| 00:00.11 | jaytee | Qwell, DING!!! We have a winner!!! "Tell him what he's won, Jay!" |
| 00:00.15 | Corydon76-dig | for 1.6, I believe that's correct |
| 00:00.21 | Qwell | A BRAND NEW CAAAARRRRR!!! |
| 00:00.28 | jaytee | hehehe |
| 00:00.40 | *** join/#asterisk abatista (n=ariel_@c-24-127-219-186.hsd1.fl.comcast.net) |
| 00:00.41 | Corydon76-dig | but I think Qwell packaged that version, so he'd be the one to ask |
| 00:00.46 | *** join/#asterisk DJ_HaMsTa (i=k@c-69-136-240-75.hsd1.nj.comcast.net) |
| 00:00.51 | Qwell | what version? |
| 00:00.52 | Corydon76-dig | grins at Qwell |
| 00:01.10 | DJ_HaMsTa | any one having a problem with les.net where it fails to re-register once in a while ? |
| 00:01.25 | emrahpbx | Corydon76-dig: yeah installed that also... make checkconfig tells me everything is fine, but still getting errors. also after doing make uninstall-all and reinstall again... |
| 00:01.26 | drmessano | DJ_HaMsTa: Turn qualify off |
| 00:01.35 | drmessano | DJ_HaMsTa: They're probably throttling you |
| 00:02.00 | LuisTorres | Hi |
| 00:02.10 | DJ_HaMsTa | whats qualify ? |
| 00:02.25 | ACK-NAK | Qwell: Is /etc/init.d/dahdi restart a different concept than "service dahdi restart" |
| 00:02.28 | LuisTorres | does anybody know how to use Outbound fax detection? |
| 00:02.29 | Corydon76-dig | YAY FOR GUIS |
| 00:02.54 | Qwell | ACK-NAK: sort of. the former isn't too distro-specific. the latter does the former, on RH-based system.s |
| 00:03.25 | Corydon76-dig | DJ_HaMsTa: ask your GUI provider |
| 00:03.52 | ACK-NAK | Qwell: I appreciate it. So therefore best pracitce may be to use /etc/init.d/asterisk restart over service ast... |
| 00:04.07 | Qwell | ACK-NAK: RedHat would tell you otherwise, but yes. :) |
| 00:04.12 | *** join/#asterisk MaliutaLap (n=biteme@203.171.192.119) |
| 00:04.36 | ACK-NAK | Qwell: :-) |
| 00:05.06 | *** join/#asterisk speedwagon (n=ariel_@c-24-127-219-186.hsd1.fl.comcast.net) |
| 00:05.33 | MrNeutr0n | are there any tools to help parse through the "full" log to see what is happening with each call? |
| 00:05.39 | MrNeutr0n | a visual tracing type of thing... |
| 00:05.45 | MrNeutr0n | (speaking of guis...) |
| 00:05.46 | Qwell | MrNeutr0n: no |
| 00:06.14 | rue_mohr | MrNeutr0n, possibly you dont have enough lines? |
| 00:06.36 | MrNeutr0n | ah, rue_mohr I do wish that was the case |
| 00:06.55 | MrNeutr0n | however there are two bonded PRIs |
| 00:06.57 | rue_mohr | MrNeutr0n, I do agree its REALLY hard to follow a single call in the logs |
| 00:07.00 | *** join/#asterisk cheriff (n=davidm@58.96.27.155) |
| 00:08.22 | MrNeutr0n | and I did some grepping to find out that at no time was there ever any channel greater than Zap/21-1 for instance |
| 00:09.22 | MrNeutr0n | So at this point my thinking is that I need to determine if the problem is coming from some sort of asterisk misconfiguration |
| 00:09.24 | rue_mohr | hmm, maybe you should have asterisk phone itself half the number of lines you have |
| 00:09.42 | rue_mohr | you using zaptel or dahdi? |
| 00:09.59 | MrNeutr0n | rue_mohr, I've got zaptel with the wanrouter modules for an A200 |
| 00:10.18 | rue_mohr | would you like to pastebin your config? |
| 00:10.28 | MrNeutr0n | rue_mohr, I've thought about that, but again I am having the problem with only a very few simultaneous calls |
| 00:11.11 | *** join/#asterisk riddlebox (n=victoria@75-132-225-75.dhcp.stls.mo.charter.com) |
| 00:16.02 | LuisTorres | anybody knows anything for outgoing fax detection? |
| 00:17.05 | rue_mohr | so I have the office going over their user guide |
| 00:17.08 | rue_mohr | their excited |
| 00:17.11 | rue_mohr | this is good |
| 00:17.31 | *** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu) |
| 00:17.52 | rue_mohr | hmm the user guide for the aastra is pretty thick |
| 00:18.32 | *** join/#asterisk nicoAMG (n=superunk@201.203.50.42) |
| 00:19.19 | *** join/#asterisk dlewis (i=4579ba4a@about/security/staff/dlewis) |
| 00:19.57 | MrNeutr0n | ok so here is something that I thought might be a part of the problem: |
| 00:20.09 | MrNeutr0n | from the asterisk console when I type "zap show channel X" |
| 00:20.29 | rue_mohr | anyone want to tell me how to have a dialplan put somone on hold? Im reading ... |
| 00:20.30 | MrNeutr0n | it will show that there is a Caller ID |
| 00:20.43 | MrNeutr0n | however it also appears to be hug up |
| 00:20.54 | MrNeutr0n | and it's not ringing either |
| 00:21.18 | MrNeutr0n | and i end up with a lot of these in such a state |
| 00:21.49 | MrNeutr0n | but some others show "Caller ID:" blank when (I'm guessing) they're supposed to be |
| 00:22.26 | MrNeutr0n | Could this lead to the error that: Not yet hungup... calling hangup once with icause"? |
| 00:22.39 | *** join/#asterisk coppice (n=chatzill@201.193.17.210.dyn.pacific.net.hk) |
| 00:22.57 | _charly_ | has anyone already made a res_irc for asterisk? |
| 00:23.00 | rue_mohr | you have a T1, I have NO idea WHY your disconnect signaling wouldn't work |
| 00:24.19 | rue_mohr | can extensions.conf put a call on hold? |
| 00:24.40 | rue_mohr | I realize the problem with that idea, I dont care |
| 00:24.47 | *** join/#asterisk JonOnt (n=nonya@72.34.90.74) |
| 00:25.16 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 00:25.27 | rue_mohr | do parked calls get hold music, they must |
| 00:26.19 | JonOnt | Hey guys, is any one familiar with this issue, if i transfer an incoming call to another extention and that extentions transfers the call to another extention, if it gets tranfered again the call is lost? Any one know whats going on here? |
| 00:26.30 | *** join/#asterisk MaliutaLap (n=biteme@203.171.192.118) |
| 00:26.40 | Qwell | JonOnt: Asking in multiple channels like that is considered incredibly rude |
| 00:26.43 | rue_mohr | and between them all you can transfer ok? |
| 00:27.19 | rue_mohr | does it get anymore answers? |
| 00:27.32 | JonOnt | Qwell: sorry about that, i can see why that would be rude, wont do it again |
| 00:28.24 | *** part/#asterisk rhombus (n=rhombus@dsl-vlan435-66-18-218-36.nucleus.com) |
| 00:28.30 | JonOnt | rue_mohr: yes, transfers work fine, but we can only transfer a call three times max then it is lost, my boss calls alot and will talk to every one in the office, he ends up having to call two or three times |
| 00:29.03 | rue_mohr | do you think the phones might be doing the trasfers locally between them? |
| 00:29.23 | rue_mohr | aka, are you sure asterisk is handling the transfers |
| 00:29.32 | _charly_ | JonOnt: i've seen this too, using snom phones over sip and debians asterisk, it seems that the call is only lost when it is transferred back to the first phone. haven't checked this any further yet (just noticed it 2 days ago) |
| 00:30.38 | JonOnt | rue_mohr: was pretty sure asterisk is handling the transfers, im using aastra 57i's |
| 00:31.09 | rue_mohr | do you see them in the console? |
| 00:31.18 | JonOnt | _charly_: at least some one else has heard of this, ive been googling and havnt found any real hits on that yet |
| 00:31.25 | JonOnt | rue_mohr: yes |
| 00:35.51 | _charly_ | i haven't checked that yet because transfers are no problem for us, we only have 14 phones, and transfers are done about once a week. but we have some stability issues, 2 of the snoms are crashing and rebooting very often, i still have no clue why :/ |
| 00:37.10 | _charly_ | anyway, i have to go, good night :) |
| 00:40.19 | JonOnt | I just captured some CLI logs on this issue, looks like macro-hangupcall is the one doing the hanging up |
| 00:40.29 | JonOnt | This might be a trixbox issue |
| 00:41.30 | jaytee | huh? how'd I end up joining #trixbox? damn IRC client screwed up again. |
| 00:48.42 | *** join/#asterisk Gopher_77 (n=Jim@cpe-71-72-19-206.neo.res.rr.com) |
| 00:50.03 | Gopher_77 | I have cable internet service, and the NAT that they control seems to be getting in the way of my SIP service. Is there a way to get around this? |
| 00:52.17 | riddlebox | Gopher_77, are you forwarding the ports? |
| 00:52.29 | Gopher_77 | riddlebox: I can't forward the ports; I don't have control of that |
| 00:53.23 | riddlebox | Gopher_77, do you have a router? |
| 00:53.48 | Gopher_77 | riddlebox: yes, and I can forward those ports, but my router itself is provided a private IP address by my cable company |
| 00:54.21 | riddlebox | Gopher_77, where are you? |
| 00:54.28 | Gopher_77 | riddlebox: Ohio |
| 00:55.13 | riddlebox | which cable co |
| 00:55.18 | Gopher_77 | riddlebox: Time Warner |
| 00:55.49 | riddlebox | do you have an asterisk server up right now? |
| 00:55.52 | *** join/#asterisk doolph (n=doolph@190.141.71.191) |
| 00:55.53 | Gopher_77 | riddlebox: yes |
| 00:55.54 | doolph | hi |
| 00:56.02 | doolph | anyone got a softswitch or somethng? |
| 00:56.45 | riddlebox | Gopher_77, check that msg I query I sent you |
| 00:59.29 | *** join/#asterisk sack (n=sack@196.Red-83-49-103.dynamicIP.rima-tde.net) |
| 00:59.51 | *** join/#asterisk edibrac (n=elusive4@206.173.193.34.ptr.us.xo.net) |
| 01:05.48 | bmoraca | Gopher_77: call your cable company and tell them to switch your cable modem to bridge mode. if they provided the cable modem, they should be able to do that. if you provided it, then you'll need to figure out how to get into it and do that via its web interface. any cable modem should be able to do this easily. |
| 01:07.10 | Gopher_77 | bmoraca: thanks |
| 01:08.11 | drmessano | Bridge mode on a cable modem? |
| 01:08.14 | bmoraca | that will give your router's WAN side a public IP and remove the possibility of double NAT |
| 01:08.26 | bmoraca | drmessano: cable companies do some stupid things sometimes. |
| 01:08.42 | drmessano | What kind of modem is it? |
| 01:08.46 | bmoraca | drmessano: most cable modems and DSL modems now-adays include a basic NAT router |
| 01:10.07 | *** join/#asterisk dlewis (i=4579ba4a@about/security/staff/dlewis) |
| 01:10.46 | drmessano | DSL modems yeah, but not cable |
| 01:11.02 | drmessano | Most are dumb DOCSIS boxes unless you get business class service |
| 01:13.17 | bmoraca | i'd agree...though i've seen it more and more often. personally, i think it's a waste of money. give me a native bridge from the delivery media to ethernet and i'll be happy. |
| 01:16.17 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-7fd0205809486c52) |
| 01:17.52 | bmoraca | it's quittin time |
| 01:33.53 | *** join/#asterisk nix8n82 (n=nate@63.162.27.243) |
| 01:34.38 | *** part/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 01:42.34 | *** join/#asterisk eric2 (n=nobody@69.60.247.142) |
| 01:43.11 | *** join/#asterisk killown (n=Yamato@unaffiliated/killown) |
| 01:44.29 | *** join/#asterisk digitalirony (i=digitali@my.grandma.uses.shellium.org) |
| 01:53.04 | *** join/#asterisk StephenF (n=none@198.144.201.109) |
| 01:58.27 | *** join/#asterisk killown (n=Yamato@unaffiliated/killown) |
| 01:58.43 | *** join/#asterisk Steve_J-obs (n=Chris123@pool-71-190-72-110.nycmny.east.verizon.net) |
| 01:58.55 | Steve_J-obs | hello everybody |
| 01:59.45 | digitalirony | steve jobs huh? |
| 02:00.01 | digitalirony | I didn't think you were well enough to get on a computer anymore :P |
| 02:01.32 | drmessano | Hows the AZT? |
| 02:01.56 | Steve_J-obs | I am well enough to hang out with my friends in this forum...just dont tell the press |
| 02:02.11 | Corydon76-dig | You have friends in this forum? |
| 02:02.30 | Corydon76-dig | ducks |
| 02:02.33 | Steve_J-obs | drmessano: AZT? |
| 02:02.58 | drmessano | yeah, isnt Steve Jobs HIV positive? |
| 02:03.06 | Corydon76-dig | Cancer, not AIDS |
| 02:03.09 | digitalirony | nah, he has pancreatic canceer |
| 02:03.13 | digitalirony | *cancer |
| 02:03.15 | Steve_J-obs | dr messano: no man, it is cancer |
| 02:03.46 | drmessano | Well, rumor has it he's HIV positive, and his "pancreatic cancer" is really KS |
| 02:03.53 | Steve_J-obs | wow |
| 02:03.58 | digitalirony | hrmm |
| 02:04.17 | Steve_J-obs | coming to think of it.. he is very thin |
| 02:04.29 | digitalirony | doesn't mean he has aids |
| 02:04.34 | icebrew54 | yeah that's fubar'd |
| 02:04.40 | icebrew54 | apple stock is going to drop like a mofo when he gets sick |
| 02:04.44 | icebrew54 | and/or passes ;\ |
| 02:04.50 | Corydon76-dig | Pancreatic cancer is a much more serious affliction nowadays and would spook the market more than HIV |
| 02:04.50 | digitalirony | I don't think so |
| 02:04.52 | icebrew54 | not that I own any... |
| 02:04.56 | drmessano | and theres documents, which may or may not be forged, that show he failed the test |
| 02:05.08 | digitalirony | hmm |
| 02:05.09 | drmessano | Dunno |
| 02:05.09 | *** join/#asterisk Daejeo (n=chatzill@114.201.159.78) |
| 02:05.18 | Steve_J-obs | pancreatic cancer means that he is not going to be around in 4 years |
| 02:05.20 | icebrew54 | yeah pancreatic cancer is no joke, grandma had that stuff and it took her in under 6 months |
| 02:05.34 | digitalirony | well I don't think steve jobs death will hurt apple too much, they are a rather solid company |
| 02:05.35 | icebrew54 | chemo, meds etc don't work on it |
| 02:05.37 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 02:05.39 | digitalirony | well more so than M$ |
| 02:05.43 | icebrew54 | digitalirony: ha |
| 02:05.46 | drmessano | I dont buy the "Hormonal inbalance" crap |
| 02:05.46 | Corydon76-dig | So the idea that pancreatic cancer is a cover story holds no water |
| 02:05.48 | icebrew54 | digitalirony: how old are you? |
| 02:05.51 | drmessano | Theres no way |
| 02:05.54 | icebrew54 | :P |
| 02:06.04 | digitalirony | icebrew54: 22 |
| 02:06.05 | icebrew54 | apple was hurting until the ipod showed up |
| 02:06.07 | icebrew54 | as I recall... |
| 02:06.16 | digitalirony | might HAVE been |
| 02:06.19 | icebrew54 | and I mean MS giving them money because MS was scared of a monopoly |
| 02:06.19 | digitalirony | but is NO longer |
| 02:06.22 | drmessano | He may have had pancreatic cancer too.. who knows.. its not an opportunistic infection, but anything is possible |
| 02:06.23 | Corydon76-dig | icebrew54: You recall incorrectly |
| 02:06.27 | Daejeo | Sun Fire X2200 M2 Server/Sun Fire X2250 Server which one would be better for running asterisk? |
| 02:06.35 | Corydon76-dig | icebrew54: their savior was the original iMac |
| 02:06.36 | icebrew54 | Corydon76-dig: yeah I'm pretty sure they were hurting |
| 02:06.42 | icebrew54 | yeah okay timing was off |
| 02:06.47 | icebrew54 | same difference I guess |
| 02:06.48 | Daejeo | http://www.sun.com/servers/x64/x2250/ |
| 02:06.59 | Steve_J-obs | Steve Jobs made Apple, and if he is gone, Apple will collapse |
| 02:07.00 | Daejeo | http://www.sun.com/servers/x64/x2200/ |
| 02:07.00 | icebrew54 | they were hurting for a couple years...financially speaking |
| 02:07.09 | digitalirony | icebrew54: well for that matter microsoft/billG wass hurting till he stole and sold DOS |
| 02:07.17 | Steve_J-obs | he has rescued Apple over and over |
