fine: analogphone--->augiocodes118fxs-(711u/20)50ms-->asterisk-(711u/20)-->pstn |
| 16:49.30 | eKo1 | WEH: what does the CDR show? |
| 16:50.38 | swytch | mountainm2k: and the really weird thing is that analogphone--->augiocodes118fxs-(711u/20)50ms-->voicemaster-(711u/20)-->mera-(729b/40)100ms-->pstn works fine too. i just substituted asterisk with voicemaster. ??! |
| 16:51.02 | *** join/#asterisk ramtha (n=tk@p5088BE2E.dip0.t-ipconnect.de) |
| 16:51.07 | ramtha | hey |
| 16:51.27 | ramtha | some carrier whants to send us traffic to our asterisk |
| 16:51.32 | ramtha | signaling works |
| 16:51.35 | ramtha | rtp not |
| 16:51.39 | ramtha | he says |
| 16:51.50 | ramtha | From what I can see the gateway supports non symmetric rtp, where we don’t. |
| 16:51.50 | ramtha | We need the gateway to reply with one codec from the list in the original |
| 16:51.50 | ramtha | invite. |
| 16:51.54 | mountainm2k | swytch: Dunno what to tell you -- lot of transcoding in there... |
| 16:52.06 | eKo1 | ramtha: firewall? |
| 16:52.12 | ramtha | can any one explain what he meen? |
| 16:52.20 | ramtha | no firewall is ok |
| 16:52.29 | ramtha | no ports where blocked |
| 16:52.43 | ramtha | i can not see any rtp packets comming to our gateway |
| 16:52.49 | eKo1 | What type of signaling are you using? |
| 16:52.55 | ramtha | sip |
| 16:53.17 | swytch | mountainm2k: only the mera trancodes. for all calls. when i place voicemaster inplace of asterisk the sound gets better. it kind of strokes out the rtp. its magic. is it possible this is the case? |
| 16:53.55 | *** join/#asterisk postel (n=jp@unaffiliated/postel) |
| 16:54.00 | eKo1 | ramtha: strange |
| 16:54.25 | WEH | the cdr shows two calls |
| 16:54.31 | WEH | B-> A and B-> C |
| 16:54.35 | WEH | i want it to show A->C |
| 16:54.51 | WEH | how would i configure asterisk and craft a script so that it shows A -> C? |
| 16:55.23 | *** join/#asterisk ginvent (n=ginvent@adsl-63-199-244-27.dsl.sndg02.pacbell.net) |
| 16:56.09 | ginvent | Ok, so anyone have a good voip provider that will list me in the 411 and whitepages? teliax doesn't |
| 16:56.11 | eKo1 | Would you pastebin those two CDRs somewhere for me to look at? |
| 16:57.00 | rpsgb | I have a seemingly strange problem I was hoping one of you guys could help me out with... |
| 16:57.00 | eKo1 | I think there is something wrong with options.h in braches/1.0. |
| 16:57.38 | eKo1 | Welcome to the club. |
| 16:58.22 | rpsgb | if I dial "number a" using 'number b' as callerID and connect to the following context exten => s,1,Dial(SIP/${CALLERIDNUM}@voipdiscount,30,r) why don't I get voice? |
| 16:59.20 | rpsgb | It calls both numbers, the log says that it's attempting a native bridge, but I don't get voice. It's not a NAT issue because it works if I park the first call and answer the second |
| 16:59.56 | [TK]D-Fender | rpsgb: Actually that sounds EXACTLY like a NAT issue. |
| 17:00.12 | rpsgb | oops |
| 17:00.22 | rpsgb | but why then would the parked call work? |
| 17:00.30 | *** part/#asterisk QbY (n=Kelvin@cm-64-221-172-232.dhcp.southerncoastalcable.net) |
| 17:01.07 | rpsgb | dial 'numberA' into parking extension, dial 'numberB' into 701 |
| 17:01.11 | rpsgb | is fine |
| 17:02.31 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:02.35 | blitzrage | hey all |
| 17:02.39 | rpsgb | hi |
| 17:02.53 | blitzrage | just curious what free SIP client people are liking in Linux nowadays? |
| 17:02.58 | blitzrage | still X-Lite? |
| 17:03.06 | rpsgb | that's what I'm using |
| 17:04.15 | eKo1 | X-Lite is OK. |
| 17:04.42 | ginvent | You have to be kidding me. I can't get my business voip line listed in 411... jeez... now I have to go back and get an analog line... |
| 17:04.58 | hmmhesays | ok what is the best way to figure why asterisk is segfaulting? |
| 17:05.34 | blitzrage | probably use gdb to look at the core file |
| 17:05.47 | eKo1 | hmmhesays: gdb backtrace |
| 17:06.16 | [TK]D-Fender | blitzrage: Ekiga. Does Video, has transfer, etc... |
| 17:06.30 | Qwell | and most importantly, it does h323! |
| 17:06.51 | [TK]D-Fender | Qwell: If you said that other unspeakable protocol you could STAY there! ;) |
| 17:07.29 | blitzrage | [TK]D-Fender: doesn't support a # in the username |
| 17:07.51 | blitzrage | might have been fixed since I reported the bug a while ago I'd hope |
| 17:08.39 | [TK]D-Fender | blitzrage: Dev side is better than it used to be I hear, though I've never actually used it yet |
| 17:08.49 | Qwell | [TK]D-Fender: What, skinny? ;) |
| 17:08.51 | blitzrage | yah... trying to use minisip right now |
| 17:09.23 | [TK]D-Fender | Qwell: Like Iron Maiden said... "Run too the hills!!!!!!" |
| 17:09.45 | file | yay I updated my PBX to trunk |
| 17:09.45 | blitzrage | another accident outside my house.. |
| 17:09.53 | Qwell | blitzrage: You need to...move |
| 17:09.55 | *** join/#asterisk pollohawk (n=pollohaw@mmail.picksend.com) |
| 17:10.09 | Qwell | like, more than 10 ft from the freeway, might be a better option |
| 17:10.09 | file | blitzrage: stop killing people |
| 17:10.15 | *** join/#asterisk anthonyl (n=anthonyl@adsl-68-250-57-22.dsl.chcgil.ameritech.net) |
| 17:10.33 | blitzrage | Qwell: I just moved here! :) It's because it's the busiest highway in Toronto (goes right through city centre) |
| 17:10.40 | Dr-Linux|work | i'm fedup of this :( |
| 17:10.40 | Dr-Linux|work | Aug 31 09:53:27 WARNING[18016]: chan_zap.c:6105 ss_thread: CallerID returned with error on channel 'Zap/2-1' |
| 17:10.42 | brif8 | ok made progress. I got the phone to show with 67ms and when I dial the extensions it rings. but when I pick up the caller still hears ringing and asterisk reports and error |
| 17:10.52 | Dr-Linux|work | why this happens with FXO lines? |
| 17:11.09 | brif8 | Maximum retries exceeded on transmission 6a16b3f61a308adf4392f6d3261addbf@71.41.50.162 for seqno 102 (Critical Request) |
| 17:11.12 | Dr-Linux|work | sometime callerID works sometime not? |
| 17:11.14 | blitzrage | I need to setup some webcams |
| 17:11.15 | Dr-Linux|work | any clue? |
| 17:11.24 | Qwell | blitzrage: accidentcam.com? |
| 17:11.25 | blitzrage | too bad wireless cams are still so expensive |
| 17:11.39 | file | blitzrage: make some extra $$$ d00d |
| 17:11.40 | blitzrage | Qwell: pretty much -- fatal motorcycle accident just a couple days ago |
| 17:11.54 | blitzrage | trafficanddancing.com |
| 17:12.18 | Dr-Linux|work | Qwell, :) |
| 17:14.08 | *** join/#asterisk ginvent (n=ginvent@adsl-63-199-244-27.dsl.sndg02.pacbell.net) |
| 17:15.04 | *** join/#asterisk ginvent (n=ginvent@adsl-63-199-244-27.dsl.sndg02.pacbell.net) |
| 17:15.39 | ginvent | Anyone suggest a good voip provider for asterisk that will list numbers in 411? Sunrocket does, but they are a pain in the butt. |
| 17:18.32 | Dr-Linux|work | bye bye |
| 17:19.35 | *** join/#asterisk nain (i=nain@61.5.137.42) |
| 17:19.55 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 17:20.52 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 17:20.53 | *** mode/#asterisk [+o mog] by ChanServ |
| 17:21.31 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.80.244) |
| 17:22.25 | brif8 | [TK]D-Fender: I got the phone to ring when dialed and now also shows a valid qualify 68ms. But the voice doesn't go through ? |
| 17:23.31 | [TK]D-Fender | bri8 : Your phone is still natted right? |
| 17:23.38 | brif8 | [TK]D-Fender: yes |
| 17:24.35 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 17:24.50 | brif8 | being a cisco phone I had to set additional settings in SIP<maCAddress>.cnf nat_enable: nat_address nat_receved_processing |
| 17:25.20 | brif8 | sip now works but RTP is failing ? |
| 17:25.35 | brif8 | or voice doesn't go through |
| 17:28.34 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 17:28.45 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 17:28.45 | *** mode/#asterisk [+o mog] by ChanServ |
| 17:32.27 | nain | Hi i am trying to configure asterisk-stat-v2 but unable to connect with database: getting Database error: pconnect(localhost, root, $Password) failed. |
| 17:32.59 | *** join/#asterisk Gregabyte (i=greg@nat/digium/x-abf60b590ab4a9fb) |
| 17:34.49 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 17:34.57 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 17:35.23 | *** join/#asterisk Gregabyte (i=greg@nat/digium/x-827fbb7312e2375e) |
| 17:35.37 | *** join/#asterisk foxmjay (n=root@ll81-144-114-192-81.ll81.iam.net.ma) |
| 17:35.50 | eKo1 | nain: that has nothing to do with * proper |
| 17:37.11 | *** join/#asterisk Gunnar (n=gunnar@nat.sigmasoft.com) |
| 17:37.43 | *** join/#asterisk Gregabyte (i=greg@nat/digium/x-02e135d8994a7b02) |
| 17:37.55 | nain | eKo1: yes i know also asterisk is logging cdr in mysql with same user and passwd, but unable to connect using asterisk-stat with mysql |
| 17:39.17 | foxmjay | plz , i have strange problem with CHANNEL STATUS ZAP/1-1 agi command, it gives me -1 instade of giving me 0, but the others status work fine ? |
| 17:40.00 | eKo1 | nain: as I said, that is NOT an * problem |
| 17:40.30 | brif8 | in the SIP debug where do you find the port number for RTP ? |
| 17:40.34 | nain | eKo1: ok i understand |
| 17:41.01 | nain | eKo1: any other reason or idea or can let me know where mysql has problem |
| 17:41.13 | eKo1 | try #mysql |
| 17:41.33 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 17:41.41 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
| 17:41.49 | eKo1 | brif8: good question. You will find the answer to that in the SIP RFC. |
| 17:43.24 | *** join/#asterisk angom (n=angom@red-corp-200.79.133.82.telnor.net) |
| 17:43.44 | nain | ok |
| 17:44.06 | foxmjay | fwrite($stdout,"CHANNEL STATUS ZAP/1-i \n"); : gives me the result -1, but the line is OK |
| 17:50.42 | *** join/#asterisk justinu|laptop (n=Justin@12.44.122.130) |
| 17:51.51 | *** join/#asterisk yogurt2ungue (n=yogurt2u@24-48-231-201.fibertel.com.ar) |
| 17:56.26 | jbalcomb | User says "Every time I try to pick a call up from being on hold it goes directly to speaker instead of to my handset" |
| 17:56.36 | jbalcomb | Answers or clues? phones are grandstream gxp-2000 |
| 17:58.50 | *** join/#asterisk Avalone (n=Avalone_@83.239.191.113) |