| 02:07.20 | digitalirony | everyone was hurting before they weren't hurting |
| 02:07.22 | digitalirony | thats dumb |
| 02:07.34 | drmessano | Kids dont know Apple = Steve Jobs.. only fanboys do |
| 02:07.35 | icebrew54 | digitalirony: apple has been in the dumps for quite some time, I mean it was an ample amount of time they were hurting man |
| 02:07.40 | Corydon76-dig | In any case, this is rather off-topic for #asterisk, so please let that be the end of it |
| 02:07.40 | icebrew54 | macII to iMAC = a long time |
| 02:07.51 | Steve_J-obs | that's why out of admiration for the man, btw |
| 02:07.57 | Daejeo | drmessano: any advise |
| 02:08.10 | drmessano | iPods mean "iPod" to 14 year olds.. they could care less about some balding freak with black turtlenecks and jeans with no belt |
| 02:08.20 | digitalirony | LOL |
| 02:08.55 | icebrew54 | I don't think they are as solid as described |
| 02:09.07 | digitalirony | ipods? |
| 02:09.14 | Corydon76-dig | icebrew54: Stop, please |
| 02:09.14 | icebrew54 | I think the ipod was a savior for sure... apple speaking |
| 02:09.18 | Corydon76-dig | digitalirony: you, too |
| 02:09.20 | digitalirony | :P |
| 02:09.23 | digitalirony | Im quite |
| 02:09.26 | icebrew54 | ok join me in #stevejobstalk |
| 02:09.31 | digitalirony | nah |
| 02:09.37 | digitalirony | *quiet |
| 02:09.40 | icebrew54 | bleh, I win then by default :P |
| 02:09.52 | digitalirony | shrugs...doesn't matter to him |
| 02:09.52 | Corydon76-dig | icebrew54: I really mean it. Stop. |
| 02:10.54 | Steve_J-obs | by the way guys, I have a question how to setup yum on this new godaddy server... I want to install asterisk on it, but it does not have the kernel-devel |
| 02:10.55 | Corydon76-dig | If you want to talk about Asterisk on the Mac, that's fine, but talk about Jobs' health is off-topic in here |
| 02:11.26 | icebrew54 | ok, I'm done....I got off topic no worries from me |
| 02:11.28 | Steve_J-obs | anybody wants to help Steve Jobs here? |
| 02:11.29 | Corydon76-dig | Setup yum? |
| 02:12.07 | Corydon76-dig | If yum isn't installed, it's probably not the right package manager |
| 02:12.32 | Steve_J-obs | well, godaddy provided me this dedicated server, and when I do "yum install kernel-devel" it gives me a message about.. |
| 02:12.56 | Steve_J-obs | ...no packages available for update |
| 02:13.05 | *** join/#asterisk JAMMAN2110 (n=James@unaffiliated/jamman2110) |
| 02:13.07 | icebrew54 | centos? |
| 02:13.08 | rob0 | yum on a Godaddy server is a bit off topic as well. :) |
| 02:13.13 | Corydon76-dig | Ah, you've probably got RHEL, then |
| 02:13.21 | icebrew54 | rob0: ohhhh SNAP! |
| 02:13.26 | Steve_J-obs | well... the topic is about installing asterisk on centos |
| 02:13.31 | Gopher_77 | anyone here familiar with voipuser? |
| 02:13.38 | Steve_J-obs | actually, on RHE 4 |
| 02:13.45 | Corydon76-dig | Yeah, that's the problem |
| 02:13.48 | rob0 | I never used Centos, so I'd better go. Bye. |
| 02:14.05 | Corydon76-dig | You aren't subscribed to that update repo, so you can't retrieve any packages on it |
| 02:14.06 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 02:14.06 | *** mode/#asterisk [+o russellb] by ChanServ |
| 02:14.15 | Corydon76-dig | Evening, Russell |
| 02:14.20 | Gopher_77 | anybody here know if voipuser is congested right now? |
| 02:14.24 | Steve_J-obs | yes, I agree...what do I do? |
| 02:14.43 | Corydon76-dig | Talk to your sysadmin about changing your RHEL subscription |
| 02:14.55 | Steve_J-obs | I am my own sysadmin |
| 02:15.01 | *** join/#asterisk djMax (n=chatzill@c-65-96-17-196.hsd1.ma.comcast.net) |
| 02:15.39 | Corydon76-dig | Well, you'd either need to get CentOS or get a RH subscription |
| 02:15.55 | Steve_J-obs | whats an rh subscription? |
| 02:16.02 | djMax | anybody have good SIP trunk provider recommendations? And perhaps ballpark pricing? |
| 02:16.10 | Corydon76-dig | Purchase an update agreement from RedHat |
| 02:16.10 | djMax | RH = RedHat |
| 02:16.14 | *** join/#asterisk CrazyTux (n=brandon@ip68-111-67-4.oc.oc.cox.net) |
| 02:16.22 | Steve_J-obs | oh noo |
| 02:16.25 | Corydon76-dig | djMax: there are a ton of them on voip-info.org |
| 02:16.33 | djMax | yeah, that's the problem. :) |
| 02:16.41 | icebrew54 | djMax: http://www.dslreports.com/forum/remark,15568032 |
| 02:16.43 | Corydon76-dig | djMax: by location, pricing, etc. |
| 02:16.49 | icebrew54 | I found that link helpful Djmax |
| 02:16.56 | Steve_J-obs | are there any other repo servers I can use? |
| 02:16.56 | djMax | thanks, perfect. |
| 02:16.56 | icebrew54 | in fact I bookmarked it... |
| 02:17.07 | icebrew54 | djMax: you owe me a russian bride for that link |
| 02:17.19 | icebrew54 | djMax: it was gold for me :P |
| 02:17.20 | Corydon76-dig | Steve_J-obs: You'd do better asking in a centos forum |
| 02:17.21 | icebrew54 | hehe |
| 02:17.47 | djMax | hmmm. How about Ukranian, I'm all out of Russian brides at the moment. |
| 02:17.51 | Corydon76-dig | Steve_J-obs: because converting to centos is basically what you'd need to do |
| 02:18.33 | Steve_J-obs | centos and RHEL as far as I know, is the same thing |
| 02:18.45 | djMax | these look consumerish, I'm looking for biz provider. Super low latency, 99.9+, etc. |
| 02:19.02 | Corydon76-dig | Steve_J-obs: in terms of the underlying platform, yes. In terms of the support system, repos, and packaging, no |
| 02:19.57 | drmessano | djMax: yes, that list is consumerish |
| 02:20.00 | Steve_J-obs | mmm. sounds like I have to change that flavor of linux...I tell you, godaddy really sucks |
| 02:20.02 | drmessano | djMax: and outdated |
| 02:20.14 | drmessano | djMax: Try voip-info |
| 02:21.03 | djMax | yeah, reading it now. I saw one price like 2.5 cents/min, that seem right? |
| 02:21.04 | Corydon76-dig | djMax: the only way you're going to know is to try a whole bunch of them |
| 02:21.16 | drmessano | djMax: No, thats VERY expensive |
| 02:21.23 | Corydon76-dig | djMax: for domestic? |
| 02:21.24 | drmessano | Flowroute.com is a good one |
| 02:21.30 | djMax | yeah, domestic |
| 02:21.31 | Pryon | likes flowroute |
| 02:21.39 | Corydon76-dig | 2.5 c/min is good to Europe, though |
| 02:21.40 | drmessano | Les.net as well |
| 02:21.46 | djMax | basically trying to offer a VXML service in a datacenter |
| 02:21.54 | djMax | (where getting a real T would be hell) |
| 02:22.10 | Corydon76-dig | djMax: to do what? |
| 02:22.11 | djMax | will be almost ALL calls within Massachusetts |
| 02:23.00 | djMax | click-to-call initially, voice search later. |
| 02:23.00 | Corydon76-dig | djMax: your best bet is actually a cross-connect within the facility. Takes a fraction of the time to provision |
| 02:23.00 | djMax | probably Prophecy-powered. |
| 02:23.00 | Corydon76-dig | Find out what telcos have equipment in the same facility |
| 02:23.42 | djMax | wow, .0098. Yeah, I asked our provider (Internap), they basically said "we're great for everybody" |
| 02:25.06 | Corydon76-dig | So all your calls are outbound? |
| 02:25.14 | djMax | initially, yeah. |
| 02:25.54 | djMax | given expected volume, assuming Flowroute latency and perf was good, I'd be done because it's so low that I don't mind a mistake. |
| 02:26.08 | djMax | As opposed to voxeo hosting which is like $500/month + per minute |
| 02:26.51 | Corydon76-dig | I have no problem with flowroute (I've met a few of their people) |
| 02:27.09 | [TK]D-Fender | Corydon76-dig: isn't a "few" all of them? :) |
| 02:27.35 | Corydon76-dig | [TK]D-Fender: dunno, I assume they left a few people at home to keep the phones answered |
| 02:28.06 | djMax | I like that you can signup online without going through some opaque quoting process. |
| 02:28.35 | [TK]D-Fender | Corydon76-dig: ..... please stay on the line to maintain your calling priority! |
| 02:29.25 | [TK]D-Fender | djMax: Opaque has been deprecated in favour of milky-translucence :) |
| 02:29.36 | LemensTS | TK: ive been reading everything i can find on google, i am still stuck http://pastebin.com/m1765c582 |
| 02:29.38 | djMax | I've used * with Prophecy, but anybody used another VXML server they liked? |
| 02:29.47 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
| 02:30.08 | *** join/#asterisk legis (n=wad@unaffiliated/legis) |
| 02:31.20 | LemensTS | TK: Im not sure if i need to forget about get_variable, and look into STDIN like on http://www.voip-info.org/wiki/view/Asterisk+AGI+php, or what I should do. Ive tried that and other stuff already...need to be pointed in the right direction at least :) |
| 02:31.43 | icebrew54 | god, now that I used the asterisk-gui...to make my life easier |
| 02:32.10 | icebrew54 | it's making life much more difficult since I want to do some stuff outside of the gui's realm |
| 02:32.32 | *** join/#asterisk obnauticus (n=lol@about/windows/regular/obnauticus) |
| 02:33.43 | [TK]D-Fender | LemensTS: DeadAGI is supposed to be for hung-up channels, not LIVE channels. Dead channels don't have access to channel vars <- |
| 02:34.05 | [TK]D-Fender | LemensTS: try AGI, not DeadAGI. |
| 02:35.52 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
| 02:38.07 | Corydon76-dig | [TK]D-Fender: or use 1.6.0, where AGI becomes DeadAGI at hangup |
| 02:38.32 | [TK]D-Fender | Oh Asterisk 1.6.0 is there nothing you CAN'T do! :) |
| 02:38.50 | [TK]D-Fender | watches * 1.6.0 ride in on a Unicorn... |
| 02:41.55 | *** join/#asterisk Deeewayne (n=dwayne@c-76-29-245-9.hsd1.al.comcast.net) |
| 02:41.55 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 02:42.10 | djMax | Yeah, I'm running 1.4 and the *Now 1.5 CD is sitting in the tray, tantalizingly waiting for me to press the reset button and go for broke to 1.6 |
| 02:43.18 | [TK]D-Fender | djMax: See you wouldn't have to flush your OS if you'd just roll your own... |
| 02:46.21 | LemensTS | TK: http://pastebin.com/m3d106ecf I set it to AGI, and added line 8...still getting same thing...any other pointers? |
| 02:51.18 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
| 02:52.19 | docelmo | ~book |
| 02:52.20 | jbot | extra, extra, read all about it, book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 02:53.20 | [TK]D-Fender | HAHAH! |
| 02:53.25 | LemensTS | docelmo: i read ch. 9 about an hour ago |
| 02:53.27 | LemensTS | :P |
| 02:53.29 | [TK]D-Fender | LemensTS: pwned |
| 02:53.55 | [TK]D-Fender | LemensTS: Got me a guess! |
| 02:53.59 | [TK]D-Fender | 1 sec |
| 02:54.09 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 02:54.09 | *** mode/#asterisk [+o denon] by ChanServ |
| 02:55.37 | [TK]D-Fender | LemensTS: DARN, so close, thought I had it |
| 02:56.40 | [TK]D-Fender | LemensTS: I'm pretty sure its a PHP issue, not AGI |
| 02:56.56 | [TK]D-Fender | LemensTS: thought it was an assignment issue.. reviewing my syntax |
| 02:57.16 | *** join/#asterisk CamelMenthol (n=ben@rrcs-67-53-153-186.west.biz.rr.com) |
| 02:57.42 | *** join/#asterisk Khratos (n=Khratos@190.80.197.20) |
| 02:57.58 | CamelMenthol | Hi everyone :) |
| 02:58.10 | eric2 | any 416 or 647'ers around? |
| 02:58.57 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.194) |
| 02:59.03 | [TK]D-Fender | eric2: 404\ |
| 02:59.05 | CamelMenthol | Anyone work much with the Manager API? |
| 02:59.15 | *** join/#asterisk Rabenklaue (n=Rabe@f049012041.adsl.alicedsl.de) |
| 02:59.21 | LemensTS | TK: yea i tried $matt = array(); before $matt = $agi->get_variable("var1"); and that didn't help. Im just a little above average php programmer tho. |
| 02:59.42 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 02:59.42 | *** mode/#asterisk [+o denon] by ChanServ |
| 03:00.29 | CamelMenthol | Hi denon :) |
| 03:01.04 | Rabenklaue | Hello, I have a small problem concerning my asterisk server and the openvpn server running on the same host. PC1 is connected to the internet over the openvpn network on SERVER. If I call the asterisk on SERVER with the 192.168.0.2 it works, but when calling it with 192.168.20.2 via ekiga it doesn't react at all. |
| 03:01.12 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
| 03:01.30 | Rabenklaue | Could anybody please help me with the setup of the asterisk NAT configuration? |
| 03:01.30 | CamelMenthol | Rabenklaue: Have you explicitly set a bind address? |
| 03:01.56 | Rabenklaue | bindaddr=0.0.0.0 |
| 03:02.09 | Rabenklaue | from sip.conf |
| 03:02.14 | Rabenklaue | But this looks like the default value |
| 03:02.44 | CamelMenthol | Rabenklaue: Yea that should be the default value. And it should work |
| 03:02.59 | CamelMenthol | Rabenklaue: Is your 192.168.20.2 interface a tap/tun? |
| 03:03.29 | Rabenklaue | CamelMenthol: Yes, it's a tap interface |
| 03:03.40 | *** join/#asterisk Chilling_Silence (n=Josiah@121.98.143.77) |
| 03:04.06 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
| 03:04.12 | [TK]D-Fender | LemensTS: I'm not much better.... |
| 03:04.25 | [TK]D-Fender | Rabenklaue: READ <- |
| 03:04.27 | [TK]D-Fender | ~sipnat |
| 03:04.28 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 03:04.30 | [TK]D-Fender | ^^^^^^^^^^^^^^^^ |
| 03:04.33 | CamelMenthol | Rabenklaue: Can you try setting the bindaddr to the tap ip? |
| 03:05.00 | CamelMenthol | [TK]D-Fender: I had similar issues. I think it's something to do with the virtual interface |
| 03:05.10 | Chilling_Silence | Quick Q - Does anybody have a moment to help me try and diagnose a CDR logging bug? |
| 03:05.16 | CamelMenthol | Chilling_Silence: I can try |
| 03:05.25 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-deb5f6f49a3dff10) |
| 03:05.28 | [TK]D-Fender | CamelMenthol: I'd want to see configs & SIP debug. |
| 03:06.28 | Chilling_Silence | Legend CamelMenthol :) thanks |
| 03:06.28 | Chilling_Silence | Basically, inbound call -> Ext (205 for example), then transfer to External number |
| 03:06.28 | Chilling_Silence | My theory is the inbound call should be logged, the transfer should be logged, but the call to the external number and then the joining with the inbound call *wont* be logged, when it should be |
| 03:07.30 | CamelMenthol | What sort of CDR logging do you have? |
| 03:08.16 | Rabenklaue | CamelMenthol: Nope, this also doesn't work |
| 03:08.49 | CamelMenthol | Rabenklaue: Why is it an issue to have to use the real interfaces address instead of the VPN ? |
| 03:09.09 | CamelMenthol | Chilling_Silence: Have you tried setting "unanswered = yes" in cdr.conf? |
| 03:09.32 | Chilling_Silence | CamelMenthol: No, to be honest I havent, what does that do? |
| 03:09.47 | Chilling_Silence | Ive been using the MySQL logging |
| 03:10.25 | CamelMenthol | Chilling_Silence: It forces a CDR record to be created even if a call is never answered by asterisk or an extension. Which may be the case with your setup, I'm not sure what your config is. I also use MySQL and using this asterisk manager desktop app I wrote, I needed to enable this to keep track of some calls that get strewn about |
| 03:10.48 | [TK]D-Fender | LemensTS: What ver of *? |
| 03:10.57 | *** join/#asterisk sah-work (n=Bawbatos@adsl-75-63-18-243.dsl.pltn13.sbcglobal.net) |