| 18:01.07 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:01.36 | CunningPike | [TK]D-Fender: Do you know what the "allow everything" dialstring is for Polycoms? Is it x+#? |
| 18:05.14 | *** join/#asterisk anthonyl (n=anthonyl@adsl-68-252-207-35.dsl.chcgil.ameritech.net) |
| 18:06.08 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 18:06.47 | *** join/#asterisk QbY (n=Kelvin@cm-64-221-172-88.dhcp.southerncoastalcable.net) |
| 18:07.05 | brif8 | eKo1: care to share which line it is in ? |
| 18:08.13 | *** join/#asterisk gheorghe (n=gheorghe@86.123.192.142) |
| 18:08.14 | *** join/#asterisk zekeonfire (n=zekeonfi@208.186.65.172) |
| 18:08.32 | awannabe | with 1FB Digium cards is it possible to have it use another port if the primary is alread in use, and not have the phone user have any ideal about it? |
| 18:08.34 | *** part/#asterisk gheorghe (n=gheorghe@86.123.192.142) |
| 18:08.40 | *** join/#asterisk gheorghe (n=gheorghe@86.123.192.142) |
| 18:08.42 | QbY | Ok, this is the 100th time it has happened, but asterisk shows a ton of open channels (none of them there), unable to do show channels, or show queues again as it is dead now.. Before I kill -9 it, is there anything that will tell me what is wrong with this thing? |
| 18:12.50 | Strom_C | awannabe: what do you mean 1FB> |
| 18:12.50 | awannabe | just analog POTS lines |
| 18:12.50 | Strom_C | for outbound? |
| 18:12.51 | awannabe | yeah |
| 18:12.51 | Strom_C | yeah, you put the zap channels in a group, and then you dial Zap/Gx/number |
| 18:12.51 | mercestes | Any clue how I can correct this?: WARNING[25815]: interface.c:215 decodeMP3: Junk at the beginning of frame 49443302 |
| 18:12.51 | Strom_C | asterisk will take care of hunting |
| 18:12.51 | awannabe | and if the first one is busy, it just dials the next? |
| 18:12.52 | awannabe | ahh, ok |
| 18:12.52 | awannabe | sweetness, i knew asterisk was smart! lol |
| 18:12.52 | Strom_C | :) |
| 18:12.52 | Strom_C | but what is "1FB"? which card do you have? |
| 18:12.52 | awannabe | well 1FB is the code/slang everyone here uses, heh, it will be a TDM400O with FXO ports |
| 18:13.06 | Strom_C | i've been hanging around here a long time and ive never heard of 1FB :) |
| 18:13.16 | awannabe | maybe its a arizone only thing..lol |
| 18:13.17 | Strom_C | unless thats a nortel line class code |
| 18:13.37 | Strom_C | I remember 1FR is single party flat-rate service in DMS |
| 18:13.57 | Strom_C | but id have to get my DMS manuals out to remember the others |
| 18:14.08 | awannabe | and i cant remember what the FB stands for, its something business |
| 18:14.27 | awannabe | cause also called a 1FH for home (i think its 1FH) lol |
| 18:14.34 | Strom_C | weird |
| 18:14.40 | awannabe | crap, stuff direct thru digium is hella pricey |
| 18:14.50 | Strom_C | go through one of their resellers |
| 18:15.10 | awannabe | yeah, i was just curious of the price difference |
| 18:15.14 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 18:15.33 | crochat | When I call from another IP phone in my LAN, the CallerID is displayed on my analog phone which is connected on the AG168V FXS ! But when I receive an incoming call through my SIP provider, Asterisk tell the CallerID on all the IP phones, but on the phone behind the AG168V, it's just displayed "Incoming call" !! WHY ?? |
| 18:15.40 | Strom_C | that, i believe is intentional; with the volume of cards digium sells, handling direct orders would be bonkers |
| 18:15.53 | awannabe | heh |
| 18:16.15 | awannabe | wow 150 for a single port FXO card, bummer |
| 18:16.30 | Strom_C | awannabe: how much were you expecting to spend? |
| 18:17.05 | awannabe | i was gonna get one to mess around with, but to pricey, heh |
| 18:17.11 | syzygyBSD | it should be free right? i mean asterisk is free isnt' it? |
| 18:17.14 | teknoprep | trying to setup an outbound route with a dialing prefix so if the dialing prefix is specified it uses a specific outgoing trunk... i setup the outbound route dial pattern to ... 9699|1NXXNXXXXXX ... am i doing anything wrong becuase i can't make calls outgoing that trunk when i dial that prefix |
| 18:17.37 | Strom_C | teknoprep: this is not freepbx support |
| 18:17.43 | Strom_C | teknoprep: please go to #freepbx |
| 18:18.18 | awannabe | haha, yeah free digium cards!!! wouldnt that be nice |
| 18:18.26 | awannabe | the markster has to eat to afterall i guess :) |
| 18:18.26 | Strom_C | awannabe: if you want something cheap to mess around with, get one of the x100p cards that's floating around out there. Just a warning though: you get what you pay for ;) |
| 18:18.35 | syzygyBSD | teknoprep: | isn't a valid character for extensions |
| 18:18.42 | Strom_C | syzygyBSD: he's using freepbx |
| 18:18.43 | syzygyBSD | well, it is, but you can't dial it |
| 18:19.07 | teknoprep | syzygyBSD its a dialing prefix in a dial pattern for an outbound route |
| 18:19.12 | awannabe | "our flagsip product" lmao |
| 18:19.27 | teknoprep | syzygyBSD its supposed to strip that |
| 18:19.33 | *** part/#asterisk gheorghe (n=gheorghe@86.123.192.142) |
| 18:19.36 | Strom_C | ~freepbx |
| 18:19.40 | jbot | it has been said that freepbx is the Microsoft BOB of PBXes and NOT supported here! People using it should join #freepbx (FreePBX is the new name of AMP) |
| 18:19.52 | syzygyBSD | you use a lot of words that don't matter (at least to asterisk) |
| 18:20.20 | *** join/#asterisk deb_user (n=debian_u@albuquerque.agroinnovations.com) |
| 18:20.22 | awannabe | x100p is only a single port max to, what a cheapo |
| 18:20.35 | awannabe | that looks like it will blow up, not dial a call, lol |
| 18:20.49 | Strom_C | hahahah |
| 18:21.14 | awannabe | wait, 19 bucks though? |
| 18:21.22 | syzygyBSD | so is jbot making fun of Mrs Gates? |
| 18:21.22 | benjk | x100p is a zaptel timing pci card for those who have no zaptel cards and don't want to use ztdummy |
| 18:21.41 | *** join/#asterisk novafirst (n=kosta@wrt1.niclab.com) |
| 18:22.22 | awannabe | the x100p is ideal for: create your own calling card business, and much more! |
| 18:22.24 | awannabe | lmao |
| 18:22.32 | Strom_C | hahahaha |
| 18:22.56 | Strom_C | thats like saying you can build an entire office supply store with nothing but two boxes of Bic pens |
| 18:23.59 | awannabe | haha |
| 18:23.59 | awannabe | http://www.digitnetworks.com/store/product_info.php?products_id=28&osCsid=0b10ec805ff2e968b5fec8c258c83014 |
| 18:23.59 | awannabe | it has several uses |
| 18:23.59 | Strom_C | it slices |
| 18:23.59 | benjk | absolutely it does |
| 18:23.59 | Strom_C | it dices |
| 18:24.00 | benjk | you can wipe your ass |
| 18:24.00 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 18:24.00 | Strom_C | it makes echo problems the likes of which have never been seen before! |
| 18:24.02 | awannabe | im half tempted to buy one for 19 bucks, lol |
| 18:24.10 | benjk | you can use it as a placeholder |
| 18:24.22 | benjk | you can use it as a timer |
| 18:24.27 | benjk | at least 3 uses |
| 18:24.30 | awannabe | this is the one the old voip company had and told me to "test" with, haha |
| 18:24.57 | awannabe | the digum cards (non PRI) can act as a timing source, correct? |
| 18:25.06 | Strom_C | yes |
| 18:25.09 | benjk | sure |
| 18:25.27 | awannabe | speaking of high quailty hardware, i tell ya the grandstream 286 ATA is WORLD CLASS |
| 18:25.58 | Strom_C | oh, don't stop there. EVERYTHING granstream makes is top-quality!!!!! |
| 18:26.00 | QbY | Are there any * Devs in here? |
| 18:26.05 | benjk | I dont think so, its not heavy enough as a paper weight |
| 18:26.07 | *** join/#asterisk _alex_mx_ (n=_alex_mx@200.78.229.18) |
| 18:26.10 | awannabe | its got a blinking light on it, and the web interface was built by a kindergarten kid with frontpage |
| 18:26.17 | Strom_C | AHAHAHHAHHA |
| 18:26.46 | awannabe | i got one for my parents so they can have a phone is their other house...i was like, yeah it will work good for that, hah |
| 18:26.58 | benjk | like the x100p, its not big enough as a door stop |
| 18:27.06 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 18:27.19 | syzygyBSD | qby I think there is an asterisk dev channel, why do you ask? |
| 18:27.27 | awannabe | the digium cards work well as a data pass thru, or is that all a myth? |
| 18:27.42 | Strom_C | what do you mean "data pass through"? |
| 18:27.45 | QbY | i'm reading a mailing list that is about an error close to mine, and it says contact a dev on irc for a proper backtrace |
| 18:27.46 | Qwell | hdlc? |
| 18:27.52 | awannabe | for a frac t1 |
| 18:27.57 | Strom_C | yes, that's easy |
| 18:28.09 | Strom_C | hi Qwell |
| 18:28.22 | awannabe | so it just gets passed to the ethernet interface then? |
| 18:29.01 | syzygyBSD | awannabe: I had HDLC and a fractional T1 setup on 2 boxes |
| 18:29.27 | awannabe | so the sangomas are really better then? |
| 18:29.33 | awannabe | i didnt know casue they werent digium |
| 18:29.38 | syzygyBSD | it seemed like it worked better on the digium card then the other one i had |
| 18:30.44 | awannabe | heh |
| 18:30.50 | *** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it) |
| 18:31.22 | awannabe | wow, and its only 475 |
| 18:31.26 | *** part/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it) |
| 18:31.31 | awannabe | doesnt say it has hardware echo cancelation thoguh |
| 18:31.50 | CunningPike | awannabe: 475 and all your hair getting it configured....... |
| 18:32.01 | Strom_C | awannabe: you only generally need hardware echo can on really long T1s |
| 18:32.26 | CunningPike | Or really crappy hybrid circuits |
| 18:32.28 | Aurs | does anyone have a list over directories to delete if I want to remove zaptel? |
| 18:32.36 | awannabe | heh, yeah, thats the question |
| 18:32.38 | Aurs | ...other than the Makefile |
| 18:32.43 | awannabe | with this carrier, i think it will be needed, heh |
| 18:33.05 | syzygyBSD | ya, I thought echo was caused by shitty wiring |
| 18:33.16 | syzygyBSD | having a canceler isn't really solving the problem |
| 18:33.26 | CunningPike | awannabe: I would try software e/c first and, if that doesn't work for you, you can add a hardware one after |
| 18:33.34 | *** join/#asterisk bkw__ (n=bkw_@asterisk/friend-and-developer/bkw) |
| 18:33.46 | awannabe | yeah, thast true |
| 18:35.33 | awannabe | so i guess sangoma it is then |
| 18:35.59 | Qwell | awannabe: Some Digium hardware has echo cans... |
| 18:36.08 | *** join/#asterisk tdi (n=tdi@80.48.205.2) |
| 18:36.10 | Strom_C | TE207/212P for example |
| 18:36.16 | awannabe | yeah, just hella pricey |
| 18:36.27 | *** join/#asterisk ZX81 (n=ZX81@83.225.58.27) |
| 18:36.30 | ZX81 | hi |
| 18:36.32 | Qwell | Isn't Sangoma more expensive? I honestly don't know |
| 18:36.43 | Strom_C | per span, i believe sangoma is more expensiver |
| 18:36.44 | Strom_C | -r |
| 18:36.51 | awannabe | hrmm |
| 18:37.00 | ZX81 | anyone seen "wrong elf class" |
| 18:37.05 | ZX81 | with g729 |
| 18:37.08 | ZX81 | from digium |
| 18:37.10 | tdi | hi, ive got problem with meetme on asterisk 1.2.9/1.2.11, I call into the conference room. meetme executes, i leave, but the channels stay. BYE is being sent. Using SIP |
| 18:37.12 | awannabe | im looking at the wanpipe docs right now, im ultra confused on how it does all this |
| 18:37.35 | syzygyBSD | ZX81: no, but I am guessing you need to update your elf drivers and recompile g729 again |
| 18:37.58 | mercestes | awannabe: For the HW echo cancellers? |
| 18:38.05 | awannabe | no, for the wanpipe |
| 18:38.18 | CunningPike | awannabe: Digium is easier to set up....... |
| 18:38.20 | awannabe | i guess it gets a interface and you route to it basically |
| 18:38.23 | syzygyBSD | tdi: there is an option when transfering into a meetme to close the conference on exit |
| 18:38.29 | Qwell | ZX81: Which binary are you getting, and what arch is your system? |
| 18:38.32 | mercestes | awannabe: Oh...mine came with a CD but..I dl'd them and followed the how to..it seemed pretty flawless with wanpipe-cfg |
| 18:38.55 | awannabe | mercestes: so your ethernet interface is the gateway then? |
| 18:38.55 | tdi | syzygyBSD: you mean the option to _leave_ conference with a # DTMF ? |
| 18:39.04 | syzygyBSD | no |
| 18:39.07 | awannabe | or do you have to route to the sangoma card? |
| 18:39.14 | tdi | syzygyBSD: or the last exit closes the conference? |
| 18:39.24 | awannabe | cause im doing another firewall, so im trying to see how this would work, heh |
| 18:39.27 | syzygyBSD | that one |
| 18:39.29 | tdi | i have tried everything from meetme :) |
| 18:39.54 | mercestes | awannabe: IT's just a zap interface. You run wanrouter than zaptel and you just zap to it... |
| 18:40.12 | syzygyBSD | so meetme(blah|dx) |
| 18:40.19 | Qwell | ZX81: ? |
| 18:40.23 | *** join/#asterisk TPC (n=andreas@pdpc/supporter/student/TPC) |
| 18:40.29 | tdi | syzygyBSD: i am not using dynamin conferences |
| 18:40.44 | syzygyBSD | um... why do you want to close it then? |
| 18:41.15 | Qwell | ~lart ZX81 |
| 18:41.19 | tdi | what is the point of the channel to stay ? |
| 18:41.22 | syzygyBSD | and how do you add a conference if not dynamically? |
| 18:41.24 | TPC | hello, I have bought a so-called voip usb phone. it shows up as a sound card and a usb HID device when I plug it in. Can I use it directly with asterisk or do I have to go trought some kind of softphone? |
| 18:41.27 | *** join/#asterisk Ebola (i=1000@81-86-155-65.dsl.pipex.com) |
| 18:41.36 | tdi | i just do not have ,d option |
| 18:42.28 | *** join/#asterisk eliXier (i=GTI16V@gti.twice-irc.de) |
| 18:42.40 | TPC | (I run debian linux, and if possible I would prefer not to have it as a gui program since I prefer to use terminal apps) |
| 18:42.44 | tdi | syzygyBSD: dx also does not close the channel: SIP/tdi-081b3ee8 1234@default:1 Up MeetMe(1234|dx) |
| 18:42.46 | *** join/#asterisk jamincollins (n=jcollins@ptech7-44.acdmis.com) |
| 18:42.53 | tdi | show channels still shows that |
| 18:43.11 | syzygyBSD | sorry, i missread what you were saying before |
| 18:43.24 | *** join/#asterisk DrukenHME (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 18:43.29 | tessier__ | Is it possible to implement an IVR using the asterisk manager api? |
| 18:43.41 | tessier__ | hmm...maybe this should be an AGI actually. |
| 18:43.47 | syzygyBSD | tessier__: yes, i have done several of them |
| 18:44.01 | syzygyBSD | I have done them both |
| 18:44.22 | jamincollins | does anyone know of a way to allow queue agents to use an external number? It looks like AgentCallbackLogin is expecting the extension/number provided to be local |
| 18:44.42 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 18:44.47 | syzygyBSD | jamincollins: what do you mean use an external number? |
| 18:44.53 | syzygyBSD | do you mean dialout? |
| 18:44.56 | jamincollins | yes |
| 18:45.00 | tessier__ | I am trying to find the best and easiest way for my agents to be able to log onto all queues or some queues, and log off all queues. Without having to enter their extension or pin code/ |
| 18:45.17 | tessier__ | The only thing they should have to enter is their agent id |
| 18:45.25 | syzygyBSD | well, the agent's phone is just a sip extension, it should use whatever is in the context you put it in |
| 18:45.36 | jamincollins | say an agent at a remote location calls in, logs in with their agent id and pass, then provides their cell (or hotel number) |
| 18:46.00 | syzygyBSD | ok.... |
| 18:46.06 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
| 18:46.24 | jamincollins | I've tried supplying a context to the AgentCallbackLogin but it seems to ignore it and instead trigger [default] |
| 18:46.27 | tessier__ | In my case agents will not and should not be logging in from anywhere but on our phones. |
| 18:46.59 | *** join/#asterisk _deg_ (n=deg@200.163.193.247) |
| 18:47.27 | jamincollins | tessier if their CALLERIDNUM is the same as their agentid and pin, then you could use that variable |
| 18:47.29 | syzygyBSD | jamincollins: I think you need a couple || before |
| 18:47.39 | *** join/#asterisk hrmmm (i=blake@tata.cluebie.net) |
| 18:47.53 | jamincollins | tried without, and with one, I'll try a second one too |
| 18:48.21 | _deg_ | Im facing trouble using variables on Goto cmd |
| 18:48.29 | _deg_ | Anyone knows if it is possible? |
| 18:48.46 | _deg_ | Example: exten => Goto(${ARG1}) |
| 18:48.47 | syzygyBSD | jamincollins: it depends on the version of asterisk, 1.2 needs 2 of them |
| 18:48.59 | _deg_ | where ARG1 = extensions,200,1 |
| 18:49.30 | jamincollins | syzygyBSD, that did it |
| 18:49.34 | jamincollins | thanks man |
| 18:49.40 | syzygyBSD | np :) |
| 18:50.05 | syzygyBSD | _deg_: I think variables show up as 1 parameter/option |
| 18:50.27 | syzygyBSD | that might work if they were an array though, but I couldn't tell you for sure |
| 18:51.04 | _deg_ | syzygyBSD, Something like this: Goto(${ARG1},${ARG2},${ARG3}) |
| 18:51.17 | *** part/#asterisk tdi (n=tdi@80.48.205.2) |
| 18:51.26 | _deg_ | syzygyBSD, It didnt work as well |
| 18:51.41 | nortex | I'm trying to install it and want to test the imap support for voicemail and can not figure out how to get imap_tk on my CentOS 4 system. Any ideas? |
| 18:55.08 | syzygyBSD | _deg_: that works for me... |
| 18:55.40 | _deg_ | syzygyBSD, hmmm |
| 18:56.04 | [TK]D-Fender | _deg_: Pastebin a specific example. |
| 18:56.06 | [TK]D-Fender | ~pb |
| 18:56.09 | jbot | pb is, like, a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/, or http://www.paste-it.net/ |
| 18:56.18 | c4t3l | is there a size limit to the Pcom mac-directory.xml file? |
| 18:56.22 | c4t3l | is there a size limit to the Pcom mac-directory.xml file? |
| 18:56.30 | syzygyBSD | _deg_: as a debugging option the line before do a NOOP of the same information, make sure it is right |
| 18:56.32 | c4t3l | sorry |
| 18:57.17 | *** join/#asterisk xxarmiexx (n=armie@arm.enter.net) |
| 18:58.42 | syzygyBSD | nortex: I haven't seen imap with voicemail yet, do you have some docs I could read about this? |
| 18:59.44 | awannabe | c4t3l: i havent seen a offical one |
| 18:59.50 | awannabe | you got that many numbers? |
| 19:00.07 | c4t3l | about 95. the file is 12K |
| 19:00.23 | awannabe | are you having problems, or just wondering? |
| 19:00.38 | *** join/#asterisk DrukenHME (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 19:00.48 | c4t3l | probs. my ftp logs say it downloaded, but its not in the phone |
| 19:01.12 | awannabe | parse error somewhere? |
| 19:01.42 | c4t3l | the phone gives no indication of any errors |
| 19:01.53 | c4t3l | i'll look at the xml again |
| 19:01.55 | deb_user | i'm running x-lite behind a firewall...as soon as I open the client and register everything is fine, after about 2 or 3 minutes the connection seems to time out, and all calls to that client go directly to voicemail, saying that the line is busy |
| 19:02.25 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 19:02.29 | deb_user | i don't see any sip pokes in the asterisk cli |
| 19:02.35 | syzygyBSD | deb_user: deb_user set nat=1 |
| 19:02.35 | deb_user | for that particular client |
| 19:02.37 | *** part/#asterisk TPC (n=andreas@pdpc/supporter/student/TPC) |
| 19:02.46 | jamincollins | is there a way to announce to the agent receiving an agentcallbacklogin call that they need to press pound? or did I miss something? |
| 19:02.55 | *** join/#asterisk eBody (n=icechat5@207.71.51.162) |
| 19:03.02 | _deg_ | syzygyBSD, I will pastebin an error. |
| 19:03.05 | deb_user | syzygy: my configuration as it is has nat=yes |
| 19:03.10 | deb_user | what is the difference? |
| 19:03.17 | c4t3l | deb_user do you have your qualify statement set? |
| 19:03.25 | c4t3l | in sip.conf |
| 19:03.28 | syzygyBSD | um... 4 letters? |
| 19:03.32 | nortex | syzygyBSD, I'm looking at what info is in the trunk on SVN, I just want to try it out. |
| 19:03.38 | syzygyBSD | I prefer 1 over yes, it works for me |
| 19:03.40 | eBody | how can i get the wcfxs module?? |
| 19:03.41 | eBody | i already have the fxs module installed on my tdm400 |
| 19:04.00 | deb_user | c4t3l: yes, qualify is set to yes |
| 19:04.01 | awannabe | what is a wcfxs? |
| 19:04.20 | mercestes | Can anyone please tell me how to fix: WARNING[26721]: interface.c:215 decodeMP3: Junk at the beginning of frame 49443302 |
| 19:04.25 | Strom_C | eBody: you want wctdm |
| 19:04.28 | Qwell | awannabe: a deprecated module name |
| 19:04.30 | c4t3l | deb_user do you have canreinvite set |
| 19:04.37 | syzygyBSD | mercestes: fix your mp3 file |