| 03:11.09 | Rabenklaue | CamelMenthol: As I'm not sure whether I'm at home to do my calling jobs or not. I'm using a mobile device to get connected to my home network. So I want it to be a consistent way of using it. |
| 03:11.29 | Rabenklaue | On the other hand, as I'm connected to the local network the internal IPs work, too |
| 03:11.39 | rob0 | Speaking of Chilling, it's darn COLD in Alabama, and I want to lodge a complaint with Digium support about that. |
| 03:11.48 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.226.70) |
| 03:11.52 | Chilling_Silence | rob0: :D |
| 03:11.55 | CamelMenthol | rob0: I'm in Wisconsin... |
| 03:12.03 | Chilling_Silence | CamelMenthol: Thanks mate, will give that a whirl, appreciate your time |
| 03:12.06 | Rabenklaue | Ok, I think it isn't necessary at all right now, but thanks anyhow |
| 03:12.31 | CamelMenthol | Rabenklaue: No problem. I had to settle on doing a little more intense config as well because of using OpenVPN w/ * |
| 03:12.42 | [TK]D-Fender | rob0: Oh, how cold? |
| 03:13.39 | CamelMenthol | Chilling_Silence: There is also another param in cdr.conf "endbeforehexten" try messing with that as well |
| 03:13.40 | rob0 | 24F now, headed down to 16F. And I know it's colder in Canada, but it's your own damn fault for being there. :) |
| 03:14.03 | CamelMenthol | 17F, feels like 2F here.... |
| 03:14.24 | [TK]D-Fender | rob0: try sub-zero :) Farenheit :p |
| 03:14.40 | [TK]D-Fender | rob0: Feels WORSE |
| 03:15.18 | rob0 | But guys, I'm a thin-blooded redneck in the deep south! I moved here to get away from that!! |
| 03:15.21 | Chilling_Silence | CamelMenthol: No idea what that does, looks semi-useful, will give it a whirl. Thanks for pointing me in the right direction :) |
| 03:15.40 | rob0 | Let's start a flame war to stay warm. |
| 03:15.51 | [TK]D-Fender | gets some kindling |
| 03:15.58 | CamelMenthol | Chilling_Silence: http://pastebin.com/d2f612edf |
| 03:16.26 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 03:16.27 | CamelMenthol | Shit I'll bring in a truck load of trolls if someone can point me in the direction of some manager 1.1 documentation :) |
| 03:16.47 | Chilling_Silence | CamelMenthol: Ah yeah, just saw that in the cdr.conf :D |
| 03:17.12 | Chilling_Silence | CamelMenthol: AMI? |
| 03:17.40 | *** join/#asterisk RobertLaptop (n=rmiddle@96.244.48.200) |
| 03:17.48 | CamelMenthol | Chilling_Silence: Yes |
| 03:18.01 | Chilling_Silence | CamelMenthol: what specifically are you wanting to do with it ? |
| 03:18.40 | *** join/#asterisk keebler (n=keebler@h20.148.20.98.dynamic.ip.windstream.net) |
| 03:19.05 | CamelMenthol | Chilling_Silence: I was using asterisk-java but my company upgraded to * 1.6 w/ AMI 1.1 and now asterisk-java no longer works. I am basically looking for enough documentation on what parameters to send with commands for asnwering, transferring, getting peer/channels statues, dbput/get, dahdi channel mgmt, etc. |
| 03:19.28 | CamelMenthol | I think I am going to write a full java library for AMI 1.1 |
| 03:21.32 | Chilling_Silence | Theres some semi-decent stuff on voip-info.org, and I saw another page just recently which helped me enough for making a Web-to-Call page |
| 03:21.46 | Chilling_Silence | http://www.the-asterisk-book.com/unstable/asterisk-manager-api.html -- Is a start :) |
| 03:21.48 | CamelMenthol | I've seen voi-info.org, some of their info is out of date |
| 03:22.56 | CamelMenthol | I think what I'm going to do is write a program to pry through the source code for various modules to find out what parameters they look for in the AMI |
| 03:23.10 | CamelMenthol | I'm not even seeking documentation I guess, just a list lol |
| 03:23.24 | *** join/#asterisk RouterWeasel (n=johnm@core.spokanecomputing.com) |
| 03:23.43 | russellb | isn't that info available through the asterisk CLI? |
| 03:24.13 | CamelMenthol | russellb: No, just the commands and what scopes are required to use it |
| 03:24.23 | CamelMenthol | all, system, call, command, etc... |
| 03:25.09 | CamelMenthol | Not scopes, permissions. my bad |
| 03:25.45 | CamelMenthol | There are some commands that are well documented, but others have nothing |
| 03:25.48 | russellb | It should also print out the description, though, which includes all the parameters |
| 03:25.50 | russellb | ah .. |
| 03:27.01 | CamelMenthol | My boss is willing to pay me to create a fairly concise Java library for * integration so I'm seeking to do most of the commands. Or atleast the relevant ones for use in a workstation-based end-user or administrative application |
| 03:27.03 | russellb | i would be ok with calling it a bug for any actions that are not sufficiently documented to tell you what headers are required (and optional) |
| 03:27.24 | CamelMenthol | I'm just gonna blame it on laziness heh :) |
| 03:27.32 | russellb | meaning if you want to put it on bugs.digium.com, we can get it taken care of |
| 03:27.44 | russellb | otherwise you're stuck reading the code to figure it out, i guess, heh |
| 03:27.58 | russellb | the o'reilly book has a section on AMI, as well |
| 03:27.58 | CamelMenthol | For now I'm just going to get the basics taken care of for my program then in the future maybe I'll press into the issue further |
| 03:28.03 | russellb | can't remember how much they have |
| 03:28.08 | russellb | k. |
| 03:30.04 | russellb | where'd the nick come from btw? |
| 03:30.09 | russellb | just a smoker? heh |
| 03:30.16 | CamelMenthol | My cigarettes I'm smoking today |
| 03:30.22 | russellb | ah :) |
| 03:30.45 | CamelMenthol | I usually hate menthols but I'm enlisting in the marine corps and leaving in 19d so I need to quit and these nasty cigs are forcing me to start stopping |
| 03:31.14 | russellb | heh, oh dear .. |
| 03:31.21 | russellb | boot camp? |
| 03:31.30 | CamelMenthol | I always thought people were full of shit when they said quitting gives you the most unusual thoughts and dreams. But damn. The last two weeks I've been slowing myself down I have had the most vivid and ridiculous dreams ever |
| 03:31.33 | *** join/#asterisk chendy (n=chatzill@58.60.219.128) |
| 03:31.34 | CamelMenthol | Yup |
| 03:31.41 | russellb | well i wish you the best |
| 03:31.44 | CamelMenthol | Thanks a lot |
| 03:32.21 | Chilling_Silence | CamelMenthol: Unfortunately still having the same issues, its not logging, even though Ive changed both those options in cdr.conf and restarted asterisk .. :( At least its some info for me to add to the bugreport |
| 03:32.22 | CamelMenthol | I was going to a private uni but then ran out of money to continue. and in fear of getting stuck in a stagnant job market and working the same freelance crap for years, I decided to enlist and do reserves for a year to finish my last year of uni |
| 03:32.43 | CamelMenthol | Chilling_Silence: Other calls are logging just fine though eh? |
| 03:33.57 | Chilling_Silence | Yup, everything else logs brilliantly |
| 03:35.52 | Chilling_Silence | http://bugs.digium.com/view.php?id=14398 |
| 03:36.55 | CamelMenthol | Wish I could help you more pal, sorry |
| 03:37.44 | Chilling_Silence | No worries, thanks for pointing me towards the cdr.conf though, it was a good start :) appreciate your time |
| 03:38.11 | CamelMenthol | No problem :) |
| 03:39.25 | russellb | Chilling_Silence: the first thing you're going to hear on the bug is that you need to try the latest version |
| 03:39.28 | russellb | which is 1.4.23.1 |
| 03:39.40 | *** join/#asterisk slashdotfx (n=slashdot@ip-252-29.mitra.net.id) |
| 03:39.57 | LemensTS | any agi people have a clue on why GET VARIABLE var1 will show 41283, but when I do vardump($matt) 41283 is not contained in $matt |
| 03:40.00 | LemensTS | http://pastebin.com/m729ef76d |
| 03:42.11 | Gopher_77 | I have asterisk using an SIP account at voipuser, it's registered, but the CLI with verbose says this for every call: == Everyone is busy/congested at this time (1:0/0/1). Does anybody know how to fix this? |
| 03:42.29 | Chilling_Silence | russellb: Yeah Im just trying to put together a non-production box that I can use for testing |
| 03:43.39 | Chilling_Silence | Gopher_77: Outbound routes? |
| 03:43.50 | Gopher_77 | Chilling_Silence: yes, outbound |
| 03:44.32 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 03:44.33 | Chilling_Silence | Gopher_77: No i mean have you defined outbound routes? |
| 03:44.49 | Gopher_77 | Chilling_Silence: apparently not; how do I do that? |
| 03:45.42 | Gopher_77 | Chilling_Silence: I hope you don't live up to your name now... |
| 03:46.04 | russellb | Chilling_Silence: ok. |
| 03:47.09 | Chilling_Silence | russellb: There has been a bit of activity I can see in the changelog, but yeah .. :-/ |
| 03:47.29 | Chilling_Silence | Gopher_77: You installed just asterisk or you using a distro like trixbox / elastix / asteriskNOW? |
| 03:50.08 | Gopher_77 | Chilling_Silence: just * |
| 03:50.23 | Gopher_77 | Chilling_Silence: do you know if I can check to see if there are minutes in the pot at voipuser? |
| 03:50.40 | Gopher_77 | Chilling_Silence: nvm, there are plenty :) |
| 03:50.50 | Chilling_Silence | Gopher_77: No idea, Im from New Zealand ;) |
| 03:51.12 | Gopher_77 | Chilling_Silence: location doesn't matter; what matters is that it's free ;) |
| 03:51.31 | MrNeutr0n | hi everyone i have two T1s in an A102 and I want to only use a single D-chan |
| 03:51.48 | MrNeutr0n | however i am pretty clueless |
| 03:51.51 | Chilling_Silence | And to be honest, Im no whizz with plain asterisk, prefer to fluff around with the likes of FreePBX (Plz dont shoot me people), but yeah, sounds like you need to define an outbound route so your calls know which sip trunk to use |
| 03:51.52 | MrNeutr0n | any idea where i should look first? |
| 03:52.33 | *** part/#asterisk drfreeze (n=Jim@207.191.114.82) |
| 03:52.34 | *** join/#asterisk baliktad (i=baliktad@c-24-16-23-12.hsd1.wa.comcast.net) |
| 03:53.09 | MrNeutr0n | actually i think i have the zaptel.conf/zapata.conf thing down |
| 03:53.18 | MrNeutr0n | but wanpipe is where i run into trouble |
| 03:53.49 | MrNeutr0n | can i configure two physical interfaces as a single logical one? |
| 03:53.51 | *** part/#asterisk Khratos (n=Khratos@190.80.197.20) |
| 03:53.57 | CamelMenthol | MrNeutr0n: I've never dealt with anything more than POTS as far as telco side, is there some reason you dont upgrade to dahdi? |
| 03:54.34 | MrNeutr0n | unfortunately my client is currently ... bewitched? by trixbox |
| 03:54.39 | MrNeutr0n | 2.6.1.13 nevertheless... |
| 03:54.45 | CamelMenthol | Gotcha |
| 03:54.54 | MrNeutr0n | (: |
| 03:56.33 | CamelMenthol | What kind of hardware connects the T1's to your box? |
| 03:56.45 | MrNeutr0n | CamelMenthol, ha - actually I haven't ever really dealt with anything beyond pots myself! |
| 03:56.54 | MrNeutr0n | sangoma A102 |
| 03:57.03 | CamelMenthol | Well I can try and help you work through it :) I always like learning new shit |
| 03:57.33 | MrNeutr0n | haha cool - well, at this point i think we are reduced to finding somebody else's config files =D |
| 03:58.21 | russellb | if you have the zaptel config down, guess you should just get a digium card |
| 03:58.26 | russellb | then that would be all you have to configure :-p |
| 03:59.02 | *** join/#asterisk docelmo (n=vircuser@pool-151-199-187-233.lyn.east.verizon.net) |
| 03:59.13 | CamelMenthol | MrNeutr0n: Do you have one A102 getting both T1s or two? |
| 03:59.43 | MrNeutr0n | just the one A102 |
| 03:59.53 | MrNeutr0n | setup-sangoma gave me two files |
| 03:59.58 | MrNeutr0n | wanpipe1.conf and wanpipe2.conf |
| 04:00.08 | CamelMenthol | PasteBin? |
| 04:00.11 | MrNeutr0n | each are similar with the exception that some "1"s are replaced with "2"s |
| 04:01.58 | MrNeutr0n | 1 sec i nuked it |
| 04:02.12 | Juggie | is there a doc anywhere on chan_mobile? |
| 04:03.03 | CamelMenthol | Juggie: Have you read voip-info.org |
| 04:03.50 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 04:04.19 | Juggie | CamelMenthol, nope, i guess i should start there :) |
| 04:04.40 | *** join/#asterisk hadi- (n=Hadi@CPE002129717ae3-CM001a668ee8b2.cpe.net.cable.rogers.com) |
| 04:05.00 | hadi- | hello... is there a way to set a max of 2 concurrent calls for an incoming DID |
| 04:05.02 | hadi- | on asterisk 1.4? |
| 04:06.10 | CamelMenthol | hadi-: Have you tried using a channel status split in the dialplan |
| 04:06.21 | hadi- | nope.. |
| 04:09.58 | [TK]D-Fender | hadi-: "core show function GROUP_COUNT" |
| 04:10.03 | CamelMenthol | There ya go! |
| 04:10.09 | CamelMenthol | Built in solution :) |
| 04:11.32 | [TK]D-Fender | #freepbx [23:10]<hadi->is there a way to set a max of 2 concurrent calls for an incoming DID under freepbx? |
| 04:11.40 | [TK]D-Fender | CamelMenthol: Now with strings attached! |
| 04:12.08 | *** join/#asterisk ocnarf (n=chatzill@122.2.251.67) |
| 04:13.30 | ocnarf | Need help.. Im using AgentcallbackLogin for my queues. The problem is even they are already logged in, asterisk doesnt say that they are already login.. Any idea? |
| 04:13.47 | ocnarf | Here is my dialplan: exten => 2323,1,AgentCallbackLogin(,,${CALLERID(num)}@from-internal) |
| 04:14.12 | ocnarf | Agent can login again even they are already logged in |
| 04:15.56 | [TK]D-Fender | ocnarf: Show us the login attempt, your agent dump from CLI, your queue dumps, etc. One big pastebin |
| 04:15.59 | [TK]D-Fender | ~pb |
| 04:15.59 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 04:16.01 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^ |
| 04:16.10 | *** join/#asterisk riddlebox (n=victoria@75-132-225-75.dhcp.stls.mo.charter.com) |
| 04:16.47 | CamelMenthol | riddlebox: ICP fan? |
| 04:20.01 | ocnarf | D-Fender: http://pastebin.com/d182a3435 |
| 04:20.09 | ocnarf | Hope that helps |
| 04:20.36 | CamelMenthol | [TK]D-Fender: Just curious, how long you been working with asterisk and linux? |
| 04:20.56 | [TK]D-Fender | CamelMenthol: About 5 years |
| 04:21.01 | riddlebox | CamelMenthol, used to be |
| 04:21.48 | [TK]D-Fender | ocnarf: OH, I think I misread. Indeed * does not say they are already logged in. |
| 04:22.08 | [TK]D-Fender | ocnarf: thats jsut the way it is... agents can hop around because there is no dedicated log OUT feature |
| 04:22.29 | riddlebox | Gopher_77, did you get everything working? |
| 04:22.38 | Juggie | [TK]D-Fender, have you ever setup chan_mobile? |
| 04:22.41 | [TK]D-Fender | ocnarf: Its only if you don't pass an exten to login to that it will then prompt and if you tell it "blank" THEN it will log you out |
| 04:23.17 | [TK]D-Fender | Juggie: Nope... I should though since I don't have POTS..... don't need VoIP... * might be more for call recording, so using my cell as FXO would be nifty.. |
| 04:23.27 | [TK]D-Fender | Juggie: Gotta get me a compatible BT adapter first :) |
| 04:23.45 | CamelMenthol | [TK]D-Fender: Can you do it via USB? |
| 04:24.06 | [TK]D-Fender | CamelMenthol: "It"? You mean chan_mobile? IIRC its only BT |
| 04:24.23 | CamelMenthol | Yea alright |