| 19:04.40 | eBody | i have wctdm and wcfxo already modprobed |
| 19:04.44 | eBody | and running |
| 19:04.48 | deb_user | c4t3l: canreinvite=no |
| 19:04.51 | mercestes | They're the default mp3 files...>.< |
| 19:04.52 | Strom_C | eBody: why wcfxo? |
| 19:05.04 | eBody | i just got a fxs module so we can use an analog phone |
| 19:05.06 | syzygyBSD | sox mymp3.mp3 fixedmp3.mp3 |
| 19:05.11 | eBody | but i still don't get a dialtone |
| 19:05.25 | Strom_C | eBody: if you only have a tdm400, you only need wctdm |
| 19:05.28 | syzygyBSD | mercestes: default mp3 files? |
| 19:05.36 | mercestes | syzygyBSD: Yea, the MOH files... |
| 19:05.41 | syzygyBSD | i didn't know mp3s came with asterisk... |
| 19:05.48 | mercestes | Yea, the default music on hold. |
| 19:05.50 | Strom_C | eBody: and if that's the case, then your config is probably isn't set up correctly |
| 19:06.30 | eBody | Strom, then why do i need wcfxo for my fxo modules?? |
| 19:06.42 | Strom_C | eBody: uh, you don't |
| 19:06.54 | Strom_C | eBody: if you have an x100p card, you need wcfxo |
| 19:07.15 | Strom_C | eBody: but if it's just a tdm400p, you only need wctdm |
| 19:07.16 | eBody | oh ok ok. so if i were to "rmmod wcfxo" my sip extensions would still work?? |
| 19:07.30 | Strom_C | yes; sip has nothing to do with zaptel |
| 19:07.30 | deb_user | but why do incoming calls work fine within the first 2 or 3 minutes or so, but afterwards the connection seems to timeout? |
| 19:08.13 | deb_user | sip_poke_noanswer: Peer 'test' is now UNREACHABLE! Last qualify: 19 |
| 19:08.42 | deb_user | that was a couple of minutes after initiating x-lite, getting a register, and being able to dial in |
| 19:09.01 | deb_user | i can still dial out |
| 19:09.05 | syzygyBSD | deb_user: because your nat refreshes after that |
| 19:09.06 | deb_user | just can't receive incoming calls |
| 19:09.22 | deb_user | syzygy: what should I do? |
| 19:09.43 | deb_user | already set nat=1 btw |
| 19:09.46 | syzygyBSD | in the phone config somewhere there should be an option of nat keepalive |
| 19:09.58 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 19:10.14 | syzygyBSD | deb_user: that just makes sure that it maps to the correct port/ip, it doesn't try to keep alive |
| 19:10.17 | deb_user | there is a "send sip keep alives" |
| 19:10.23 | syzygyBSD | that would be it |
| 19:10.32 | deb_user | its clicked on |
| 19:10.44 | deb_user | i mean, its enabled |
| 19:10.52 | syzygyBSD | then your router just hates you |
| 19:11.12 | syzygyBSD | you can always just forward port 5060 UDP to your computer |
| 19:11.19 | *** join/#asterisk Avalone (n=Avalone_@dial-016.vl-cen-as3.avtlg.ru) |
| 19:11.32 | deb_user | ha... |
| 19:11.34 | deb_user | yeah |
| 19:11.48 | deb_user | my router does hate me |
| 19:12.09 | deb_user | you think those keep alives aren't making it out of the router? |
| 19:12.17 | deb_user | dropped packets? |
| 19:14.02 | syzygyBSD | no, I think it is something else, but I dont' really care about learning how to configure xlite |
| 19:14.26 | syzygyBSD | it is easy enough just to forward the port if you can |
| 19:14.38 | jamincollins | is there a way to announce to the agent receiving an agentcallbacklogin call that they need to press pound when ackcall=yes? |
| 19:15.02 | *** join/#asterisk murf (n=steve_mu@216.166.159.235) |
| 19:15.18 | hrmmm | On an Asterisk/FreePBX setup I am attempting to rewrite/override CallerID(name/num/all) on inbound calls coming in via a Zap trunk(pri), I call Set(CALLERID(name)=WANG), right afterward a noop(${CALLERID}) reflects what I've done, freepbx uses a dial macro which calles dialparties.agi, seems that the AGI is passed the original callerid information. Now if I originated the call from a sip extension my rewriting works. Any ideas? |
| 19:15.25 | eliXier | x-litw sucks... try to www.phoner.de ;) |
| 19:15.29 | eliXier | e |
| 19:16.41 | _deg_ | syzygyBSD, http://pastebin.com/780851 |
| 19:17.03 | syzygyBSD | ~freepbx |
| 19:17.13 | jbot | freepbx is, like, the Microsoft BOB of PBXes and NOT supported here! People using it should join #freepbx (FreePBX is the new name of AMP) |
| 19:17.29 | *** join/#asterisk anthonyl (n=anthonyl@adsl-68-252-207-35.dsl.chcgil.ameritech.net) |
| 19:18.36 | *** join/#asterisk GerbilWrk (i=GerbilNu@65.88.144.41) |
| 19:18.50 | syzygyBSD | _deg_: all your arguments are one variable still, split them up |
| 19:18.59 | *** join/#asterisk smackus (n=ckwall@c-67-169-248-217.hsd1.ut.comcast.net) |
| 19:18.59 | _deg_ | ok, i will |
| 19:19.08 | L-info | quick q.. can you set asterisk to deny registration requests for a peer whom already is registered with different details (i.e. limit the number of users that can register with the same iax user/pass)? |
| 19:19.12 | smackus | can anyone shed some light on why i got this message? |
| 19:19.13 | smackus | Aug 31 12:04:12 NOTICE[7715]: channel.c:1917 ast_read: Dropping incompatible voice frame on Local/18014407188@progrexion-30b4,2 of format ulaw since our native format has changed to slin |
| 19:19.58 | syzygyBSD | smackus: in your sip.conf do you have disallow=all? |
| 19:20.06 | smackus | yes |
| 19:20.30 | hrmmm | jbot: yes, but this seems like some oddity in what the asterisk agi is passing a script |
| 19:20.31 | jbot | hrmmm: what are you talking about? |
| 19:20.45 | hrmmm | ererr syzygybsd :) |
| 19:20.53 | syzygyBSD | lol |
| 19:20.59 | Filar | awsome |
| 19:21.02 | Filar | he replied to a bot |
| 19:21.14 | GerbilWrk | I've got a SIP connection to one of our providers, and they are currently down, but I'm not getting a "channel busy/unavailable" response for nearly a minute after the call is placed |
| 19:21.27 | GerbilWrk | anyone know how to take care of that so it'll failover sooner? |
| 19:21.37 | hrmmm | filar: yes a complete novice, I bow to your mastery |
| 19:22.01 | Filar | please it happens to every |
| 19:22.08 | Filar | someone just has to point it out thats all ;) |
| 19:22.34 | syzygyBSD | hrmmm: I don't know what freepbx is using but when I do it with agi's it works for me |
| 19:23.51 | *** join/#asterisk jjhall (n=chatzill@72.24.119.202) |
| 19:25.04 | jjhall | Does anyone have some experience unlocking PAP2s? If so, I'd like to get some experience. |
| 19:25.29 | jjhall | or help rather. :-) |
| 19:25.51 | *** join/#asterisk oej (n=oej@062249177046.customer.alfanett.no) |
| 19:26.05 | hrmmm | syzygyBSD: your call originates via some Zap interface? |
| 19:26.50 | n0n0x | Is there any way to avoid multiple agents login at the same extension? |
| 19:27.21 | syzygyBSD | if I want it to.. like have my cell phone call in, change the Callerid and dial out |
| 19:31.38 | hrmmm | not at all familiar with how this type of thing works.. when asterisk calls an agi is the data available through agi api for that call instantiated at the point of the Agi() call, or is it always maintained for everycall? That is is it possible that my Set(CALLERID...) is not modifying the data kept for Agi? |
| 19:31.48 | *** join/#asterisk grabeez (n=gaving@grabes2.enter.net) |
| 19:32.09 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 19:32.24 | *** join/#asterisk budairc (n=chatzill@proxy01.mhnet.com.br) |
| 19:34.04 | grabeez | I just switched queue members to static members... is it normal for asterisk to keep trying to call the members IE: chan_sip.c: Call to user '115' rejected due to usage limit of 1 |
| 19:34.34 | grabeez | It doesn't seem to be saving the call states when I do a show queues either |
| 19:35.28 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 19:35.28 | *** mode/#asterisk [+o russellb] by ChanServ |
| 19:41.20 | GerbilWrk | I've got a SIP connection to one of our providers, and they are currently down, but I'm not getting a "channel busy/unavailable" response for nearly a minute after the call is placed. Anyone know how to make it failover quicker? |
| 19:42.00 | eKo1 | pass a timeout to dial? |
| 19:42.24 | GerbilWrk | will that affect it going through and the other party just not answering? |
| 19:43.26 | GerbilWrk | I need to make sure it only happens when the channel is down |
| 19:44.23 | eKo1 | then you have no choice but to wait for a busy/unavailable |
| 19:46.25 | GerbilWrk | well that's horrible |
| 19:47.18 | *** join/#asterisk xyklopz (n=xyklopzi@86.122.8.28) |
| 19:47.37 | eKo1 | tough |
| 19:48.04 | smackus | eKo1: could he use the chanisavail command? would that do what he needs? |
| 19:48.12 | syzygyBSD | hrmmm: with every AGI call the variables that are set for that channel are passed through stdin |
| 19:49.13 | eKo1 | smackus: with a provider? I don't think that will work. |
| 19:49.42 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 19:49.49 | syzygyBSD | it depends on how the provider is down |
| 19:50.32 | syzygyBSD | if they just arn't letting your register it will come back almost immediatly, if there is a network timeout (there server is offline) then it will take much longer |
| 19:51.20 | crochat | When I call from another IP phone in my LAN, the CallerID is displayed on my analog phone which is connected on the AG168V FXS ! But when I receive an incoming call through my SIP provider, Asterisk tell the CallerID on all the IP phones, but on the phone behind the AG168V, it's just displayed "Incoming call" !! WHY ?? |
| 19:52.24 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 19:52.38 | eKo1 | crochat: what? |
| 19:53.01 | *** part/#asterisk Avalone (n=Avalone_@dial-016.vl-cen-as3.avtlg.ru) |
| 19:53.43 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 19:53.44 | *** mode/#asterisk [+o mog] by ChanServ |
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| 19:56.26 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 19:57.04 | *** join/#asterisk WEH (n=bob@pool-71-250-60-139.nwrknj.east.verizon.net) |
| 19:57.29 | *** part/#asterisk malaxmo (n=malexmi@N93e0.n.pppool.de) |
| 19:57.40 | *** part/#asterisk smackus (n=ckwall@c-67-169-248-217.hsd1.ut.comcast.net) |
| 19:58.03 | WEH | can chan_sip.c handle a 202 Accepted response? |
| 19:58.22 | WEH | the error i get is: Aug 31 15:51:34 NOTICE[3790]: chan_sip.c:10307 handle_response: Dont know how to handle a 202 Accepted response from 66.227.100.21 |
| 19:58.26 | eKo1 | grep it and find out |
| 19:58.29 | niter3 | http://paste.lisp.org/display/25059 |
| 19:58.32 | niter3 | Hey guys |
| 19:58.45 | eKo1 | WEH: I think that notice says it all |
| 19:58.50 | niter3 | I'm setting up an extension and my asterisk box is behind a nat. Am i missing any settings? |
| 19:59.26 | eKo1 | niter3: are you having problems? |
| 19:59.38 | oej | WEH: 202 Accepted on what transaction? |
| 19:59.49 | niter3 | eKo1: i was earlier but added a few things and having hadn't time to check. |
| 20:00.22 | niter3 | Just wanted somebody to review it real quick |
| 20:00.22 | eKo1 | niter3: well, when you do have a problem, come back to us |
| 20:00.22 | niter3 | all my RTP ports are forward |
| 20:00.22 | niter3 | and my 5060 ports too |
| 20:00.22 | niter3 | I did have a problem. |
| 20:00.29 | niter3 | I'm just verifing i'm not missing anything |
| 20:00.48 | eKo1 | It is easier to verify by testing. |
| 20:01.06 | niter3 | since my problem. And my problem was i couldn't communicate with another sip person behind nat and the other on the internet. However both could dial in and listne to messages and leave messages. |
| 20:01.26 | GerbilWrk | syzygyBSD, there server appears to be down completely, so it's taking a while |
| 20:02.00 | teknoprep | how do i watch what calls are using what codecs outbound? |
| 20:02.13 | niter3 | i figure somebody could review my extenion over real quick and let me know if i'm missing anything before trying. |
| 20:02.24 | *** join/#asterisk topping (n=topping@207.47.6.201.static.nextweb.net) |
| 20:02.25 | oej | WEH: What transaction did you get 202 on? |
| 20:03.00 | [hC] | anyone here using smokeping to monitor asterisk boxes latency? |
| 20:03.12 | [hC] | Im seeing something strange and am curious if other people see the same thing |
| 20:04.02 | [hC] | during peak calling periods, when monitoring from my main IAX trunk server, (all clients call out thru it() my latency graphs spike for everyone |
| 20:04.18 | [hC] | meaning that when calls are taking place, the latency for the main server itself is bouncing around |
| 20:06.58 | *** join/#asterisk DrukenHME (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 20:07.59 | *** join/#asterisk pyrom (n=pyro@86.84-48-44.nextgentel.com) |
| 20:09.27 | pyrom | When i call from my voIP phone via my spa3102 (lineout) it's very noisy and i can't hear a thing, however if i in the CLI use the dial command, then transfer the call to the voIP phone: it's crystal clear. |
| 20:10.22 | hmmhesays | bah what a crap day |
| 20:10.31 | hmmhesays | I got asterisk half running on my mips box |
| 20:10.47 | niter3 | hrm... Is it 5pm yet?! |
| 20:10.48 | niter3 | gez |
| 20:11.57 | nettie | Hi guys, when I'm dialling one of my phones which is busy I get this error message.. anyone know what does it means? where should I define the t extension? |
| 20:11.59 | nettie | Timeout, but no rule 't' in context blah |
| 20:12.38 | sx-wks | what's wrong here ? |
| 20:12.41 | sx-wks | exten => i,1,Set(STAR_COUNT=$[1+${STAR_COUNT}]) |
| 20:12.41 | sx-wks | exten => i,n,GotoIf($[${STAR_COUNT}>5]?no_star,1:s,prompt_star) |
| 20:13.00 | eKo1 | nettie: define it in your dialplan; e.g. exten => t,1,hangup |
| 20:13.21 | sx-wks | it doesn't seem to like what's inside the "set" |
| 20:13.41 | nettie | eKo1 in the context? or at the top? I did that but if I rember correctyl the compained.. lemme rety |
| 20:13.42 | nettie | thanx |
| 20:13.43 | nettie | btw |
| 20:13.51 | eKo1 | nettie: yes, in the context |
| 20:14.32 | eKo1 | sx-wks: maybe because ${STAR_COUNT} doesn't exist? |
| 20:15.07 | nettie | eKo1 it shows: == Everyone is busy/congested at this time (1:1/0/0) which is corrent |
| 20:15.09 | nettie | correct |
| 20:15.28 | nettie | eKo1: but later says: WARNING[9145]: pbx.c:1700 pbx_extension_helper: No application '' for extension (enerjetica-local, t, 1) |
| 20:15.30 | sx-wks | I have : exten => s,n,Set(STAR_COUNT = 0) a few lines earlier |
| 20:15.37 | nettie | seems lik eit doesnt find it |
| 20:15.37 | nettie | uhmm |
| 20:15.42 | nettie | maybe a typo |
| 20:15.47 | nettie | lemme doublecheck |
| 20:17.00 | nettie | npe seems correct uhmm |
| 20:17.43 | eKo1 | sx-wks: what does noop($[1+${STAR_COUNT}]) show? |
| 20:17.44 | sx-wks | nettie: hah. it didn't like the spaces around the '=' in the "exten => s,n,Set(STAR_COUNT = 0)" |
| 20:17.52 | pyrom | Anyone got any idea? why my voip calls(call from voip phone via asterisk and out the pstn-spa3k trunk) get's very lousy quality (e.g stumbling, can't allmost hear anything) but when i call from asterisk(out the pstn-spa3k lineout) then transfer it to the voip phone it's crystal clear. |
| 20:17.55 | sx-wks | (wtf ??) |
| 20:17.57 | eKo1 | sx-wks: figures. |
| 20:18.58 | Strom_C | pyrom: what codecs are you using? |
| 20:19.19 | E-bola | Are there any advantages to using ulaw over alaw? |
| 20:19.24 | E-bola | or vice versa |
| 20:19.31 | eKo1 | no |
| 20:19.54 | pyrom | Strom_C, alaw i think |
| 20:20.06 | sx-wks | E-bola: depends on your country |
| 20:20.07 | Strom_C | pyrom: double check |
| 20:20.14 | nettie | sx-wks yeah! I removed the spaces |
| 20:20.16 | pyrom | disallow=all ; First disallow all codecs |
| 20:20.16 | pyrom | allow=ulaw |
| 20:20.23 | nettie | sx-wks and worked :( |
| 20:20.26 | *** join/#asterisk cekc (n=cekc@rrcs-24-199-36-210.west.biz.rr.com) |
| 20:20.27 | pyrom | that's the only *allow* lines in my sip.conf |
| 20:20.35 | sx-wks | nettie: wtf ?? |
| 20:20.36 | cekc | I want to have Asterisk's babies |
| 20:20.39 | E-bola | sx-wks: why? |
| 20:20.43 | E-bola | i meant technical advantages |
| 20:20.46 | E-bola | not compatability |
| 20:20.53 | *** join/#asterisk rene- (n=rene1@gea-gye-internet.telconet.net) |
| 20:21.03 | sx-wks | E-bola: well, you'll save a pinch in not having to convert to sent to your provider :D |
| 20:21.16 | cekc | I have my GXP2000 working with BLF |
| 20:21.16 | nettie | sx-wks : == Spawn extension (enerjetica-local, t, 1) exited non-zero on 'SIP/e61-0987d900' |
| 20:21.29 | sx-wks | E-bola: there's nor real advantage. it's like PAL vs NTSC |
| 20:21.37 | nettie | sx-wks now works asterisk finds it |
| 20:21.46 | E-bola | sx-wks: dont get me started on why pal is superior to ntsc :) |
| 20:21.54 | sx-wks | E-bola: see :D |
| 20:22.02 | E-bola | but ok i get ur point |
| 20:22.12 | Cresl1n | hey, let's all go to #asterisk-newbies |
| 20:22.12 | E-bola | it seems my provider (in denmark, europe) only supports ulaw though |
| 20:22.15 | E-bola | thats kinda odd |
| 20:22.27 | sx-wks | should be alaw |
| 20:22.27 | pyrom | Strom_C, G.711 u-law , but since it works when i transfer the call, i find this quite odd. |
| 20:23.16 | pyrom | changed. |
| 20:23.48 | Strom_C | pyrom: if you're in the US, use ulaw |
| 20:24.06 | Strom_C | pyrom: and it's ulaw all the way through? are all devices on the same LAN> |
| 20:24.07 | Strom_C | ? |
| 20:24.28 | nettie | sx-wks the problem is that I hear the busy tome only when the default tiemout is spawned and not when asterisk says 486 "Busy Here" |
| 20:24.35 | nettie | any idea? |
| 20:24.47 | pyrom | Strom_C, All the devices within same LAN, i believe my sipura spa3k has the same codec by default. |
| 20:24.56 | Strom_C | pyrom: check please |
| 20:25.22 | pyrom | Strom_C, when i switched to a-law i get chan_sip.c:3691 process_sdp: No compatible codecs! after (attempting native bridge) |
| 20:25.23 | Strom_C | nettie: because you need to immediately do a conditional branch based on the DIALSTATUS that Dial() returns |
| 20:25.24 | pyrom | i will. |
| 20:25.27 | rene- | my E1 board starts restarting at random times, keeps restarting for a while and then it recovers, while is in mad restart phase i can not dial out.. i am not sharing IRQs |
| 20:25.48 | rene- | it is taking interupts even in the mad restart phase |
| 20:26.06 | beu | E-bola: o/ |
| 20:26.27 | rene- | i am using DMA mode 5 and i have unmaskirq set to on |
| 20:26.39 | rene- | help? |
| 20:26.57 | E-bola | beu? |
| 20:27.36 | jhb | hi *. How do I find out which sip users are currently registered with my server? |
| 20:27.40 | beu | ah, different E-bola :o |
| 20:27.51 | hmmhesays | sip show registry |
| 20:27.52 | rene- | this all happens while asterisk is mostly idle, it only answers to 2 or 3 ami requests per second, and i have mysql running it this box but its foot print isnt even noticeable in top |
| 20:28.42 | jhb | hmmhesays: mmmh, I meant clients that use my * to register themselfes |
| 20:28.46 | WEH | is there a way to set caller ID number using Asterisk MAnager Api? |
| 20:28.51 | tessier__ | Anyone know why my AGI would end the call with "Declined: 804" when I do a hangup() command in the agi? 804 is the extension I called to execute the AGI. It should be hanging up properly. |
| 20:29.24 | tessier__ | Do I have to exit the agi script itself in some special way? |
| 20:29.49 | pyrom | Now sip.conf has disallow=all, and allow=alaw : the sipura has used pref. codec G711a under both (line1 & pstn line tab) and used pref. codec only. |
| 20:31.14 | *** part/#asterisk cekc (n=cekc@rrcs-24-199-36-210.west.biz.rr.com) |
| 20:31.46 | pyrom | Same issue, still so much stumbling that i can't seperate an "Hello World" with an "Hi world" |
| 20:35.48 | pyrom | The voip phone is an Nokia E70, which IS reported to be working with * |
| 20:36.09 | *** join/#asterisk dalekurt (n=DaleKurt@65.183.3.229) |
| 20:36.51 | dalekurt | Anyone got IAXTEL (Gnophone) or FWD account I can call |
| 20:37.17 | E-bola | pyrom: do u have any cell phones working with *? |
| 20:37.22 | E-bola | im considering bying a nokia n series |
| 20:40.10 | *** join/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 20:40.35 | pyrom | E-bola, nokia e70 |
| 20:40.43 | pyrom | E-bola, with inbuilt voip client. |
| 20:41.02 | E-bola | and it works flawlessly? |
| 20:41.05 | pyrom | E-bola, The N series currently does not have any "open" voip clients that work with a specified ... |
| 20:41.21 | pyrom | E-bola, well..... i am experiencing troubles atm, and i just got the phone |
| 20:41.33 | tessier__ | Why is my dialplan causing auto-fallthrough when a call ends when I have my agi on step 2 and a Hangup() command on step 2? |
| 20:41.43 | tessier__ | It should not be falling through. I am explicitly hanging up the call. |