| 04:25.10 | Gopher_77 | riddlebox: sip isn't working |
| 04:25.28 | riddlebox | on voicepulse? |
| 04:25.57 | riddlebox | or voipuser |
| 04:26.14 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com) |
| 04:27.39 | [TK]D-Fender | Gopher_77: IAX may be for you... |
| 04:27.49 | [TK]D-Fender | Gopher_77: Or proxy it through a VPN relay |
| 04:28.46 | riddlebox | <PROTECTED> |
| 04:29.23 | [TK]D-Fender | riddlebox: Registering isn't the issure |
| 04:29.29 | riddlebox | ohh ok |
| 04:29.46 | Gopher_77 | [TK]D-Fender: x-lite uses my SIP provider ok |
| 04:29.47 | [TK]D-Fender | riddlebox: its getting that inbound call when the mapping isn't fresh and it decides to start hijacking... |
| 04:30.07 | riddlebox | eww |
| 04:30.09 | ocnarf | D-Fender: Is there anything i can do so agent wont be able to login when they are already logged in? |
| 04:31.39 | [TK]D-Fender | ocnarf: What is the negative consequence to a double-login for you? |
| 04:31.58 | [TK]D-Fender | ocnarf: it still requires a pass.... |
| 04:33.09 | ocnarf | D-Fender: We have an app which counts the login time of the agent. But looking at the logs, it shows the there are multiple login |
| 04:33.10 | Gopher_77 | very interesting: when I use x-lite successfully, my voipuser says that I'm NOT behind NAT |
| 04:34.33 | [TK]D-Fender | ocnarf: Ah... then you'll need to make some special script to see if they are logged in first and if they are call the AgentCallbacklogin for them. |
| 04:34.52 | *** join/#asterisk ScribbleJ (n=nsj@c-67-172-6-141.hsd1.il.comcast.net) |
| 04:35.09 | [TK]D-Fender | Gopher_77: Maybe your modem is proxying it. I've heard of some that track inside clients.... those with soft-phone accounts, etc... |
| 04:35.41 | ocnarf | D-Fender: hmm.. I guess there is no workaround using AgentCallbackLogin |
| 04:35.51 | ocnarf | D-Fender: thanks! |
| 04:35.56 | [TK]D-Fender | ocnarf: Not directly |
| 04:38.30 | Gopher_77 | [TK]D-Fender: when I set * up as non-nat, voipuser still detects the nat |
| 04:41.16 | Gopher_77 | [TK]D-Fender: when I set up * as nat, voipuser doesn't detect nat |
| 04:41.48 | *** join/#asterisk loompek (n=NoName@noname.rula.net) |
| 04:41.52 | *** part/#asterisk loompek (n=NoName@noname.rula.net) |
| 04:41.56 | *** join/#asterisk loompek (n=NoName@noname.rula.net) |
| 04:41.58 | loompek | morning |
| 04:42.00 | loompek | smee again |
| 04:43.13 | [TK]D-Fender | Gopher_77: thats the point... |
| 04:43.41 | [TK]D-Fender | Gopher_77: When you tell * its behind NAT then it will present the WAN IP appropriately so the remote end doesn't not NEED to go hunt you down. |
| 04:46.42 | loompek | can asterisk answer with 'moved temporarily' command in sip? |
| 04:47.12 | loompek | 3xx redirection |
| 04:47.49 | loompek | transfer... |
| 04:47.50 | loompek | hmm |
| 04:48.33 | [TK]D-Fender | loompek: Might work if you haven't answered and they are OK getting a 100 Trying |
| 04:48.45 | *** join/#asterisk CunningPike (n=arodgers@S01060014bf81366b.vc.shawcable.net) |
| 04:51.31 | Gopher_77 | what is the third argument to Dial? |
| 04:52.00 | carrar | show application dial |
| 04:52.15 | carrar | Dial(Technology/resource[&Tech2/resource2...][|timeout][|options][|URL]) |
| 04:52.47 | carrar | Do you mean "options"? |
| 04:53.03 | thehar | i want asterisk to respond with http error 418 |
| 04:53.09 | thehar | and Hangup() |
| 04:53.10 | Gopher_77 | carrar: that's probably it, after the second comma |
| 04:53.26 | carrar | run the show command as there are LOTS of options |
| 04:54.51 | [TK]D-Fender | thehar: When my PHONE gets an HTTP 418 ... then I'll feel rejected |
| 04:55.02 | carrar | I'm tall |
| 04:55.09 | thehar | hehe |
| 04:55.09 | carrar | I can't get a 418 error |
| 04:55.44 | carrar | I don't think Asterisk supports rfc2324 |
| 04:57.37 | *** join/#asterisk sah-work (n=Bawbatos@adsl-75-63-18-243.dsl.pltn13.sbcglobal.net) |
| 04:59.47 | Gopher_77 | I'm getting some dialplan entries and I don't know where they came from. Does anyone know where they might be coming from? |
| 05:00.23 | Gopher_77 | "created from 'pbx_config'" |
| 05:00.35 | [TK]D-Fender | Gopher_77: extensions.conf <- |
| 05:00.51 | [TK]D-Fender | perhaps some users.conf... |
| 05:01.07 | [TK]D-Fender | Gopher_77: Perhaps you could... SHOW US |
| 05:01.07 | frogonwheels | [TK]D-Fender: I foundmy 3-way call issue. It was actually your answer to somebody else that clued me in to it.. which is that the MusicOnHold is associated with the a particular channel. |
| 05:01.22 | frogonwheels | [TK]D-Fender: All I needed to do was press <flash> once more. |
| 05:01.48 | Gopher_77 | [TK]D-Fender: looks like it was users.conf. thanks :) |
| 05:01.49 | [TK]D-Fender | frogonwheels: Glad you found it |
| 05:02.00 | [TK]D-Fender | ~users.conf |
| 05:02.00 | jbot | users.conf is a flaming pile of sh1t that takes the fine control of several perfectly usable * config files and reduces them to the lowest common denominator and makes your system behave like a "toaster grade" PBX system. |
| 05:03.10 | Gopher_77 | lol toaster grade? |
| 05:03.21 | Gopher_77 | ~toaster grade |
| 05:03.34 | frogonwheels | [TK]D-Fender: Kinda weird that the MOH was playing on a hung-up channel - but it's similar to what happens when the other party hangs up.. so not inexpicable. |
| 05:04.04 | [TK]D-Fender | Gopher_77: DUMB APPLIANCE |
| 05:04.11 | frogonwheels | [TK]D-Fender: hmm. so my instincts on that particular config were correct then (users.conf). |
| 05:12.35 | *** join/#asterisk joako (n=joako@99-153-162-33.lightspeed.miamfl.sbcglobal.net) |
| 05:13.11 | joako | I got a SwitchVox CD from Digium but when I tried to boot it, it wouldn't. Upon closer inspection the disc only contains an ISO file... :wtf: |
| 05:13.48 | frogonwheels | joako: huh.. easy mistake to make. |
| 05:13.55 | frogonwheels | joako: why don't you just mount it then? |
| 05:14.26 | frogonwheels | .. like growisofs myiso.iso instead of growisofs +myiso.iso (I think) :) |
| 05:16.11 | [TK]D-Fender | joako: Careful... its like crack.... |
| 05:16.48 | [TK]D-Fender | joako: Turn the other cheek and the shit'll never stop flowing :p |
| 05:18.14 | Qwell | joako: from where? |
| 05:18.20 | Qwell | from where did you get the CD, that is |
| 05:18.44 | Qwell | miami...itexpo? |
| 05:19.23 | *** join/#asterisk rue_mohr (n=rue@h24-207-90-17.cst.dccnet.com) |
| 05:21.47 | rue_mohr | if I may stretch my luck and ask a question without looking for an answer first, if you have a music on hold box, with an audio out, can asterisk get its hold music from the audio card to use the external box? (lets just assume the box already cost $300 and scraping the recording from it isn't an idea met in a friendly way) |
| 05:23.25 | joako | Qwell: Yes.... |
| 05:23.32 | [TK]D-Fender | rue_mohr: Sure |
| 05:23.49 | rue_mohr | I mean its not like you can just link /dev/dsp to /var/... /moh/... |
| 05:23.54 | rue_mohr | cook |
| 05:23.57 | rue_mohr | er cool |
| 05:24.12 | rue_mohr | [TK]D-Fender, you do sleep, right? |
| 05:24.18 | frogonwheels | rue_mohr: http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf /Using a sound card as the source |
| 05:24.29 | rue_mohr | ooo |
| 05:24.35 | rue_mohr | thankyou! |
| 05:24.41 | frogonwheels | don't know if it's the best way.. |
| 05:24.48 | joako | rue_mohr: I always see him here, I assume not. Then again I usually sleep mornings... |
| 05:25.15 | Qwell | joako: thanks, I'll poke somebody... |
| 05:25.25 | Qwell | unless you want to, of course. since you're there :p |
| 05:26.30 | joako | Qwell: Maybe I'll find Mark again tomorrow |
| 05:27.40 | rue_mohr | well thats enough of an answer for me |
| 05:27.52 | [TK]D-Fender | rue_mohr: Almost time for my 8 minutes.... |
| 05:27.56 | rue_mohr | hahaha |
| 05:28.05 | rue_mohr | for refernce, I'm in bed now |
| 05:28.29 | rue_mohr | I'm so geek there's a computer shelf built into the bed :/ |
| 05:28.39 | rue_mohr | with its own computer... |
| 05:29.39 | ScribbleJ | Dumb one - how do I see logging output that is LOG_DEBUG ? |
| 05:30.17 | LemensTS | rue_mohr: if you got a lap you got a computer shelf in bed... |
| 05:30.20 | *** join/#asterisk kyawthu (n=kyaw@213.206.89.31) |
| 05:30.24 | rue_mohr | core set debug something? |
| 05:30.53 | rue_mohr | well this involved making the bed about 18" longer than the matress |
| 05:31.13 | LemensTS | what do you lay on your stomach |
| 05:31.18 | LemensTS | that hurts my neck |
| 05:31.20 | rue_mohr | yea |
| 05:31.39 | rue_mohr | it involves pillow craftyness |
| 05:32.22 | LemensTS | hah. i just have a mattress on the floor against the wall, and i sit up against the wall with the laptop on my lap...kiss lol |
| 05:33.40 | rue_mohr | http://eds.dyndns.org/~ircjunk/house/dscn7993.jpg |
| 05:34.12 | [TK]D-Fender | rue_mohr: Gah |
| 05:34.20 | rue_mohr | its interesting how having a bed 5' off the floor feeds a paranoia about heights |
| 05:34.36 | rue_mohr | other downside: DONT SIT UP FAST IN MORNING |
| 05:34.37 | LemensTS | lmao u sleep on plywood? |
| 05:34.53 | [TK]D-Fender | rue_mohr: Building a a computer shelf into a bed is sacrilege. |
| 05:34.54 | rue_mohr | no, I decided there was too much clearance and put a matress on there |
| 05:35.15 | rue_mohr | and a pillow |
| 05:35.24 | LemensTS | should we donate money so rue can buy a new desktop |
| 05:35.33 | [TK]D-Fender | Proper bedroom should not have a computer... laptop or otherwise. |
| 05:35.45 | rue_mohr | no worries, thats before I finished moving in |
| 05:35.51 | rue_mohr | hah, mine has 3 |
| 05:36.16 | LemensTS | i got a 7 foot 4 post rack in my bedroom :shrug: |
| 05:36.16 | rue_mohr | 4 if you count the old 486 thats a step for the cat to get to the bed |
| 05:37.19 | rue_mohr | my asterisk machine is actually in my room, the houses data closet is out of space |
| 05:37.50 | rue_mohr | I cant use the 1U machines cause even with 60% of the fans unplugged its too loud for hte nearby bedroom |
| 05:37.56 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-8a12d774d1b272de) |
| 05:38.35 | rue_mohr | you can only put so many machines in a 1' x 2' closet before your at its thermal capacity anyhow |
| 05:39.12 | LemensTS | rue: lol yea i had a rackmount switch in my room and it didnt take long for me to move it out of here |
| 05:39.12 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 05:39.12 | *** mode/#asterisk [+o denon] by ChanServ |
| 05:39.49 | sipy | I sleep better with the hum of fans. kinda like rain |
| 05:40.01 | rue_mohr | gets real quiet in here right after a powerfailure, before I start the genorator |
| 05:40.08 | rue_mohr | mmm white noise |
| 05:41.38 | rue_mohr | tries to reclaim some realestate from the cat |
| 05:44.43 | ScribbleJ | rue_mohr, core set debug was the key, thanks. |
| 05:45.01 | rue_mohr | :) |
| 05:45.29 | ScribbleJ | I try to keep it simple at home. One server, in a 'vcr' form factor, stacked with the stereo, does mythtv, asterisk, etc. |
| 05:45.59 | rue_mohr | iirc 11 machines at last count... |
| 05:46.12 | ScribbleJ | We just got a bunch of extra racks at the office though and I've been kinda' eyeing one |
| 05:46.17 | ScribbleJ | Does it cost you a fortune in power? |
| 05:46.21 | rue_mohr | between firewalls, servers, and workstations |
| 05:46.36 | rue_mohr | only 2 monitors among them |
| 05:47.03 | rue_mohr | I claim they dont use up too much power |
| 05:47.14 | rue_mohr | it all offsets heating bills, right? |
| 05:47.21 | ScribbleJ | Hah |
| 05:47.59 | ScribbleJ | Arg |
| 05:48.04 | ScribbleJ | My C is so bad it makes me want to cry. |
| 05:48.10 | ScribbleJ | Perl has rotted my brains. |
| 05:48.21 | rue_mohr | i'm always glad to help if I can |
| 05:48.25 | *** join/#asterisk farkus_ (i=chatzill@cpe-98-14-94-76.nyc.res.rr.com) |
| 05:48.32 | rue_mohr | pastebin? |
| 05:48.50 | rue_mohr | (rue_mohr && C) == true |
| 05:48.53 | ScribbleJ | rue_mohr, the only real problem I keep haivng is stupid simple C stuff... |
| 05:49.05 | rue_mohr | like? |
| 05:49.25 | ScribbleJ | rue_mohr, nothing worth pastebinning... maybe you haev a link to a good something to read - the real problem I run into over and over is in wanting to return a string to a caller. |
| 05:49.56 | rue_mohr | k, thats not bad |
| 05:50.08 | ScribbleJ | Which I know, this is an easy one, but if I alloc memory for the string, then return the pointer, then who releases it? What if the caller doesn't care about ym return vaule and the memory is lost forever? |
| 05:50.18 | Pryon | just don't return the address of an object with automatic storage |
| 05:50.19 | rue_mohr | a) have the caller provide you a char * you can manipulate |
| 05:50.30 | rue_mohr | b) have the fn return a char* that was malloced |
| 05:51.12 | rue_mohr | void saymyname(char ** foo) |
| 05:51.41 | rue_mohr | char * foo; saymyname( &foo); |
| 05:52.03 | rue_mohr | or if its not allocing the memory |
| 05:52.14 | ScribbleJ | That's a, then, I get it. But B is the situation I'm complaining about right? |
| 05:52.25 | ScribbleJ | Oh, I see. |
| 05:52.26 | rue_mohr | char foo[1024]; saymyname( &foo, sizeof (foo)); |
| 05:52.36 | ScribbleJ | You're just saying always do it this way when I need to. Hrm. |
| 05:52.45 | ScribbleJ | Yeah... |
| 05:53.10 | ScribbleJ | I guess I'm just spoiled lazy by Perl. |
| 05:53.14 | rue_mohr | :) |
| 05:54.05 | rue_mohr | or make a wrapper object that has initialization, processing, and destruction calls |
| 05:54.16 | Pryon | If you're doing a lot of string maniplation C's probably not a good choice anyway |
| 05:54.17 | ScribbleJ | Oh yeah... I should just use C++. :P |
| 05:54.26 | *** join/#asterisk thansen (n=thansen@c-67-171-115-155.hsd1.ut.comcast.net) |
| 05:54.55 | rue_mohr | myobj_t foo; initthing(&foo); dostuffto(&foo); destroymy(&foo); |
| 05:55.03 | rue_mohr | na c++ sucks |
| 05:55.03 | ScribbleJ | Nah, Pyron, it's actually just the one thing I guess I didn't know how to handle in a 'clever' way and I suppose there isn't a 'clever' way then. I'm actually working on a plugin to the Asterisk Generic Speech API. |
| 05:55.28 | rue_mohr | out or in? |
| 05:55.37 | Pryon | Ah. I think the key to your sort of problem, then, is to decide who does the allocation, who does the freeing, and just be consistent. |
| 05:55.38 | rue_mohr | tts -> sst? |
| 05:55.51 | ScribbleJ | Pryon, I agree. :) Bother! |
| 05:55.57 | rue_mohr | you want to be consistant |
| 05:55.59 | Pryon | Well, that's C for you. |
| 05:56.18 | Pryon | I love C, but it doesn't reciprocate. |
| 05:56.28 | ScribbleJ | rue_mohr, the Generic Speech API is what LumenVox plugs into; I'm basically working on a drop-in replacemnent for it with all Free parts. I don't expect it to be practical. |
| 05:56.33 | ScribbleJ | But it does already work at this point. |
| 05:56.55 | GameGamer43 | ScribbleJ: nice |
| 05:57.21 | ScribbleJ | I'll probably put the code up on a site soon in it's present state considering how little time I have to work on it. |
| 05:57.58 | rue_mohr | this is the speach rekg that needs to understand "HUMAN!!!" at every level of screeming shouting tone and stress level? |
| 05:58.08 | ScribbleJ | HAhahahahah |
| 05:58.10 | ScribbleJ | Yes, rue |
| 05:58.12 | ScribbleJ | You nailed it |
| 05:58.13 | ScribbleJ | haha |
| 05:58.18 | rue_mohr | hmm |
| 05:58.44 | ScribbleJ | Youknow, that's basically true - I used to call comcast, get the voice tree, and politely say "fuck you" - and would promptly get transferred to a human. I liked that, don't think it works anymore. |
| 05:58.57 | rue_mohr | hah |
| 05:59.50 | rue_mohr | by no means let me bilittle your work, that sounds great and I'm SURE there is somewhere its really good for |
| 05:59.55 | ScribbleJ | No no, |
| 06:00.12 | ScribbleJ | I fully expect this project to be good only for experimentation. |
| 06:00.28 | rue_mohr | and I'm sure that if managers saw that asterisk could do it, they would totally want to get it installed |
| 06:00.53 | ScribbleJ | My coworkers are all excited to try it out; we use Lumenvox at the office for some products already and they have been wanting something they can plug in to release without making people pay for licenses - |
| 06:01.18 | ScribbleJ | I don't actually think this will fulfill that role; beyond letting them physically do it, play with it enough to realize lumenvox is wortht he price and buy it... heh |
| 06:01.30 | rue_mohr | hah |
| 06:01.50 | rue_mohr | why am I sure someone said that once about asterisk echo cancelers |
| 06:01.54 | ScribbleJ | Although, there's one other thing I wanted to try... taking call logs and parsing them through a continuous speech recognizer and making autoamtic transcripts. |
| 06:02.09 | GameGamer43 | ScribbleJ: but you will get those people who try it and decide they don't want to spend the money on lumnevox licenses |
| 06:02.15 | ScribbleJ | I'm sure it'll only get like every third work but even at that I could see it being useful. |
| 06:02.28 | ScribbleJ | GameGamer43, basically why I wrote it, I'm a cheap fuck. |
| 06:02.29 | GameGamer43 | just like all the people who use asterisk with x100p cards trying to spends little to nothing on it and make money |
| 06:02.36 | rue_mohr | hah |
| 06:02.49 | rue_mohr | x100p eh? |
| 06:02.56 | rue_mohr | hmmmm |
| 06:03.05 | rue_mohr | aren't those echo city? |
| 06:03.48 | GameGamer43 | ScribbleJ: understandable, but as you stated, u expect it to be useful for experimentation |
| 06:04.08 | rue_mohr | see, I dont get this, if the reason there is no echo on my channelbank is cause of tdm, why aren't the digium cards running tdm internal? |
| 06:05.03 | rue_mohr | somehting dosn't lign up |
| 06:05.15 | rue_mohr | when did I start speeling line like that |
| 06:05.19 | rue_mohr | ugh |
| 06:05.20 | ScribbleJ | I should actually go into the office someday and get the other guys to show me our new Asterisk implementations. Apparently we paid digium to put together an IVR for us, which must be nice. |
| 06:05.43 | ScribbleJ | But all I have seen is what I've been playing with at home, which is basically just my upstream SIP provider and this speech stuff. |
| 06:05.49 | rue_mohr | ScribbleJ, do your company sell * systems? |
| 06:06.12 | ScribbleJ | rue_mohr, no, we actually do payment processing - credit cardy things, the IVR is for making payments. |
| 06:06.17 | rue_mohr | ah |
| 06:06.34 | ScribbleJ | But |
| 06:06.56 | rue_mohr | have you been though the ivr? digium wrote the do/dont book on ivr didnt' they? |
| 06:06.58 | ScribbleJ | I've been there since almost the beginning, and we've had this horrible Cisco phone system for six years now... |
| 06:07.06 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 06:07.09 | rue_mohr | OOoo.. |
| 06:07.18 | ScribbleJ | And we just replaced that with Asterisk in the office, too, so tht's swell! |
| 06:07.33 | ScribbleJ | Finally I might get the /office/ to SIP me at home... brilliant. None of this SKINNY crud. |
| 06:08.04 | ScribbleJ | rue_mohr, I /haven't/ - I've been uninvolved in the project so far, but I'm excited to get time to ask the kid who's been on it to show it all to me. |
| 06:08.08 | rue_mohr | I know a big office that just got a voip nortel system, they dont like it, wish it'd been cisco, so whats the downsides with the cisco? |
| 06:08.12 | ScribbleJ | rue_mohr, like you said, I'd expect Digium to do it right. |
| 06:08.37 | rue_mohr | mm |
| 06:08.50 | ScribbleJ | rue_mohr, mainly price, licensing. TBH Cisco Unity / Cisco Callmanager are probably /fine/ systems, but I wouldn't know it because we could never afford to keep current or buy modules for interesting functionality. |
| 06:09.10 | rue_mohr | huh |
| 06:09.15 | ScribbleJ | So we were stuck with an outdated unmaintainable thing with severe limitations as to how many lines it would handle and what data it could provide... |
| 06:09.34 | ScribbleJ | Whereas Asterisk dropped on the same hardware gives us the whole freaking world. |
| 06:10.23 | ScribbleJ | I know it seems weird that we'll pay hundreds for the fancy Cisco phones, and not the money for the servers - |
| 06:10.54 | ScribbleJ | But you know how it is... the phones sit on the managers desks, they are real. The server is a nearly imaginary thing that /might/ have a physical embodiment in some dingy closet somewhere.... |
| 06:10.58 | rue_mohr | see, I work for the phone/data division of an electrical company, I WANT to be able to provide asterisk systems, the keyd systems I provide right now suck, but I dont know what other voip systems are like, I know of nortel, cisco, and mitel |
| 06:11.34 | [TK]D-Fender | rue_mohr: Nortel? LOL |
| 06:11.37 | rue_mohr | so far the price/feature for asterisk beats everything |
| 06:11.52 | [TK]D-Fender | rue_mohr: BCS = garbage... picture a web interface to DR5 |
| 06:12.01 | ScribbleJ | rue_mohr, it's inifitely good if you list it as feature/price. |
| 06:12.06 | rue_mohr | well, the nortel keyd systems cost less than the panasonic ones |
| 06:12.13 | [TK]D-Fender | rue_mohr: No.. not more feaures.. same shit, jsut not configured via the phone display |
| 06:12.48 | rue_mohr | no, a panasonic tda30 costs about $1400 to add voicemail to |
| 06:13.21 | rue_mohr | I alsmost refuse to sell nortel 616s anymore |
| 06:13.50 | rue_mohr | see where I am? why I want to get into providing asterisk? |
| 06:14.37 | ScribbleJ | Yeah - you're basically just on the other end of the same problem we had. |
| 06:14.42 | GameGamer43 | rue_mohr: nortel still won't be around in another 5 years |
| 06:14.46 | rue_mohr | there are so many types of phones avilable, and their all optional, customers can not buy sets and use their workstations if they like |
| 06:15.27 | rue_mohr | I have to remmeber to demo that at the office |
| 06:18.14 | *** join/#asterisk sah-work (n=Bawbatos@adsl-75-63-18-243.dsl.pltn13.sbcglobal.net) |
| 06:18.43 | Gopher_77 | so I guess dialing an invalid number comes back as a congested or busy line? |
| 06:18.56 | Gopher_77 | on sip that is |
| 06:19.11 | [TK]D-Fender | Gopher_77: Depends |
| 06:19.40 | *** join/#asterisk fiddur (i=fiddur@c042.rit.se) |
| 06:19.52 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 06:19.52 | *** mode/#asterisk [+o denon] by ChanServ |
| 06:20.26 | Gopher_77 | [TK]D-Fender: well, I'm getting this busy/congested signal from voipuser whenever I try to use voipuser |
| 06:20.29 | ScribbleJ | Does anyone have any more complicated examples of how to use the Speech API in the dialplan than the one int he readme? |
| 06:20.54 | ScribbleJ | And the pizza one, I've seen that! |
| 06:21.30 | [TK]D-Fender | Gopher_77: "signal".... lot of things are "signals" |
| 06:22.03 | [TK]D-Fender | Gopher_77: This last one was "annoyed' :) |
| 06:24.14 | rue_mohr | Gopher_77, so you cant dial anything without it just being annoyed? |
| 06:25.57 | Gopher_77 | rue_mohr: nothing routed to voipuser |
| 06:26.16 | Gopher_77 | rue_mohr: I see something about a trunk configuration? How do I do this? |
| 06:26.28 | rue_mohr | if I may clearify your terminology... |
| 06:26.44 | rue_mohr | no phone calls can get to voipuser? |
| 06:26.48 | rue_mohr | or from? |
| 06:27.23 | Gopher_77 | outgoing from asterisk through voipuser to a number |
| 06:27.25 | rue_mohr | trunks have to do with zaptel or dahdi channsls |
| 06:27.38 | Gopher_77 | oh, ok, so not a trunk |
| 06:28.05 | rue_mohr | I'm still trying to work out what your talking baout, through voipuser, is the hangup |
| 06:28.17 | Gopher_77 | rue_mohr: my SIP provider |
| 06:28.26 | rue_mohr | ah |
| 06:28.51 | rue_mohr | ok so phone->asterisk->external sip provider -> pots network |
| 06:29.15 | rue_mohr | right? |
| 06:29.43 | Gopher_77 | right |
| 06:30.20 | rue_mohr | ok, now, you cant get your sip provider to give you anything other than a 'it didn't work' tone of some sort |
| 06:30.49 | rue_mohr | (warning I might fall asleep any second) |
| 06:30.57 | rue_mohr | unless you answer quick |
| 06:31.29 | rue_mohr | quick is a reply in less then 4 seconds |
| 06:31.45 | Gopher_77 | rue_mohr: it fails the dial, CLI returns a message of Busy/Congested, and it goes to the next priority |
| 06:32.17 | rue_mohr | ok, so, what codec is your sip provider trying to use with you |
| 06:32.23 | [TK]D-Fender | Gopher_77: CLI warning means NOTHING |
| 06:32.26 | Gopher_77 | rue_mohr: I have no idea |
| 06:32.33 | [TK]D-Fender | Gopher_77: Look at the SIP debug respons |
| 06:32.39 | rue_mohr | well make sure its not gsm729 |
| 06:32.56 | Gopher_77 | [TK]D-Fender: what am I looking for in the SIP debug? |
| 06:33.08 | rue_mohr | error |
| 06:33.10 | rue_mohr | :) |
| 06:33.12 | [TK]D-Fender | Gopher_77: the answer it comes back with on the INVITE |
| 06:33.20 | [TK]D-Fender | which... i wil not be here to see.. |
| 06:33.24 | [TK]D-Fender | bed calls... |
| 06:33.36 | rue_mohr | he does sleep... |
| 06:33.46 | rue_mohr | I'm already in bed |
| 06:34.43 | rue_mohr | Gopher_77, make sure they aren't using gsm729, if they are, asterisk prolly has fialusre messages to do with not being able to transcode |
| 06:34.46 | Gopher_77 | voipuser has a section on what protocol is used, but unfortunately, it doesn't tell which one |
| 06:35.31 | rue_mohr | look in the console of asterisk |
| 06:36.04 | rue_mohr | you will have to be ready to copy /paste the terminal data, lots of stuff scrolls by and its impossable to try to read it live |
| 06:36.18 | Gopher_77 | rue_mohr: what's the command? |
| 06:36.33 | rue_mohr | well, from the command line asterisk -r |
| 06:36.53 | rue_mohr | in the console you can dial up all the messages with |
| 06:36.57 | rue_mohr | core set verbose 10 |
| 06:36.58 | rue_mohr | and |
| 06:36.59 | Gopher_77 | rue_mohr: sip show registry shows that it's registered, but not which protocol |
| 06:37.03 | rue_mohr | core set debug 10 |
| 06:37.18 | rue_mohr | the protocol dosn't happen till the connection is made |
| 06:37.28 | Gopher_77 | done |
| 06:37.40 | Gopher_77 | I see |
| 06:37.57 | rue_mohr | they go back and forth over what is available, asterisk wil accept gsm729, then choke cause it cant do anything with it |
| 06:38.25 | rue_mohr | to do debug for a particular sip thing, sip set debug |
| 06:38.45 | rue_mohr | you will need to do the help, there is a way of specifying the ip of the end to debug |
| 06:39.09 | rue_mohr | your almost at the end of what I can help you with |
| 06:39.25 | rue_mohr | I can tell you to look for errors, and try to find out why they happned |
| 06:39.37 | rue_mohr | 'failure' or 'error' |
| 06:39.45 | rue_mohr | watch when you try to dial |
| 06:40.01 | Gopher_77 | I have a bunch of stuff from sip debug |
| 06:40.46 | Gopher_77 | SIP/2.0 404 Not Found |
| 06:40.51 | rue_mohr | ah |
| 06:41.06 | rue_mohr | sounds like your hot on the trail |
| 06:41.33 | Gopher_77 | feels cold to me :) |
| 06:41.48 | rue_mohr | no, numbers are called on as urls |
| 06:41.50 | Gopher_77 | hmmm... SIP/2.0 401 Unauthorized |
| 06:41.55 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.199) |
| 06:41.57 | rue_mohr | ah |
| 06:42.24 | Gopher_77 | I guess I should pastebin? |
| 06:42.41 | Gopher_77 | http://nopaste.com/p/ad0WkNngF |
| 06:42.46 | rue_mohr | I doubt I can help ya, but sure |
| 06:42.48 | rue_mohr | Qwell, ? |
| 06:43.23 | *** join/#asterisk sandorp (n=sandor@wsip-98-172-95-66.ph.ph.cox.net) |
| 06:43.53 | sandorp | I recently had to reboot my asterisk server (had been up for 300+ days) |
| 06:44.16 | rue_mohr | Gopher_77, you need to include more I think |
| 06:44.21 | sandorp | now remote users are unable to connect using SIP phones (x-lite, actually) |
| 06:44.26 | rue_mohr | I'm not good at reading these |
| 06:44.46 | sandorp | I get "Network unreachable" error in the CLI whenever they connect |
| 06:44.46 | rue_mohr | sandorp, check your sip.conf ? |
| 06:44.49 | Pryon | sandorp: have they tried de and re-registering? |
| 06:44.53 | Pryon | oh |
| 06:44.59 | Gopher_77 | more > http://nopaste.com/p/aage4ihUdb |
| 06:45.24 | rue_mohr | sandorp, is the network connectin on your machine working? |
| 06:46.14 | sandorp | yes, I can connect from a local SIP phone |
| 06:46.44 | sandorp | my * machine has a public IP; my working SIP phone is behind a corp firewall |
| 06:47.32 | rue_mohr | Qwell, ? |
| 06:48.07 | rue_mohr | sandorp, can your cients get to port 5060? |
| 06:48.30 | sandorp | http://nopaste.com/p/aa8s82viv |
| 06:48.38 | rue_mohr | network unreachable is a big network fualt though |
| 06:48.49 | sandorp | yes, they are opening a connection to the * machine |
| 06:48.54 | rue_mohr | means it cant get to the dest network |
| 06:48.57 | rue_mohr | no route |
| 06:49.16 | Gopher_77 | default gateway? |
| 06:49.17 | sandorp | so I did a traceroute and I get as far as their provider's network |
| 06:49.36 | sandorp | I believe the provider is blocking ICMP to their home user's DSL |
| 06:49.59 | rue_mohr | k, they get network unreachable? |
| 06:50.05 | sandorp | hmm, didn't check the default gw |
| 06:50.15 | sandorp | I figured it was working if I can reach it |
| 06:50.22 | rue_mohr | its not trying to return to your local ip is it? |
| 06:50.41 | rue_mohr | always test from outside |
| 06:50.41 | Gopher_77 | yeah, gw should be good if you can reach it |
| 06:51.02 | Gopher_77 | but doesn't the "network unreachable" mean that you can't reach it? |
| 06:51.08 | rue_mohr | nothing like a machine on the internet trying to reach 192.168.... |
| 06:51.15 | sandorp | when I run "route" I don't see an entry for "default" |
| 06:51.24 | rue_mohr | it means there is no route to the network its trying to get to |
| 06:51.28 | rue_mohr | from where it is |
| 06:51.42 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 06:52.17 | *** join/#asterisk TrentCreek (n=kvirc@adsl-75-14-6-143.dsl.hrlntx.sbcglobal.net) |
| 06:52.41 | Gopher_77 | gw problem would also mean he can't get to any ip outside his lan |
| 06:52.44 | rue_mohr | dont forget that resolv.conf and routing data get overwritten for machines using dhcp |