| 20:41.44 | pyrom | The thing is when i transfer a call to the phone from asterisk, the sound quality is great, |
| 20:41.57 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 20:42.24 | pyrom | But when i try to make an outgoing call via my spa3102 i get stumbling (e.g the call is made from the phone) but since transferring the call works: I dont exactly think it's a phone issue. |
| 20:43.07 | pyrom | But it registers very quickly, and it rings when it should :-) |
| 20:43.59 | rene- | Strom_C: do you have any experience with thise E1 slip problems? |
| 20:44.08 | rene- | s/thise/these |
| 20:44.39 | TripleFFFF | <PROTECTED> |
| 20:44.46 | TripleFFFF | why is DEL always 40ms |
| 20:44.52 | TripleFFFF | is there a way to make buffer smaller ? |
| 20:45.13 | Strom_C | rene-: what kind of card are you using |
| 20:45.13 | TripleFFFF | coz they have a 3ms RT to me.. i gyess.. a 40ms is lame |
| 20:45.16 | nettie | Strom_C: what dialstatus can I use for a SIP messages 603 "Decline" please? That's what happen when then user actually declines the call .. |
| 20:45.37 | rene- | a 2 port digium. TE205 i think |
| 20:45.40 | Antax | can dsl modems be used as ATA devices? |
| 20:45.41 | Strom_C | nettie: what dialstatus does Dial() return in that case? |
| 20:45.45 | nettie | Strom_C I would like the caller to get just a busy signal if possible |
| 20:46.02 | Strom_C | rene-: your best bet, honestly, is to call digium |
| 20:46.20 | nettie | Dial returns s congestion |
| 20:46.31 | nettie | Strom_C you think it handles it as busy uhmm |
| 20:46.40 | rene- | ok |
| 20:46.44 | nettie | the sip response is different anyway |
| 20:46.49 | *** join/#asterisk lowlevel (n=Stuart@bigbrother.vermeulens.com) |
| 20:47.19 | Antax | or are ATA devices be used as dsl modems? |
| 20:47.43 | nettie | because on the Nokia e61 I get a message saying internet telephone service no available instead of a busy tone |
| 20:48.01 | jjhall | Antax: No, ATAs and DSL modems are completely different beasts. |
| 20:48.15 | nettie | and asterisk read the phone "reject" button as a Decline |
| 20:49.10 | Antax | thnx |
| 20:50.21 | pyrom | "Whee, transferring SIP/pstn-spa3k-c858 to 60@nokia." |
| 20:50.31 | *** join/#asterisk anthonyl (n=anthonyl@66.146.207.99) |
| 20:50.38 | pyrom | How can i check what codecs is in use, when i transfer a call between two "extensions" |
| 20:51.01 | eKo1 | are they sip? |
| 20:51.18 | pyrom | yes |
| 20:51.24 | *** join/#asterisk websae (n=websae@adsl-64-149-206-121.dsl.milwwi.sbcglobal.net) |
| 20:52.30 | pyrom | "Attempting native bridge of SIP/pstn-spa3k-c858 and SIP/60-2850 |
| 20:52.30 | pyrom | " |
| 20:52.45 | eKo1 | sip show channels |
| 20:54.56 | pyrom | Strom_C, Then i'm pretty sure: says alaw alaw even if i call directly from the voip phone, or i transfer the call via asterisk. |
| 20:59.13 | Mugatu | hey, talking SIP to an asterisk host, anyone know why after a successful REGISTER transaction, I get a NOTIFY from the server (simple-message-summary), and almost immediately following, a 481, Subscription does not exist |
| 21:00.55 | Mugatu | uh, scratch that, I've got my wires crossed a little bit :) |
| 21:06.28 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 21:07.26 | *** join/#asterisk FlyboySR22 (n=rsears@gateway.americanis.net) |
| 21:08.55 | *** join/#asterisk Preytell (n=7049c6d6@216.63.37.117) |
| 21:09.33 | Preytell | Hello, before I make anyone mad, can I ask questions about Asterisk and SpanDSP here? |
| 21:09.36 | *** join/#asterisk bmg505 (n=leon@c1-95-5.rndf.isadsl.co.za) |
| 21:09.46 | TripleFFFF | y |
| 21:10.05 | Preytell | Has anyone successfully compiles spandsp in 1.2.11? |
| 21:10.49 | rene- | no |
| 21:10.54 | rene- | it shouldnt be a problem |
| 21:10.59 | Preytell | Mine keeps barfing on app_[rt]xfax.c |
| 21:11.19 | rene- | can you compile on 1.2.9.1? |
| 21:11.36 | Preytell | Haven't tried. |
| 21:11.45 | sx-wks | 1.2.9.1 sounds ancient |
| 21:12.10 | rene- | what is what you get? |
| 21:13.03 | Preytell | Hundreds of declaration for paramenter errors. Such as "r" but no such parameter. |
| 21:13.49 | rene- | when building asterisk or when building spandsp? |
| 21:14.03 | Preytell | spandsp compiles just fine. Asterisk compiles just fine. But when you include the app_rxfax.c and app_txfax.c it fails. |
| 21:14.38 | rene- | weird |
| 21:15.13 | rene- | you are using spandsp and [rt]fax versions for 1.2.x right? |
| 21:15.23 | Preytell | Everyone seems to want to use TrixBox because spandsp is already there. But trixbox has it's own issues. And I much prefer Asterisk direct. |
| 21:15.28 | Preytell | yes |
| 21:15.39 | Preytell | I just downloaded them again to make sure. |
| 21:16.27 | Preytell | My overall goal is to get AsterFax working. |
| 21:16.44 | Preytell | The only thing holding me up is spandsp. |
| 21:17.43 | rene- | weird.. paste your output in pastebin.ca to see if there are any clue |
| 21:20.59 | bkw_ | rx/tx fax isn't ment to be used for a full on fax machine |
| 21:21.01 | bkw_ | its there for testing |
| 21:21.15 | tessier__ | Testing what? |
| 21:21.18 | Preytell | I was just trying again. What's pasebin.ca? |
| 21:21.26 | Preytell | paste |
| 21:21.42 | bkw_ | tessier__, spandsp :P |
| 21:21.47 | bkw_ | even coppice said that |
| 21:22.01 | bkw_ | but people are trying to build full fax platforms using it and its not really ment for that |
| 21:22.08 | Preytell | asterfax is basing it's commercial application on it. |
| 21:22.10 | tessier__ | And spandsp is? |
| 21:22.24 | *** join/#asterisk pollohawk (n=pollohaw@mmail.picksend.com) |
| 21:22.26 | bkw_ | the software the emulates the software faxing and does t.38 |
| 21:23.09 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 21:23.24 | mtgh | Does anyone know if the cisco config tool that comes with AAH can be downloaded with out getting AAH |
| 21:23.38 | sx-wks | bkw_: I take it that's not yet ready for primetime ? |
| 21:25.16 | tessier__ | Can someone explain to me how asterisk dialplan fall-through changed between 1.0 and 1.2? I just set up my first 1.2 system and AGI isn't working like I expect. I call my AGI, then do a Hangup() but the console still says it fell through and the phone gets a declined message. |
| 21:25.29 | tessier__ | sx-wks: FAX and VOIP don't mix in general it seems. |
| 21:25.36 | tessier__ | sx-wks: Scan the document and email. |
| 21:25.36 | sx-wks | tessier: ah that :D |
| 21:25.40 | *** part/#asterisk hads (n=hads@mail.nice.net.nz) |
| 21:25.47 | rene- | bkw_ should unmaskirq be on or off to avoid PRI slip hell |
| 21:25.55 | [Outcast] | has anyone seen this while trying to configure an E1, sorry about sounding like a Noob. http://pastebin.com/780934 |
| 21:26.01 | bkw_ | rene-, use sangoma cards |
| 21:26.02 | sx-wks | tessier: problem is, we have some moron state agencies here that haven't discovered email yet :D |
| 21:26.13 | [Outcast] | bkw, -agreed |
| 21:26.17 | bkw_ | rene-, they do all that in hardware so you have no slips |
| 21:27.00 | *** join/#asterisk f0urtyfive (i=f0urtyfi@c-67-165-5-232.hsd1.ct.comcast.net) |
| 21:27.05 | rene- | bkw_: i agree i will try to buy sangoma hardware |
| 21:27.17 | bkw_ | but yes check the dma settings on the HD also |
| 21:27.27 | bkw_ | and make sure you're not sharing IRQ's |
| 21:27.32 | rene- | it was udma5 but docs say to shoot for 3 |
| 21:27.37 | rene- | i am not sharing irqs |
| 21:27.42 | bkw_ | then that isn't the problem |
| 21:27.49 | rene- | and i am not using apic |
| 21:27.51 | bkw_ | also sangoma doesn't have this IRQ issue |
| 21:27.54 | bkw_ | try using apic |
| 21:27.56 | [Outcast] | never mind i found the problem |
| 21:28.09 | pyrom | Seems like when i call someone, they can hear me very well, but i can't hear them. (e.g just garbage noise, stumbling and cut off words) |
| 21:28.12 | tessier__ | Kinda wish all asterisk extensions didn't have to be written in C. |
| 21:28.12 | Preytell | I haven't tried 1.2 with AGI yet but it sounds like it is setting the continue fields to something that does not exist in the dialplan. |
| 21:28.20 | bkw_ | tessier__ they don't |
| 21:28.21 | f0urtyfive | anyone have any idea why zaptel-1.2.8 isnt installing ztdummy? |
| 21:28.23 | bkw_ | check out res_perl |
| 21:28.24 | bkw_ | and res_js |
| 21:28.25 | sx-wks | rene-: you SHOULD be using Apic |
| 21:28.33 | tessier__ | It would be easier if this queueing stuff weren't C so I didn't have to kludge agi's to make it work like I want. |
| 21:28.52 | rene- | sx-wks: it makes no difference regarding the slip problem |
| 21:29.01 | sx-wks | rene-: hmmm ? |
| 21:29.03 | rene- | what i could try is using apic and using udma |
| 21:29.08 | tessier__ | bkw_: For some reason they all are. There is so little incentive to write them in any other language it is as good as if they were required to be C. |
| 21:29.57 | bkw_ | tessier__ but you can learn C eh? |
| 21:30.10 | rene- | it is the last combination for me to try, before rebuilding the kernel as per asteriskguru docs. i wonder if unmaskirq should be 1 or 0 for my drive |
| 21:30.21 | tessier__ | bkw_: Sure, I already know C to some degree. But it is a lot more work than writing in a high level language. |
| 21:30.29 | sx-wks | rene-: what sort of application are you doint ? |
| 21:30.44 | zekeonfire | anyone know why I get a few hundred of these a second in my /var/log/asterisk/full log at the debug level "channel.c: Nobody there, continuing..."? |
| 21:31.21 | rene- | sx-wks: the machine is mostly idle when this is happening, i have a pair of astmanager instances probably querying asterisk 2 times per second or so |
| 21:31.50 | rene- | sx-wks i intend to do predictive dialing with asterisk |
| 21:32.10 | rene- | but the dialing script and app server live in a different machine |
| 21:32.18 | sx-wks | what motherboard are you using ? have you tried using another PCI port ? |
| 21:32.22 | bkw_ | rene-, what are you trying to do? |
| 21:32.38 | bkw_ | what do you query from the box every 2 seconds? |
| 21:33.06 | *** join/#asterisk techie (n=techie@adsl-10-4-109.mia.bellsouth.net) |
| 21:33.12 | *** part/#asterisk techie (n=techie@adsl-10-4-109.mia.bellsouth.net) |