| 06:53.09 | Gopher_77 | that's with network manager in linux, right? Is he using linux? |
| 06:53.40 | rue_mohr | if the problem only came up when you rebooted the * server, then the problem is there |
| 06:53.40 | sandorp | yes, using linux |
| 06:53.53 | rue_mohr | how did the cat get back to the middle of the bed again? |
| 06:53.54 | sandorp | and I had no default route after reboot |
| 06:54.04 | Gopher_77 | not surprising |
| 06:54.08 | rue_mohr | that wouldn;t help |
| 06:54.10 | Gopher_77 | have to put it in the network-script |
| 06:54.15 | sandorp | just added it and having user try it again |
| 06:54.21 | sandorp | woo hoo |
| 06:54.24 | sandorp | that was it |
| 06:54.26 | rue_mohr | where does its network config come form? |
| 06:54.32 | sandorp | <- feels stupid now |
| 06:54.37 | rue_mohr | it happens |
| 06:54.51 | rue_mohr | its good to reboot a machine once in a while to make sure its configsafe |
| 06:54.51 | Gopher_77 | try /etc/sysconfig/network-scripts |
| 06:54.55 | sandorp | form /etc/sysconfig/network? |
| 06:55.07 | rue_mohr | ok, its static then? |
| 06:55.13 | rue_mohr | not dhcp? |
| 06:55.18 | sandorp | gateway address is set |
| 06:55.23 | sandorp | yeah, static |
| 06:55.25 | Gopher_77 | not network, network-scripts |
| 06:55.42 | Gopher_77 | ifcfg-eth0, for example |
| 06:55.45 | rue_mohr | on debian its /etc/networking/interfaces |
| 06:55.59 | sandorp | looking at ifcfg-eth0 |
| 06:56.00 | rue_mohr | on redhurt its different |
| 06:56.04 | sandorp | gateway is missing |
| 06:56.11 | Gopher_77 | yes gateway= |
| 06:56.35 | Gopher_77 | I happen to use redhurt flavors |
| 06:56.43 | rue_mohr | ah |
| 06:57.12 | sandorp | ok, so next reboot should be ok ... it had a route to the local corp firewall, so I guess that's why I was able to connect |
| 06:57.30 | sandorp | thanks for the nudge in the right direction |
| 06:57.34 | Gopher_77 | np |
| 06:57.43 | sandorp | g'night all |
| 06:57.48 | Gopher_77 | goodnight |
| 06:59.52 | *** join/#asterisk sofh (n=sofh@119.153.55.91) |
| 07:00.03 | sofh | hi all |
| 07:01.01 | sofh | suppose i've some analogue card installed on my asterisk box and a PSTN line is connected with that card |
| 07:01.33 | sofh | can i dial out to some "calling card" to place my international calls ? |
| 07:01.50 | sofh | in other words , may i use any calling card as "my termination" gw ? |
| 07:02.13 | TrentCreek | why not just pick up a phone? |
| 07:02.48 | TrentCreek | yes, you could program the system to dial for you and enter the code and number for you |
| 07:02.51 | sofh | what for my users behind the pbx ? they are on lan and i don't want to give all of them direct connectivity to PSTN line |
| 07:03.00 | Gopher_77 | should be able to dial a number through the PSTN line through an extension, right? |
| 07:03.27 | TrentCreek | yes, or have the box do it |
| 07:03.43 | sofh | i am getting an idea , suppose i want all number starting from 345 will dial the calling card first but how to keep the number user dialed |
| 07:03.53 | Gopher_77 | here's a question: is there a way to put a pause in dialing for the phone card provider? |
| 07:04.05 | Gopher_77 | ${EXTEN} |
| 07:04.06 | sofh | i mean user is on ip phone , he picks the phone and dials 00441xxx |
| 07:04.06 | TrentCreek | yes, you can do what you want |
| 07:04.35 | sofh | what i am confused is to where to store his dialed number , till asterisk dials the calling card access number and then its pin number |
| 07:05.33 | TrentCreek | commands start when the user picks up the phone |
| 07:05.55 | *** join/#asterisk bgmarete (n=marebri_@196.201.208.129) |
| 07:06.50 | sofh | i think it could be done |
| 07:06.55 | TrentCreek | no |
| 07:06.56 | sofh | but question is HOW :-S |
| 07:07.00 | *** join/#asterisk MaliutaLap (n=biteme@kiev.lusan.id.au) |
| 07:07.03 | TrentCreek | KNOW it can be done |
| 07:07.18 | GameGamer43 | ScribbleJ: you still around? |
| 07:07.41 | sofh | TrentCreek, i didn't get your "no" or "Know" |
| 07:07.57 | TrentCreek | just program it in the extension..i already told you commands start when the user picks up the phone...you could even have it play a voice message of "FUCK YOU" when they pick up their phone |
| 07:08.21 | TrentCreek | dont THINK it can be done...KNOW it can be done |
| 07:08.35 | Gopher_77 | maybe something like exten => _00411.,n,Dial(SIP/user/<number-for-card>${EXTEN:5}) |
| 07:08.35 | sofh | now i got you |
| 07:08.59 | TrentCreek | yes |
| 07:09.09 | TrentCreek | then next line they dial the number to call |
| 07:09.30 | TrentCreek | well the PIN number then number to call |
| 07:09.55 | Gopher_77 | or would that all go in <number-for-card>? |
| 07:10.23 | TrentCreek | you should put each event as seperate commands |
| 07:10.31 | TrentCreek | on seperate lines I mean |
| 07:10.42 | Gopher_77 | or just take out ${EXTEN} altogether to transfer control from * to the card provider |
| 07:10.44 | sofh | correct! |
| 07:11.11 | ScribbleJ | GameGamer43, ? |
| 07:11.20 | TrentCreek | Dial (Card Access Number) |
| 07:11.24 | TrentCreek | Dial (PIn) |
| 07:11.35 | TrentCreek | Dial (person you are calling) |
| 07:11.50 | GameGamer43 | ScribbleJ: let me know when you post that code, it'll be interesting to see what you got and play around with it |
| 07:12.00 | ScribbleJ | Sure, no problem! |
| 07:12.02 | Gopher_77 | so the third line would contain the ${EXTEN} |
| 07:12.25 | Gopher_77 | some version of it |
| 07:12.51 | sofh | ok everybody thanks for your tips..let me test it i will be back if i face something new :) |
| 07:19.41 | *** join/#asterisk shiltron (i=shiltron@206.41.112.113) |
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| 08:27.51 | *** join/#asterisk contrabanda (n=contr@92.241.69.134) |
| 08:27.55 | contrabanda | Hello |
| 08:28.40 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 08:28.41 | Gopher_77 | ~Hello |
| 08:28.42 | jbot | Howdy Bub |
| 08:29.43 | contrabanda | i need help with extensions. i receive error when calls come from PSTN - E1 ---> http://pastebin.ca/1326977 |
| 08:29.53 | contrabanda | please help me to fix problem |
| 08:31.28 | Gopher_77 | no such device as 1? |
| 08:32.08 | contrabanda | what do u mean |
| 08:32.08 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 08:32.10 | contrabanda | ? |
| 08:32.51 | Gopher_77 | dahdi show channel 1 |
| 08:34.04 | Gopher_77 | or: dahdi show channels |
| 08:34.50 | contrabanda | ok |
| 08:34.57 | contrabanda | ill paste output |
| 08:35.05 | contrabanda | http://pastebin.ca/1326981 |
| 08:35.23 | contrabanda | i have pasted my extensions.conf and output of dahdi show command |
| 08:38.31 | Gopher_77 | paste chan_dahdi.conf? |
| 08:40.15 | contrabanda | http://pastebin.ca/1326985 |
| 08:40.29 | contrabanda | here is config |
| 08:42.05 | *** join/#asterisk ScriptFanix (i=vincent@2a01:e35:2f43:ae90:21a:70ff:fea3:44ab) |
| 08:42.15 | ScriptFanix | hi |
| 08:42.40 | contrabanda | hi |
| 08:42.53 | Gopher_77 | contrabanda: so it appears you should assign a context to these channels in chan_dahdi.conf and define this context in extensions.conf |
| 08:43.27 | Gopher_77 | contrabanda: this way when a call is received on these channels they have instructions to follow as defined in the context |
| 08:43.29 | ScriptFanix | what are the best sip codecs for inband DTMF ? currently i allowed ulaw, alaw and speex, but asterisk doesn't receive DTMFs from my GSM |
| 08:43.59 | ScriptFanix | i have dtmfmode=auto |
| 08:45.48 | contrabanda | Gopher_77: should i change context=default to context=egrisigroup, which i have defined in extensions.conf? |
| 08:46.00 | *** join/#asterisk Chaz6 (n=chaz@chaz6.com) |
| 08:46.45 | Chaz6 | Hi there, does anyone know if the problem that SIP URIs containing IPv6 literal addresses violate rfc 3986, and are thus not valid absolute URIs, been fixed? |
| 08:47.21 | Gopher_77 | contrabanda: that would tell those channels to follow the instructions for context egrisigroup. If that is what you want, do that. |
| 08:48.21 | contrabanda | thanks |
| 08:48.37 | Chaz6 | For example, <sip:callee@[2001::1]> is a valid SIP URI but not a valid URI |
| 08:48.46 | Gopher_77 | contrabanda: I think I see more problems with extensions.conf though |
| 08:49.23 | Gopher_77 | Chaz6: sorry, I know nothing about IPv6 or the rfc's |
| 08:51.42 | contrabanda | Gopher_77: what problems? |
| 08:52.05 | Gopher_77 | contrabanda: for one, I don't see why you have g1 in these places |
| 08:53.11 | contrabanda | Gopher_77: is it mistake? |
| 08:54.01 | Gopher_77 | contrabanda: I think so; I don't know all the rules but I know that this place is normally used for a number representing the dahdi channel |
| 08:54.39 | Gopher_77 | contrabanda: also after this place, it is normally a comma before the ${EXTEN} |
| 08:55.26 | Gopher_77 | contrabanda: I'm sorry I think I may be wrong about the second part |
| 08:55.37 | Gopher_77 | contrabanda: I haven't used it in my configuration |
| 08:56.30 | Gopher_77 | contrabanda: yes, I believe the ${EXTEN} part of your configuration is correct |
| 08:56.41 | contrabanda | Gopher_77: I have one more problem , when im calling to PSTN throuugh DAHDI E1, A number is not displayed on the other side |
| 08:57.01 | contrabanda | where can i fix this? |
| 08:59.03 | *** join/#asterisk af_ (n=getsmart@88-149-230-108.dynamic.ngi.it) |
| 08:59.18 | Gopher_77 | contrabanda: I believe that is in the callerid directive in chan_dahdi.conf |
| 08:59.31 | Gopher_77 | contrabanda: can you now call from one line to the next line? |
| 09:02.24 | contrabanda | Gopher_77: now when i am calling from pstn i got such error: Spawn extension (egrisitrunk, 245263, 1) exited non-zero on 'SIP/499994-093bc3a8' |
| 09:02.24 | contrabanda | <PROTECTED> |
| 09:02.52 | Gopher_77 | contrabanda: I'm not familiar with the pri_cpe signalling, so forgive me if I have extra questions |
| 09:03.02 | contrabanda | Gopher_77: but in extensions.conf i have such record exten => _499XXX,1,Dial(SIP/${EXTEN}) |
| 09:03.11 | Gopher_77 | contrabanda: which dahdi channels will have telephones on them? |
| 09:03.32 | Gopher_77 | contrabanda: ${EXTEN} will return all 6 digits dialed |
| 09:03.46 | Gopher_77 | contrabanda: to reduce that to the last three, use ${EXTEN:3} |
| 09:04.00 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 09:04.00 | Gopher_77 | contrabanda: it strips the first three with this number 3 |
| 09:04.28 | Gopher_77 | contrabanda: so it will attempt to use dahdi channel 994 |
| 09:04.35 | *** join/#asterisk tamiel (n=tamiel@213.30.183.226) |
| 09:04.51 | Gopher_77 | contrabanda: sorry, I'm getting confused |
| 09:05.01 | Gopher_77 | contrabanda: this is SIP |
| 09:05.45 | contrabanda | Gopher_77: yes i have SIP clients asigned with 6 digit numbers 499XXX. they can call each other, also can call to PSTN through DAHDI |
| 09:05.52 | Gopher_77 | contrabanda: with this exten line, it will dial the 6 digit extension to SIP, but the SIP peer is not defined |
| 09:06.03 | contrabanda | but incomming calls through dahdi are not working |
| 09:06.26 | Gopher_77 | contrabanda: so this is a successful configuration for SIP telephones? |
| 09:06.36 | *** join/#asterisk bgmarete (n=marebri_@196.201.208.129) |
| 09:06.44 | contrabanda | no i have defined |
| 09:06.46 | contrabanda | [499994] |
| 09:06.46 | contrabanda | type=friend |
| 09:06.46 | contrabanda | callerid="SPQR"<499994> |
| 09:06.46 | contrabanda | username=499994 |
| 09:06.46 | contrabanda | host=dynamic |
| 09:06.46 | contrabanda | secret=samagon |
| 09:06.48 | contrabanda | nat=yes |
| 09:06.50 | contrabanda | context=egrisitrunk |
| 09:06.52 | contrabanda | canretrive=no |
| 09:06.54 | contrabanda | allow=all |
| 09:07.04 | Gopher_77 | contrabanda: oh, so 499994 is the name of the SIP device |
| 09:07.52 | Gopher_77 | contrabanda: what extensions do you want to use for the dahdi channels? |
| 09:08.14 | *** join/#asterisk LuisTorres (n=chatzill@a213-22-94-113.cpe.netcabo.pt) |
| 09:08.38 | Gopher_77 | contrabanda: first, are the dahdi channels for telephones or outgoing lines? |
| 09:08.50 | contrabanda | i have defined all extensions in content egrisigroup |
| 09:09.24 | contrabanda | dahdi chanels are for incoming and outgoing through E1 interface |
| 09:09.55 | Gopher_77 | contrabanda: ok so they are to the telecommunications company |
| 09:10.35 | Gopher_77 | contrabanda: when calls come in from outside they will enter the egrisigroup context, unless you wish to use another |
| 09:11.45 | Gopher_77 | contrabanda: unfortunately, I haven't configured this part of my system, but I would think that you Dial(dahdi/<channel>/<number>) to use these for outgoing calls |
| 09:11.54 | *** join/#asterisk ludan (n=daniele@unaffiliated/ludan) |
| 09:12.05 | Gopher_77 | contrabanda: I'm not sure how to use them in a pool |
| 09:12.24 | contrabanda | Gopher_77: yes, exactly when i call from other telephone network, call is routed to context=egrisigroup |
| 09:12.30 | ludan | is there an alternative to iax2.fwdnet.net? |
| 09:12.35 | ludan | (hi) |
| 09:12.43 | contrabanda | Extension '499994' in context 'egrisigroup' from '2245263' does not exist. Rejecting call on channel 0/1, span 1 |
| 09:13.10 | contrabanda | it says that there is not 499994 in this context |
| 09:13.13 | Gopher_77 | contrabanda: what is '2245263'? |
| 09:13.35 | contrabanda | its a number of telephone in telecom company |
| 09:13.49 | Gopher_77 | contrabanda: ok |
| 09:15.18 | Gopher_77 | contrabanda: try before the _499XXX line, exten => _499XXX,1,Answer |
| 09:15.36 | Gopher_77 | contrabanda: and change the existing _499XXX,1 to _499XXX,n |
| 09:16.04 | Gopher_77 | contrabanda: sorry, s/Answer/Answer()/ |
| 09:18.01 | *** join/#asterisk Kisu (n=k@daniel1117.broker.freenet6.net) |
| 09:21.06 | kaldemar | Gopher_77: that won't help anything |
| 09:21.40 | kaldemar | contrabanda: pastebin the context in extensions.conf |
| 09:22.23 | Gopher_77 | contrabanda: he probably knows more than I do |
| 09:22.42 | contrabanda | Gopher_77: exten => _499XXX,1,Answer() |
| 09:23.37 | kaldemar | ~pb |
| 09:23.38 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 09:23.41 | Gopher_77 | contrabanda: then after that exten => _499XXX,n,Dial(SIP/${EXTEN}) |
| 09:24.33 | kaldemar | having an answer as the first priority has nothing to do with the exten not begin found in the context. |
| 09:24.50 | contrabanda | Gopher_77: exten => _499XXX,1,Answer |
| 09:24.51 | contrabanda | exten => _499XXX,n,Dial(SIP/${EXTEN}) |
| 09:25.01 | contrabanda | i have this but the same error :( |
| 09:25.10 | *** join/#asterisk styelz (n=yoohoo@m0o0.mooo.com) |
| 09:25.40 | Gopher_77 | contrabanda: yes, I think it did nothing, now that I think about how you are successful calling from SIP to SIP |
| 09:26.22 | Gopher_77 | contrabanda: did you reload chan_dahdi after changing context for the dahdi channels? |
| 09:28.17 | contrabanda | yes |
| 09:29.43 | Gopher_77 | contrabanda: and the SIP channels are registered? |