| 21:33.19 | *** join/#asterisk designdream (n=designdr@rrcs-71-40-49-30.sw.biz.rr.com) |
| 21:33.24 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 21:33.31 | designdream | so are there any places i can get an iaxy in canada tomorrow? |
| 21:33.46 | designdream | Calgary to be specific |
| 21:33.50 | Preytell | I will try again. Let you know what I figure out. |
| 21:33.51 | *** part/#asterisk sb_mx (n=sb_mx@dsl-200-67-149-101.prod-empresarial.com.mx) |
| 21:33.56 | rene- | bkw: it is every half second, and i am querying to either originate calls or read ast events based on status, agents and queue_status output |
| 21:34.31 | sx-wks | rene-: what motherboard are you using ? have you tried using another PCI port ? |
| 21:34.38 | rene- | the machine is quite hefty P4 3.4 with 2 Gigs of ram, it is a PC in an intelboard, so it is not server class but for 1 E1 it should do |
| 21:34.49 | rene- | sx-wks: intel 865 |
| 21:35.07 | rene- | this was the only port it wouldnt share IRQs |
| 21:35.13 | *** part/#asterisk mountainm2k (n=mountain@216.87.64.218) |
| 21:35.23 | sx-wks | rene-: hah |
| 21:35.50 | rene- | it is 6 slot and i only have a nic in another slot |
| 21:35.57 | sx-wks | I see |
| 21:36.36 | syzygyBSD | how can I record something (a message) that will automatically detect 5 seconds of silence and then continue (agi or dialplan) |
| 21:37.51 | syzygyBSD | nm, record does this |
| 21:38.13 | rene- | in slot 1 it wont share its irq.. i disabled acpi and that didnt make a damn difference,, then i enabled udma and well i saw a difference in the hdparam performance tests but still i see my card going nuts every 4 hours or so for 15-30 minutes. in this events the port resets it self over and over and over until it magically stops doing it |
| 21:39.01 | rene- | i cant force it to stop not by reloading chan zap,. reloading the whole zapata driver or restarting asterisk, only rebooting the whole machine or waiting to fix itself helps\ |
| 21:39.28 | sx-wks | rene-: it's the local union that did some voodoo on the card to allow a sizeable pause to the phone attendants :D |
| 21:39.32 | rene- | after 500 passes zttest claims: 99.987468 |
| 21:39.36 | rene- | heh |
| 21:39.40 | rene- | hehe |
| 21:40.04 | rene- | the auto break feature |
| 21:42.21 | eKo1 | telephony is a black art |
| 21:42.41 | rene- | sx-wks: do you know if hdparm unmaskirq should be 1 or 0 for my particular app? |
| 21:42.47 | *** join/#asterisk _deg_ (n=deg@200.163.193.247) |
| 21:42.50 | rene- | eKo1: very black indeed |
| 21:42.50 | _deg_ | what is the maximum characters that appdata could have? |
| 21:43.03 | _deg_ | something like this: |
| 21:43.31 | eKo1 | check the source code |
| 21:43.33 | _deg_ | exten =200,1,Macro(xxx,xxxxx,xxxxx,xxxxx....almost 200 characters) |
| 21:43.34 | eKo1 | the limits are in there |
| 21:43.43 | _deg_ | eKo1, do you know? |
| 21:43.52 | eKo1 | I could find out easily. |
| 21:43.55 | sx-wks | rene-: what sort of HDD is that ? |
| 21:43.55 | _deg_ | ok |
| 21:44.03 | eKo1 | And so can you. |
| 21:44.46 | _deg_ | what is your suggestion for the first file to look fore? |
| 21:44.48 | _deg_ | for |
| 21:45.07 | _deg_ | app_macro.c? |
| 21:45.26 | _deg_ | eKo1, if you dont kno, never mind. |
| 21:45.28 | _deg_ | eKo1, tks |
| 21:47.03 | eKo1 | Familiarize yourself with the source code. |
| 21:47.21 | eKo1 | Use the source Luke. |
| 21:47.22 | rene- | sx-wks: it is an Hitachi 80 Gigs 7200 rpm ide disk |
| 21:47.27 | eKo1 | err, _deg_ |
| 21:48.02 | rene- | sx-wks: Hitachi Desktar 7K80 |
| 21:48.10 | sx-wks | rene-: hmm... "ide" says it all :D |
| 21:48.45 | _deg_ | found it. |
| 21:48.46 | _deg_ | tks |
| 21:49.01 | *** join/#asterisk muppetmaster (n=jasongoe@81.184.73.169) |
| 21:49.11 | rene- | sx-wks: but every single post that talks about irq misses or pri slips goes on and on about how ide is best than scsi for performance due to udma? |
| 21:49.30 | rene- | and zttool reports zero missed interrupts |
| 21:49.31 | sx-wks | rene-: yet SATA is better :D |
| 21:49.54 | sx-wks | now if there are zero missed interrupts, then the problem lies somewhere else |
| 21:50.05 | sx-wks | like in the options you're running the line with ? |
| 21:50.34 | *** join/#asterisk type_o (i=aaa1@bzq-84-108-62-186.cablep.bezeqint.net) |
| 21:51.55 | rene- | well i have ccs, hdb3 and nocrc4, i have tried both with 1 and 0 as clock source |
| 21:52.38 | Strom_C | rene-: did you call Digium like I suggested? |
| 21:52.57 | rene- | sx-wks: i read on asteriskguru site that i need to enable /* #define CONFIG_ZAPTEL_WATCHDOG */ in order for zttool to actuall read missed irqs |
| 21:52.57 | sx-wks | rene-: mebbe your card has a flaw of some sort |
| 21:53.23 | rene- | Strom_C: no i havent, i will do that now, maybe tha card is flawed |
| 21:53.27 | type_o | anyone experienced with click-to-call and can help out ? |
| 21:53.28 | rene- | s/tha/the |
| 21:56.53 | rene- | LOL In ztconfig.c there is this line: #define HDLC_MAINTAINERS_ARE_MORE_STUPID_THAN_I_THOUGHT |
| 21:57.22 | *** join/#asterisk Altair256 (n=Altair25@mail.clccorp.com) |
| 21:57.33 | sx-wks | rene-: hehehe |
| 21:58.10 | rene- | can i unload zapata, reload the drivers and then reload zapata channel without taking down asterisk?? |
| 21:58.10 | GerbilWrk | if the server is down, do you get a channel unavailable quicker via sip, or iax2? |
| 21:58.17 | Altair256 | Hello. I'm having an issue where the Asterisk Record() application is doing what sounds like silence suppression. The phones (Linksys SPA-941's) have silence suppression turned off, and no other calls have this issue |
| 21:58.21 | rene- | well i mean i have done it, but is it unstable or bad or anything |
| 21:59.00 | Altair256 | Basically.. single words sound quiet, and once you get talking, it sounds louder |
| 21:59.14 | Altair256 | any ideas? |
| 21:59.46 | Strom_C | Altair256: what codec are you recording in |
| 21:59.54 | Altair256 | wav and gsm |
| 21:59.56 | Altair256 | both do it |
| 22:00.26 | Altair256 | that was my first thought too |
| 22:00.42 | Strom_C | so the call path is linksys -> asterisk -> ? |
| 22:01.07 | Altair256 | tmp file |
| 22:01.22 | Altair256 | one sec and I'll get you an example script |
| 22:01.56 | Altair256 | exten => *21,1,Wait(2) |
| 22:01.56 | Altair256 | exten => *21,2,Record(/tmp/custom-weather:gsm) |
| 22:01.57 | Altair256 | exten => *21,3,Wait(2) |
| 22:02.21 | Altair256 | so Linksys -> Asterisk -> file |
| 22:03.03 | *** join/#asterisk flujan (n=flujan@internet.nube.com.br) |
| 22:03.04 | Altair256 | so no ZAP channels involved at this point |
| 22:03.45 | joerns | hi. The website talks about in-line data transmission for iax. Where is the documentation for that please? |
| 22:04.24 | Altair256 | so no other ideas, Strom_C? |
| 22:04.35 | Altair256 | I'll try ulaw just in case |
| 22:04.38 | flujan | hi guys... :) I am trying to use realtime queues... But the statistic fields like queue_callswaiting don't receive any update... :( |
| 22:04.42 | flujan | any ideas? |
| 22:12.14 | *** join/#asterisk Deeewayne (n=dwayne@ool-44c0d56e.dyn.optonline.net) |
| 22:12.42 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.71.249) |
| 22:14.07 | *** join/#asterisk _DAW (n=_DAW@adsl-222-12-239.msy.bellsouth.net) |
| 22:15.01 | _DAW | hello gents |
| 22:16.01 | *** join/#asterisk okdo (n=goldenol@67.99.235.226) |
| 22:16.08 | okdo | hi |
| 22:16.35 | okdo | does this make sense to anyone? "In order to get DTMF be out-of-band we need a 101 in the SDP of your 200 OK. |
| 22:16.36 | okdo | " |
| 22:17.35 | _DAW | payload 101 is telephone-event |
| 22:17.52 | okdo | mmmhmm mmmhmmm, Iam assuming I can add this in somehow? |
| 22:18.11 | _DAW | I have never had to. I just make sure to use rfc2833 dtmf |
| 22:20.58 | flujan | guys someone using realtime queues with asterisk? the statistics and member are not updated |
| 22:21.20 | *** join/#asterisk jbalcomb (n=jbalcomb@216.28.180.158) |
| 22:22.24 | rene- | can you try to read the queue status via the AMI, and see if it is being updated there? |
| 22:22.46 | flujan | hi rene- nice to hear from you. :) |
| 22:22.58 | flujan | rene-, how can I use the ami to get this information? |
| 22:24.00 | [Outcast] | what can play sln files? |
| 22:24.25 | sx-wks | I have a rather annoying bug with chan_sip... [Aug 31 22:33:00] WARNING[14511]: chan_sip.c:6124 determine_firstline_parts: Bad request protocol Packet |
| 22:24.33 | sx-wks | over and over again |
| 22:24.48 | [Outcast] | pastebin a sip debug |
| 22:25.05 | sx-wks | and a nice segfault when I call out |
| 22:25.10 | rene- | flujan: the ami is a tcp port that you can connect using telnet |
| 22:25.11 | Nugget | telnet is eeeeeeevil! |
| 22:25.17 | [Outcast] | hehe |
| 22:26.08 | rene- | you need to enable ami in manager.conf and check the voip-wiki to learn how to issue commands to it.. |
| 22:26.41 | sx-wks | [Outcast]: it's the cirpack keepalive packet |
| 22:26.56 | sx-wks | (was working in 1.2.11) |
| 22:27.18 | l-fy | sx-wks > that's what you get for not using bayonne |
| 22:27.29 | flujan | rene-, OK... i will triy it. |
| 22:27.39 | l-fy | sx-wks > you didn't change a bit |
| 22:27.40 | sx-wks | (she likes it) |
| 22:28.07 | l-fy | sx-wks > actually no, you are mistaken with your mom |
| 22:29.00 | sx-wks | whatever |
| 22:31.10 | rene- | sx-wks: i am running mysql on that asterisk server, do u think it has to do anything with the problems i have been describing? |
| 22:31.33 | sx-wks | rene-: possibly |
| 22:31.45 | [shodan] | anyone knows how to get that sip software 1.6.7 and bootrom 3.2.1 from polycom ? do you have to register and say you're a var or something ? because I registered as an individual and the software in the "prc" is even older than the publically available software ! |
| 22:31.53 | l-fy | rene- > which problems? |
| 22:32.05 | Altair256 | Strom_C, looks like it is in fact my hard phones |
| 22:32.22 | Altair256 | I tried from an SPA-2002 with a standard analogue phone attached and the problem isn't there |
| 22:32.26 | Altair256 | thanks for your help |
| 22:32.38 | rene- | I-fy: E1 pri slips |
| 22:33.34 | [Outcast] | sx-wks: after a breif seach you are not the only one having problem with the cirpack, have you check the bug tracker? |
| 22:33.46 | sx-wks | [Outcast]: was about to |