| 09:29.55 | Gopher_77 | contrabanda: sip show registry |
| 09:30.50 | kaldemar | contrabanda: you have "egrisitrunk" in extensions.conf and "egrisigroup" in dahdi configuration. there's your problem. |
| 09:30.55 | Gopher_77 | contrabanda: sometimes it takes a minute if you reload SIP or * |
| 09:31.06 | Gopher_77 | ah, that would do it |
| 09:32.10 | contrabanda | Gopher_77: oh noooooooo |
| 09:32.19 | Gopher_77 | contrabanda: ? |
| 09:32.58 | contrabanda | Gopher_77: ill change it now |
| 09:33.51 | contrabanda | Gopher_77: Thanks a lot dude. its working now :) many many thanks |
| 09:34.03 | Gopher_77 | contrabanda: np |
| 09:34.15 | Gopher_77 | contrabanda: it turned out to be one simple thing :) |
| 09:34.50 | *** join/#asterisk Faustov (i=user@gentoo/user/faustov) |
| 09:35.40 | Gopher_77 | kaldemar: he didn't have it defined at all before, guess it was a typo on top of that |
| 09:36.02 | Faustov | hello, is hardware echo cancelation on pstn cards a demanded feature? I got a sangoma a200d card recommended but the only available is 2x cheaper a200 which comes without this feature |
| 09:36.55 | contrabanda | Gopher_77: exactly :) |
| 09:37.47 | contrabanda | Gopher_77: I have one more question please :) When user ins not online how can i transfer call to some anouncement? For example nowonline.wav? |
| 09:38.57 | *** part/#asterisk Chaz6 (n=chaz@chaz6.com) |
| 09:39.46 | kaldemar | Faustov: it is not a demanded feature, of course you can interface pstn without any echo cancellation, but if you run into echo problems, hardware cancellation is probably the best solution. software solutions tend to perform worse and cause more work. |
| 09:40.44 | *** join/#asterisk Mr_BOnD_007 (i=bond0070@119.160.199.6) |
| 09:41.19 | Faustov | kaldemar: i see, what is the likelyhood of getting into problems with echo? |
| 09:42.06 | Gopher_77 | contrabanda: after the priority with the Dial application, put in another priority to call a file-playing application (I don't know what that is) |
| 09:42.59 | contrabanda | ok thanks |
| 09:43.00 | kaldemar | Faustov: depends on the environment, but it is likely that you will get echo. i'd recommend hardware cancellation to avoid a headache. |
| 09:43.16 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 09:44.10 | dominic1 | morning, is there any function in asterisk 1.6 to get the callerid(Name) of the person I called? I think there is a function in the sip protocol to do that.... |
| 09:44.50 | Gopher_77 | contrabanda: http://www.voip-info.org/tiki-index.php?page=Convert+WAV+audio+files+for+use+in+Asterisk |
| 09:45.15 | kaldemar | contrabanda: you might also want to put a timeout for the dial command (for syntax, see core show application dial in asterisk CLI) depending on what you mean by online, and like Gopher_77 said, use app Playback to play the announcement in the next priority. |
| 09:46.18 | Faustov | kaldemar: since u're around, mind if i ask another question? This time regarding means of connecting asterisk to cellular network - what would you suggest? I've found one pci card but it costs 3k usd... |
| 09:47.05 | Gopher_77 | kaldemar: very useful; now I know how to get a list of applications |
| 09:47.26 | *** join/#asterisk bgmarete (n=marebri_@196.201.208.129) |
| 09:48.08 | kaldemar | Faustov: i've only tried junghanns gsm cards. seemed to work and be relatively easy to configure based on brief testing. |
| 09:50.23 | kaldemar | dominic1: probably not a real straight forward way, but that MIGHT be achievable with M() parameter of the Dial app and CALLERID function. |
| 09:52.06 | Faustov | kaldemar: that's the one that's so expensive! Are we talking about the same card? http://www.junghanns.net/en/GSM-PCI_produkt.html |
| 09:52.27 | kaldemar | Faustov: yes, that's the one. |
| 09:53.37 | kaldemar | you can also find some cheaper gsm gateways, but i won't give comments on those since i haven't tried any myself. :) |
| 09:53.46 | Faustov | hmmmm |
| 09:54.09 | Faustov | this is hard, hardly anyone can share experience with connecting * to gsm |
| 09:54.28 | Faustov | this card is mostly mentioned but i'm pretty sure i won't get the funding for it |
| 09:55.01 | kaldemar | Faustov: http://www.voip-info.org/wiki/view/VOIP+GSM+Gateways |
| 09:55.28 | dominic1 | kaldemar: but I think it's not possible to change the callerid while already calling to another person. The callerid is transmitted when connecting (I think). |
| 09:55.54 | *** join/#asterisk scruz (n=scruz@41.220.73.170) |
| 09:56.54 | scruz | hello |
| 09:57.05 | kaldemar | dominic1: what exactly are you trying to do? now you're talking about changing the callerid. |
| 09:58.35 | Faustov | kaldemar: yeah i've been checking those out, one thing i'm not sure about is how a dialplan would work there (as in, it would have to be connected to that pstn fxo port, then if someone made a call to another cell, i'd like to route it via this gateway - but it's external so how? |
| 09:59.05 | *** join/#asterisk bobsaccamano (n=ckd683@203.126.136.142) |
| 09:59.29 | bobsaccamano | hi..does anyone know the right place to get info on TAPI 3.0?? |
| 09:59.35 | kaldemar | Faustov: that's no problem |
| 10:00.16 | mvanbaak | bobsaccamano: google ? |
| 10:01.37 | Faustov | kaldemar: any hint? What comes to my head would be directing that traffic to the specified fxo port on he card, but i don't have one yet so i'm just guessing |
| 10:01.51 | kaldemar | bobsaccamano: this is most likely not a great place to get information on windows API's. ggi. |
| 10:02.02 | bobsaccamano | mvanbaak, okay ill state the problem: I'm connecting a POTS phone to a laptop where Im running a SIP client..now i want to initiate a call from the phone which should be converted into a SIP URI and sent as input to the client on the laptop... |
| 10:02.25 | bobsaccamano | can i use asterisk here |
| 10:04.07 | kaldemar | Faustov: if your fxo channel is busy, direct it to some place else in the dialplan. that's really basic stuff. |
| 10:04.53 | scruz | i want to connect two asterisk servers, one with a (more-or-less) dynamic IP, the other with a static IP. how can i do this? (tried it and can't get it working) |
| 10:05.43 | mvanbaak | bobsaccamano: how are you connecting the POTS phone to your laptop ? |
| 10:05.44 | kaldemar | bobsaccamano: you can do just about anything you want with asterisk. but sounds like you're trying to get the laptop to work like an ATA. no need for asterisk in that case. |
| 10:06.32 | bobsaccamano | mvanbaak, using the RJ-11 port |
| 10:06.51 | dominic1 | If I call a person, I can only see the number I dialed in the display. On other systems like Siemens it's possible to see the name of the called person in the display |
| 10:07.06 | dominic1 | kaldemar: If I call a person, I can only see the number I dialed in the display. On other systems like Siemens it's possible to see the name of the called person in the display |
| 10:07.58 | mvanbaak | bobsaccamano: it most probably is not going to work, because your laptop modem is not supported by zaptel I think. Most modems are not. |
| 10:08.28 | bobsaccamano | kaldemar, yeah..so how do i convert the analog DTMF tones into a string containing the sip uri? |
| 10:08.58 | kaldemar | dominic1: use the phone's phonebook. personally, i think that's just a waste of time. a caller should know who is being called. |
| 10:09.18 | Faustov | kaldemar: ok i guess it will be easier once i look at it, thanks for the info |
| 10:09.45 | kaldemar | bobsaccamano: why are you trying to do that? |
| 10:10.15 | bobsaccamano | kaldemar, because i want to test a custom SIP stack |
| 10:10.27 | bobsaccamano | and want to make it interoperable |
| 10:11.19 | dominic1 | kaldemar: That's the problem, sometimes when a user is dialing a internal number, he mixes the numbers up. So if he is able to see the name of the called person, he can hangup up the call before somebody is picking it up. Enterprise class systems are able to do that. |
| 10:14.37 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 10:15.41 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 10:15.44 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 10:16.32 | kaldemar | bobsaccamano: you hardly need an analog telephone for testing a SIP stack against some SIP device. |
| 10:18.13 | bobsaccamano | kaldemar, i know it sounds silly but its important from an end-user perspective |
| 10:19.01 | kaldemar | bobsaccamano: so that the end user can plug an analog telephone in their laptop and make calls? |
| 10:19.07 | bobsaccamano | yes |
| 10:19.48 | kaldemar | jesus, use a soft phone, ATA or a hard phone. |
| 10:21.27 | kaldemar | i wouldn't re-invent the wheel since all laptops don't even have modems nowadays. besides there are plenty of handsets (for w.g. USB ports) that work with soft phones in case the users wants a traditional looking device to dial with. |
| 10:22.27 | Gopher_77 | softphone with a bluetooth headset isn't bad either |
| 10:23.15 | Gopher_77 | besides, even if the laptop has a standard modem, it's not the modem you need to plug an analog phone into the laptop |
| 10:23.15 | *** part/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-6e3a7e315cef01e6) |
| 10:23.49 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 10:24.43 | kaldemar | scruz: http://www.voip-info.org/wiki-Asterisk+-+dual+servers |
| 10:24.48 | kaldemar | ~book |
| 10:24.54 | jbot | book is, like, probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 10:25.15 | kaldemar | scruz: ^^ good detailed examples found there also |
| 10:25.39 | *** join/#asterisk clintc (n=clintc@n128-227-185-136.xlate.ufl.edu) |
| 10:28.52 | Gopher_77 | ~analog |
| 10:28.53 | jbot | analog is, like, Analog refers to a representation of a quantity that varies over any continuous range of values. Analog signals can be thought of as pure in nature and not processed. Thus, the debate over whether record albums (analog representation of sound) sound better than CDs (digital representation of sound). Think of nature as analog. Values are exact, but it is impossible to correct errors in reproduction. |
| 10:30.00 | scruz | kaldemar: i've both the book and that url open as at when i posted my question. i couldn't get it to work |
| 10:30.29 | kaldemar | scruz: what is the problem? |
| 10:31.17 | scruz | i want to call a registered extension on the second server from the first |
| 10:31.30 | scruz | 'first' being the dynamic, the second being the static |
| 10:31.44 | scruz | but the call isn't handled |
| 10:33.06 | scruz | here's the dialplan from the dynamic: http://pastebin.com/d48335a24 |
| 10:33.15 | *** join/#asterisk AdvoWork (n=AdvoWork@unaffiliated/advowork) |
| 10:33.35 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 10:33.52 | AdvoWork | Hi there, trying to eliminate echo problems, and it says echocancel should be 64 in zapata.conf but ive got echocancel=yes and thats it,whats the name of the setting to do echocancel?=64? |
| 10:34.09 | kaldemar | what is the EXACT problem? show CLI output of a failed call, channel configuration files and extensions.conf |
| 10:34.27 | scruz | call just drops |
| 10:34.36 | scruz | no o/p in cli |
| 10:35.07 | kaldemar | give "set verbose 10" in cli and try again. |
| 10:35.41 | kaldemar | and look at both cli's |
| 10:36.49 | scruz | nothing, still |
| 10:37.12 | kaldemar | then the call is not even going to either asterisk. you need to configure your client right. |
| 10:38.25 | scruz | seems the client is configured right, since it shows on the cli of the server it's registered to when i started it again |
| 10:41.57 | kaldemar | if the cli says nothing upon dial, it is not. |
| 10:44.58 | *** join/#asterisk JJ2110 (n=James@222-152-203-34.jetstream.xtra.co.nz) |
| 10:46.56 | *** join/#asterisk emrahpbx (n=pbx@87.213.128.90) |
| 10:47.07 | emrahpbx | heya all |
| 10:52.40 | scruz | i changed the context for the account i'm using with the softphone (only!) and dialled the extension 99992 , which has the sayunixtime application for that extension, and it works |
| 10:54.23 | scruz | but it doesn't do anything when i changed it back to the linkin extension |
| 10:54.34 | scruz | *linkin context |
| 10:54.50 | kaldemar | i can't help you if you don't give information. |
| 10:58.45 | kaldemar | scruz: you said that already, but didn't show a cli output of a failed call on verbosity 10 nor show channel configuration files. |
| 10:59.10 | scruz | on the cli, there was no o/p for the failed call |
| 10:59.41 | kaldemar | then the call must be going some place else. |
| 11:00.05 | scruz | http://pastebin.com/d296ffa94 |
| 11:00.14 | scruz | for the dynamic host |
| 11:01.14 | kaldemar | how have you configured the client you're using? |
| 11:01.52 | kaldemar | and change SIP/asterisk_linux/3590003 to SIP/3590003@asterisk_linux |
| 11:02.15 | *** join/#asterisk ultrav1olet (n=telnet@94.180.49.133) |
| 11:03.29 | scruz | is the register statement correct? is it in the right host? |
| 11:03.43 | ultrav1olet | We have one telephony ZAP line and when it's busy the next person trying to call receives Normal Clearing message when he or she tries to call this line. How can I turn this message into sound message saying "The line is currently busy. Call again later" or somethingh like this? |
| 11:04.12 | kaldemar | AdvoWork: check the sample config, it will tell you what you can set echocancel to. |
| 11:05.50 | kaldemar | scruz: the register statement doesn't affect outgoing calls in any way. but, if you don't use secrets, it is correct. |
| 11:05.59 | ultrav1olet | and one more question: right now our asterisk doesn't detect BUSY signal on Zap channel, so when you have finished calling you will be listening to busy signals indefinitely |
| 11:06.51 | scruz | thanks |
| 11:07.10 | scruz | now i'm going to lie down since it doesn't still work :) |
| 11:07.55 | kaldemar | scruz: just do it exactly as in the book and it will work. |
| 11:10.55 | scruz | ok |
| 11:11.06 | scruz | will buzz back and give you info |
| 11:13.19 | *** join/#asterisk xnixan (n=xnixan@unaffiliated/xnixan) |
| 11:17.13 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 11:19.21 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 11:21.40 | *** join/#asterisk ingenius (n=alektro@69.90.72.173) |
| 11:24.46 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 11:27.05 | *** join/#asterisk scruz (n=scruz@41.220.73.170) |
| 11:27.10 | scruz | ack |
| 11:27.13 | scruz | hi again |
| 11:28.33 | scruz | the remote host has registered my local asterisk, but my local asterisk keeps throwing up registration timeouts :( |
| 11:29.04 | scruz | 9 attempts & counting |
| 11:29.22 | kaldemar | is there a firewall blocking port 5060 in between? |
| 11:29.44 | scruz | nope |
| 11:29.58 | kaldemar | is 5060 open in the remote machine? |
| 11:29.58 | scruz | or the softphone wouldn't work |
| 11:30.13 | scruz | yes |
| 11:30.25 | kaldemar | your softphone is not the remote asterisk |
| 11:32.54 | *** join/#asterisk viq (n=viq@unaffiliated/viq) |