| 22:33.57 | [Outcast] | i would check there |
| 22:34.11 | [Outcast] | what type of system are you running on? |
| 22:34.28 | [Outcast] | and what type of device are you connecting to? |
| 22:35.16 | l-fy | rene- > mmmmmmm, it is possibile |
| 22:35.21 | l-fy | in certain situations |
| 22:35.26 | l-fy | but not very likely |
| 22:35.38 | l-fy | if you have e1 slips i doubt is from mysql |
| 22:35.58 | l-fy | because in that case the d channel will also have slips and you will lose the d-channel |
| 22:36.14 | sx-wks | [Outcast]: I'm in the same situation as the one in bug 7261 |
| 22:36.33 | l-fy | sx-wks > which problem with cirpack? |
| 22:36.41 | l-fy | the ack issue? |
| 22:37.01 | rene- | I-fy: i dont know if this is what you mean but messages says that chan_zap.c: No D-channels available! Using Primary channel 16 as D-channel anyway! |
| 22:37.32 | *** join/#asterisk type_o (i=aaa1@bzq-84-108-62-186.cablep.bezeqint.net) |
| 22:38.21 | rene- | I-fy: what happens is roughly every 5 hours i get half hour of the E1 port restarting, alarming, recovering and restarting again all channels over and over again, until it stops doing it by itself |
| 22:38.55 | l-fy | rene- > are you losing the mysql connection than? |
| 22:38.59 | rene- | no |
| 22:39.12 | l-fy | than why will it be mysql related? |
| 22:39.17 | rene- | mysql stays up and my scripts that connect to manager interfase stays up |
| 22:39.31 | rene- | i dont know it is just that mysql runs in the same box as asterisk |
| 22:39.45 | l-fy | rene- > i think is not related |
| 22:39.47 | rene- | and gets lightly used while this E1 port is idle |
| 22:39.53 | l-fy | rene- > may i sugest you something? |
| 22:40.00 | rene- | be my guest |
| 22:41.16 | *** join/#asterisk Rez (i=lorez@freenode/staff/lorez) |
| 22:43.22 | sx-wks | [Outcast]: then when I try to call out, I get this: http://rafb.net/paste/results/N63uey64.html |
| 22:43.30 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
| 22:43.33 | *** join/#asterisk RoyK (n=roy@ti211110a081-1433.bb.online.no) |
| 22:43.39 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-9-118.cybersurf.com) |
| 22:47.37 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 22:47.49 | [shodan] | yay ! I just got my spa-2102 to work ! |
| 22:48.00 | *** part/#asterisk WEH (n=bob@pool-71-250-60-139.nwrknj.east.verizon.net) |
| 22:48.10 | [shodan] | why do they have to mislabel the server field with "proxy" |
| 22:49.14 | TripleFFFF | hmm . cauz its basicaly not a server.. |
| 22:49.28 | TripleFFFF | not serving anything.. its redirecting a sip call to another host .. or in most case a client |
| 22:49.33 | TripleFFFF | no ? |
| 22:49.40 | RoyK | ~nickometer [shodan] |
| 22:49.41 | TripleFFFF | techniclay at least |
| 22:49.58 | TripleFFFF | #define nicoketer lameness params ? |
| 22:50.20 | RoyK | :) |
| 22:52.27 | [shodan] | some sip phones call it "server" others call it "proxy" it's more a server, because if you have a sip client , it wants a sip server , and the asterisk isn't a proxy to another "sip server" |
| 22:53.24 | eKo1 | actually, the correct term would be sip proxy server |
| 22:53.35 | TripleFFFF | hehe |
| 22:53.36 | TripleFFFF | yeah |
| 22:53.44 | TripleFFFF | coz its SERVE's as a PROXY |
| 22:53.57 | TripleFFFF | so its serving as a proxy |
| 22:54.05 | TripleFFFF | as a web server servers html pages |
| 22:54.25 | TripleFFFF | asterisk doesnot .. but hmmIVR is generated by asterisk. .hence in that case its a server |
| 22:54.36 | *** join/#asterisk Pazzo (n=thomas@host130-250-static.72-81-b.business.telecomitalia.it) |
| 22:54.44 | TripleFFFF | so if you rebound to pstn its proxying.. if you attend an ivr.. or VM its.. a server |
| 22:54.44 | TripleFFFF | ;) |
| 22:54.54 | jontow | woo, i fucked up. |
| 22:54.55 | TripleFFFF | now i see why its confusing ;) |
| 22:54.56 | jontow | heh :o |
| 22:55.04 | TripleFFFF | wonton? |
| 22:55.08 | TripleFFFF | wo town |
| 22:55.39 | *** join/#asterisk DrukenHME (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 22:55.45 | [shodan] | why not call web servers proxy too , they are just a proxy to the files on the hard drive ! ;) |
| 22:56.33 | [shodan] | and then we could call proxy servers like squid "proxy proxy" ! |
| 22:57.31 | [shodan] | I probably just caused the head of 2 or 3 persons to explode ;) |
| 22:57.31 | jontow | gaaah, no wonder.. sucker doesn't walk the agent linked list, just sits on one of them and spinlocks |
| 22:58.07 | JT | proxy^2 |
| 23:00.16 | sx-wks | [Outcast]: reported as bug 7854 |
| 23:00.19 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 23:00.55 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 23:01.38 | *** join/#asterisk DrukenHME (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 23:02.11 | *** join/#asterisk jnorell (n=jesse@slug.kci.net) |
| 23:02.14 | *** join/#asterisk Strom_C (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 23:02.19 | Strom_C | join #asstricks |
| 23:02.21 | Strom_C | er |
| 23:02.26 | yatesy | rofl |
| 23:02.42 | *** part/#asterisk jnorell (n=jesse@slug.kci.net) |
| 23:03.24 | Strom_C | i mean hello :) |
| 23:03.24 | *** part/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 23:06.29 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 23:06.32 | blitzrage | app_voicemail.c:2276 inboxcount: Failed to obtain database object for 'asterisk'! seems to show up in a loop once I load res_odbc -- how do I tell app_voicemail to ignore DB stuff? |
| 23:06.53 | blitzrage | this is trunk btw: |
| 23:06.56 | [shodan] | can I send a caller id message on a fxs without making it ring ? |
| 23:08.18 | blitzrage | [shodan]: don't think so since the callerID is transmitted during the ring cycle |
| 23:08.21 | DrukenHME | yeah, turn the ringer on the phone off :) |
| 23:08.27 | sx-wks | lol |
| 23:09.14 | JT | well technically, i'm not sure if ring current is required to send callerid |
| 23:09.34 | JT | but the likely answer is most phones are designed only to receive CID during the ringing stage, regardless |
| 23:09.55 | [shodan] | hmm , what about making the ringer ring for a very small fraction of a second ? (if you can make custom distinctive rings what about one that is very short) |
| 23:10.10 | JT | experiment? :) |
| 23:11.17 | [shodan] | I'm still far from getting the distinctive rings to work , let alone custom ones :\ |
| 23:12.34 | JT | so why don't you want it to ring? |
| 23:13.48 | [shodan] | I want to display "You received a fax" on the caller id |
| 23:14.22 | blitzrage | "FAX RCVD" might be better since it'd probably actually fit ;) |
| 23:14.31 | blitzrage | however, that doesn't really help your situation right now :) |
| 23:15.28 | *** join/#asterisk GC-mirc (n=me@63.163.62.9) |
| 23:15.37 | blitzrage | night |
| 23:15.56 | JT | all, playing silly buggers trying to get analogue phones to do more than they were designed to, right :) |
| 23:16.33 | JT | the phone would only display it while it continued to ring, or if picked up |
| 23:16.41 | hmmhesays | haha |
| 23:16.46 | JT | otherwise the user would have to manually scroll through the log |
| 23:16.53 | JT | s/all/ahh/ |
| 23:17.01 | JT | shutup jbot |
| 23:18.44 | [shodan] | yeah I know , this is just a subtle notice , because the phone's lcd also light up when it receives a call |
| 23:19.01 | JT | hmm |
| 23:24.19 | [shodan] | k , I managed to force the spa-2102 to use a custom ring , but a very short ring cause the phone not to detect ringing at all |
| 23:25.44 | *** part/#asterisk bkw_ (n=bkw_@asterisk/friend-and-developer/bkw) |
| 23:26.22 | *** join/#asterisk techie (n=techie@adsl-10-4-109.mia.bellsouth.net) |
| 23:28.08 | *** part/#asterisk GC-mirc (n=me@63.163.62.9) |
| 23:32.20 | Strom_C | you know, I have far too many voip phones |
| 23:33.18 | [shodan] | k, assuming I understand the format of those cadence ring strings , the shortest ring that at least this phone can detect (a nortel telecom 350) is 0.065 seconds , but it needs at least 0.07 to reliably detect the caller id signal |
| 23:38.05 | *** join/#asterisk DrukenHME (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 23:40.56 | [shodan] | anyone knows how to use alert_info with a linksys spa-2102 ? |
| 23:46.51 | *** join/#asterisk Here_And_There (n=Here_And@pool-71-246-6-164.phlapa.fios.verizon.net) |
| 23:47.04 | JT | Strom_C: too many traditional phones, too? |
| 23:47.21 | Strom_C | JT: oh, that's been the case for a long time |
| 23:47.36 | Strom_C | but now I also am passing the threshold of too many voip phones |
| 23:48.31 | JT | heh |
| 23:48.35 | JT | mobiles? |
| 23:49.14 | JT | got a few mobile phones too i would guess :P |
| 23:49.23 | Strom_C | i despise mobile phones |
| 23:49.26 | Strom_C | only have the one :) |
| 23:49.44 | [shodan] | is that the same as that mythical "too many computers" but for phone geeks ? ;) |
| 23:50.20 | Strom_C | yep |
| 23:50.21 | justinu|laptop | i really hate cell phones as well |
| 23:50.35 | Strom_C | i currently have on my desk two cisco 7960s, a polycom 430, and a linksys 942 |
| 23:50.40 | JT | Strom_C: ok that's weird, what about pay phones then? |
| 23:50.50 | Strom_C | JT: those are in the living room |
| 23:50.56 | blitzrage | 7970, 7960, 501, 942, UIP200 :) |
| 23:51.02 | JT | heh |
| 23:51.09 | JT | i think i have 4 mobiles |
| 23:51.12 | JT | only use 2 though |
| 23:53.17 | *** join/#asterisk DrukenHME (n=jdumais@CPE00121716da99-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 23:55.44 | *** join/#asterisk dalekurt (n=dalekurt@port0033-afw-adsl.cwjamaica.com) |
| 23:55.59 | pyrom | This is starting to get me mad, i can't understand why my E70 doesnt work correctly with asterisk (register ok, rings, dials out) When someone calls i can't hear them(just mumbling/half words), but they can hear me. when i call someone, they can hear me but i still can't hear them correctly (just half words) |
| 23:56.12 | pyrom | But if i transfer a call via asterisk to the device, it works flawlessly! |
| 23:56.21 | Strom_C | pyrom: what if you get an inbound call via SIP? |
| 23:56.24 | Strom_C | or IAX2 |
| 23:56.38 | Strom_C | and bypass the sipura? |
| 23:57.05 | pyrom | Ok, that might be the issue - but havent been able to try that. |