| 11:36.31 | AdvoWork | kaldemar, wheres the sample config? |
| 11:39.57 | kaldemar | AdvoWork: in the source package under configs/ |
| 11:40.26 | *** join/#asterisk peaquino (n=mwegrzyn@main.litex.pl) |
| 11:40.55 | peaquino | hello! |
| 11:41.25 | peaquino | I've got a problem with dahdi_cfg (or zaptel_cfg if I try the old drivers) |
| 11:42.10 | peaquino | when I run it, it completely freezes the server |
| 11:43.01 | peaquino | I've got a Digium TE220p card, the server is and AMD Phenom 9850, with Gigabyte AMD 780 motherboard |
| 11:43.29 | peaquino | I've installed Ubuntu Server 64 bit and the latest dahdi drivers and tools |
| 11:44.25 | MaliutaLap | peaquino: you want to pay someone to get it working??? |
| 11:44.26 | peaquino | on a very similar server (the only difference is in the processor, it uses Athlon 5600) everything works fine |
| 11:44.29 | MaliutaLap | has time :) |
| 11:45.22 | peaquino | MaliutaLap: I'm thinking about it, but right now I'm doing some research first |
| 11:45.26 | peaquino | :) |
| 11:46.54 | MaliutaLap | Offer only Valid until 06:30 09/02/2009 (GMT+10:00) and after release with shiny hip |
| 11:47.17 | *** join/#asterisk joelsolanki (i=joelsola@202.160.161.94) |
| 11:47.22 | joelsolanki | Hello all |
| 11:47.45 | MaliutaLap | joelsolanki: goodbye |
| 11:47.48 | joelsolanki | i want to configure 111 extension when called then the dialing extension should listen his own extension numbe |
| 11:47.51 | MaliutaLap | <PROTECTED> |
| 11:47.52 | MaliutaLap | ;) |
| 11:47.58 | joelsolanki | :) |
| 11:48.14 | MaliutaLap | joelsolanki: you want echo? |
| 11:48.28 | MaliutaLap | as in echo what the caller is saying down the line? |
| 11:48.48 | joelsolanki | i just want to know the dialer that from where he is dialing |
| 11:49.20 | joelsolanki | means if my extension is 200 and i dial 111 then i should get message " Your extension number is 200 " |
| 11:49.29 | MaliutaLap | ahhh |
| 11:49.38 | kaldemar | the user has serious problems if he doesn't know where she's dialing from. ;) |
| 11:49.48 | kaldemar | +s |
| 11:49.52 | joelsolanki | :) |
| 11:49.56 | MaliutaLap | so you want Say($(CALLERID)) or something like that |
| 11:50.09 | MaliutaLap | that is off the top of my head and probbably doesn't work |
| 11:50.24 | *** join/#asterisk scruz (n=scruz@41.220.73.170) |
| 11:50.25 | kaldemar | SayDigits(${CALLERID(num)}) to be precise |
| 11:50.29 | joelsolanki | ok |
| 11:50.34 | MaliutaLap | is adding disclaimers to everything tonight |
| 11:50.35 | dominic1 | Is it possible that the asterisk 1.6 sends local and remote tags in the notify messages? I want to display a popup with the information of a caller which calls somebody on my blf-keys |
| 11:50.55 | MaliutaLap | kaldemar: and if the callerID id "John"??? |
| 11:51.00 | scruz | if asterisk says 'no such command sip', i assume sip support wasn't compiled in? |
| 11:51.01 | MaliutaLap | s/id/is/ |
| 11:51.22 | MaliutaLap | scruz: or chan_sip isn't loaded |
| 11:51.22 | scruz | jbot seems to be quite a bot |
| 11:51.34 | MaliutaLap | scruz: show module like chan_ |
| 11:51.49 | kaldemar | MaliutaLap: John is not a number. :) |
| 11:51.51 | MaliutaLap | <3 jbot |
| 11:52.03 | MaliutaLap | kaldemar: but is a valid callerID |
| 11:52.12 | MaliutaLap | kaldemar: :) |
| 11:52.23 | scruz | nope, it's not loaded. any way to load it? |
| 11:52.37 | scruz | maybe i should build asterisk 1.4 on this box |
| 11:52.43 | MaliutaLap | kaldemar: to do it properly you'd need to test what is in the callerID string |
| 11:53.04 | MaliutaLap | scruz: well we don't know what your config is loading |
| 11:53.08 | kaldemar | MaliutaLap: well he can combine all the Say-applications if he wishes to get lots of information. |
| 11:53.33 | MaliutaLap | scruz: you may very well just have missed something in the config |
| 11:54.28 | MaliutaLap | he was no fun anyway |
| 11:55.10 | kaldemar | scruz: load chan_sip.so will probably give you some hints on what might be wrong. |
| 11:55.49 | scruz | sip support wasn't built in :'( |
| 11:55.56 | scruz | freaking office servers |
| 11:56.12 | MaliutaLap | kaldemar: probably nothing wrong except in modules.conf ... but we can't say for sure with out an attempted load and/or a config |
| 11:56.32 | scruz | Unable to load module chan_sip.so |
| 11:56.33 | scruz | Feb 4 12:33:40 WARNING[12069]: loader.c:326 __load_resource: /usr/lib/asterisk/modules/chan_sip.so: cannot open shared object file: No such file or directory |
| 11:57.33 | MaliutaLap | does that file exist? is * looking for modules in the right places? |
| 11:57.51 | MaliutaLap | does it exist somewhere else on the system? |
| 11:58.23 | scruz | new to linux...any way to search? |
| 11:58.33 | scruz | find doesn't work |
| 11:58.56 | scruz | no results from whereis |
| 11:59.36 | MaliutaLap | locate? |
| 11:59.38 | MaliutaLap | find? |
| 11:59.55 | MaliutaLap | if find doesn't work you're doing it wrong ... |
| 12:00.01 | MaliutaLap | but I bet you hear that alot ;P |
| 12:00.08 | kaldemar | if there are no results, it doesn't mean that find doesn't work. it means that there is no such file or you're using find wrong if the file really exists. |
| 12:00.18 | MaliutaLap | find / -type f -name '*sip.so' |
| 12:00.42 | MaliutaLap | if that returns nothing and #? is 0 then the file doesn't exist |
| 12:00.51 | MaliutaLap | find is a very powerful tool when used right |
| 12:04.25 | scruz | i found it somewhere, but it seems nothing short of building asterisk myself will solve this |
| 12:04.53 | scruz | i cannot load the shared object because of some undefined symbol |
| 12:05.16 | scruz | guess i need to read up the man pages for find |
| 12:09.44 | MaliutaLap | that sounds like the module you have is for the wrong arch |
| 12:09.51 | MaliutaLap | what disr |
| 12:09.56 | MaliutaLap | distro even |
| 12:19.43 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) |
| 12:20.11 | *** join/#asterisk propellerhead (n=yogurt2u@host15.190-30-186.telecom.net.ar) |
| 12:21.05 | *** join/#asterisk Dovid (n=annon@tony09-121-90.inter.net.il) |
| 12:22.29 | scruz | fedora |
| 12:22.30 | *** join/#asterisk matt_ (n=matt@mattspc.ipv6.mattstone.net) |
| 12:23.08 | scruz | i can't dl asterisk 1.2? |
| 12:24.05 | *** join/#asterisk path_ (n=path@190.21.121.27) |
| 12:24.54 | MaliutaLap | I wouldn't touch 1.2 at this stage |
| 12:25.13 | MaliutaLap | I thought someone had packaged for Fedora ... it should be usable |
| 12:25.26 | MaliutaLap | I normally roll my own packages anyhow |
| 12:26.53 | scruz | why not? touch 1.2? we use 1.2 here |
| 12:27.20 | *** join/#asterisk Subdolus (n=subby@subby.afraid.org) |
| 12:28.55 | MaliutaLap | because it's ancient and if you run into problems most ppl will tell you to upgrade |
| 12:29.06 | scruz | anyhoo, what goes for SIP should go for IAX2, right? just add the config info in the IAX config instead |
| 12:29.27 | scruz | our dinosaur works just fine, thankee ;) |
| 12:29.34 | MaliutaLap | 1.4 fixed lots of stuff, 1.6 is better still (I need to get around to putting 1.6 on my poor little PIII Celery 733) |
| 12:29.57 | MaliutaLap | IAX is a little different, 'specially with nat |
| 12:30.06 | MaliutaLap | most of the config is similar enough |
| 12:31.15 | scruz | no nat involved. local network |
| 12:31.19 | scruz | ^_^ |
| 12:31.47 | MaliutaLap | they _all_ say that at the begining |
| 12:31.49 | MaliutaLap | ;) |
| 12:32.12 | MaliutaLap | IAX is much nicer to nat |
| 12:32.40 | scruz | i gathered |
| 12:32.54 | scruz | but it's a local network. really. 0 nat inside |
| 12:33.49 | scruz | i've got to dl asterisk 1.4 on a linux box, transfer to a windows box, then transfer to another linux box for building |
| 12:33.59 | scruz | T_T |
| 12:34.13 | *** join/#asterisk AdvoWork (n=AdvoWork@unaffiliated/advowork) |
| 12:38.20 | *** join/#asterisk ingenius (n=alektro@69.90.72.173) |
| 12:42.32 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.207) |
| 12:43.59 | ultrav1olet | We have one telephony ZAP line and when it's busy the next person trying to call receives Normal Clearing message when he or she tries to call this line. How can I turn this message into sound message saying "The line is currently busy. Call again later" or somethingh like this? |
| 12:44.03 | *** join/#asterisk ScriptFanix (i=vincent@2a01:e35:2f43:ae90:21a:70ff:fea3:44ab) |
| 12:44.24 | ultrav1olet | and one more question: right now our asterisk doesn't detect BUSY signal on Zap channel, so when you have finished calling you will be listening to busy signals indefinitely. busydetect option is set to on however it doesn't help |
| 12:44.50 | MaliutaLap | ultrav1olet: user calling out of your system or into it over ZAP? |
| 12:45.09 | MaliutaLap | ultrav1olet: if it's outside you'll need to talk to the PSTN provider |
| 12:45.10 | ultrav1olet | calling out |
| 12:45.40 | ultrav1olet | I see some options related to busy signal detection - what if I need to change them? |
| 12:47.01 | MaliutaLap | so you could record the msg anyway you want (on a pc and convert or in a recording studio and convert ... even using record()) and the set the exten[num+100] to Play($file) |
| 12:49.24 | MaliutaLap | if a line is busy on an attempted dial you jump to exten[#+100] .. so if dial is at exten xxx => s,2,Dial(ZAP/foof/bar) and it's busy you end up at exten => s,102,Stuff(tm) |
| 12:49.32 | MaliutaLap | it's in the manual for Dial() |
| 12:49.36 | MaliutaLap | RTFM :) |
| 12:55.35 | *** join/#asterisk coppice (n=chatzill@201.193.17.210.dyn.pacific.net.hk) |
| 12:57.45 | *** join/#asterisk Khratos (n=khratos@190.166.103.180) |
| 12:58.20 | Khratos | 'morning |
| 12:58.24 | scruz | what would cause an iax reg req to be rejected? there's no scret |
| 12:58.32 | scruz | s/scret/secret |
| 13:01.09 | MaliutaLap | type=? |
| 13:01.40 | MaliutaLap | and host=? |
| 13:01.52 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 13:01.58 | MaliutaLap | if host is dynamic you _must_ have a secret |
| 13:02.13 | kaldemar | MaliutaLap: actually, priorityjumping is not enabled by default anymore. priorityjumping=yes needs to be set to enable jumping for the applications that support it. and the jump is +101. ;) |
| 13:02.24 | scruz | type=frined |
| 13:02.32 | scruz | type=friend |
| 13:02.34 | sipy_away | Why is that not in the docs ANYWHERE??? |
| 13:02.36 | MaliutaLap | kaldemar: that is in 1.6? da? |
| 13:02.48 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 13:02.50 | scruz | host is static, so i put in the ip addr |
| 13:02.52 | MaliutaLap | scruz: and host? |
| 13:02.56 | MaliutaLap | kewl |
| 13:03.13 | MaliutaLap | scruz: you run a debug on the connection? |
| 13:03.26 | kaldemar | MaliutaLap: it's been like that since 1.2 releases. |
| 13:03.26 | MaliutaLap | that should tell you why it failed |
| 13:03.32 | scruz | i commented the permit/deny blocks, and still the same |
| 13:03.51 | scruz | no...how can i? |
| 13:04.01 | MaliutaLap | scruz: iax debug |
| 13:04.18 | kaldemar | pretty much no point in using registers with static hosts. |
| 13:04.31 | MaliutaLap | scruz: or iax2 set debug |
| 13:04.48 | MaliutaLap | kaldemar: something seems screwy |
| 13:05.04 | scruz | Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ |
| 13:05.05 | scruz | <PROTECTED> |
| 13:05.05 | scruz | <PROTECTED> |
| 13:05.05 | scruz | <PROTECTED> |
| 13:05.05 | scruz | brooks2*CLI> |
| 13:05.05 | scruz | Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGREJ |
| 13:05.07 | scruz | <PROTECTED> |
| 13:05.07 | MaliutaLap | scruz: oh, and did you reload the conf after making all these changes? |
| 13:05.09 | scruz | <PROTECTED> |
| 13:05.11 | scruz | <PROTECTED> |
| 13:05.12 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 13:05.13 | scruz | brooks2*CLI> |
| 13:05.14 | MaliutaLap | PASTEBIN |
| 13:05.17 | scruz | eek! |
| 13:05.19 | scruz | yes |
| 13:05.26 | scruz | my bad |
| 13:05.27 | MaliutaLap | is fecking off to fix his own shite |
| 13:05.29 | scruz | sorry |
| 13:05.39 | MaliutaLap | Disclaimer: not * related |
| 13:05.58 | *** join/#asterisk aksyn (n=aksyn@94-193-98-124.zone7.bethere.co.uk) |
| 13:10.44 | *** join/#asterisk beek (n=klinebl@pdpc/supporter/professional/beek) |
| 13:12.07 | *** join/#asterisk DarkRift (n=dark@bas1-sthubert21-1096611993.dsl.bell.ca) |
| 13:12.40 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 13:20.32 | *** join/#asterisk dlewis (i=c7340d66@about/security/staff/dlewis) |
| 13:21.38 | *** join/#asterisk shido6 (n=shido6@96-28-34-156.dhcp.insightbb.com) |
| 13:30.41 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 13:34.56 | *** join/#asterisk defswork (n=andy@mx2.3gcomms.co.uk) |
| 13:36.28 | LuisTorres | Howdy |
| 13:36.50 | LuisTorres | anybody knows out to do outbound fax detection? |
| 13:38.50 | *** join/#asterisk ta^3 (n=tacvbo@189.146.186.223) |
| 13:41.07 | coppice | listen for the pleasant gentle burbling of the V.21 preamble? |
| 13:42.30 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 13:42.34 | *** part/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 13:42.36 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 13:46.17 | Mr_BOnD_007 | can i install on REDHAT ? ASterisk ? |
| 13:46.31 | Mr_BOnD_007 | zptel and other library will be there with REDHat? |
| 13:46.36 | loompek | the question should be... can i install asterisk on redhat? |
| 13:46.37 | loompek | :D |
| 13:46.45 | loompek | umm |
| 13:47.15 | loompek | i successfully installed asterisk and zaptel dummy on centos, which is some kind of a 'free rhel' |
| 13:48.47 | *** join/#asterisk propellerhead (n=yogurt2u@host15.190-30-186.telecom.net.ar) |
| 13:49.16 | [TK]D-Fender | Mr_BOnD_007: Yes <- |
| 13:49.49 | [TK]D-Fender | Mr_BOnD_007: * will not COME pre-installed, but you can install it yourself |
| 13:49.57 | [TK]D-Fender | Mr_BOnD_007: Just like everybody else. |
| 13:51.00 | MaliutaLap | waves to [TK]D-Fender |
| 13:51.02 | Mr_BOnD_007 | [TK]D-Fender okie sir i need to download from site and install just asterisk that's all ? or some thing else ? packages i need to download and install |
| 13:51.32 | [TK]D-Fender | Mr_BOnD_007: go read THE BOOK, and the INSTRUCTIONS tell you what *'s requirements are for packages. |
| 13:51.40 | MaliutaLap | Mr_BOnD_007: there are packages, there are also files it the tgz that will tell you what you need |
| 13:51.59 | [TK]D-Fender | Mr_BOnD_007: a stock install of RH can come with all the core stuff * needs right from the start |
| 13:52.00 | MaliutaLap | Mr_BOnD_007: and google has all this for you to find, you just need to use your foo |
| 13:52.01 | [TK]D-Fender | ~book |
| 13:52.02 | jbot | book is probably probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 13:52.14 | MaliutaLap | mmm boook |
| 13:52.43 | [TK]D-Fender | Mr_BOnD_007: http://www.voip-info.org/wiki/view/CentOS+5+and+Asterisk+1.4.x+installation |
| 13:52.55 | Faustov | i has the book! :P |
| 13:52.55 | [TK]D-Fender | Mr_BOnD_007: effectively the same as RHEL 5 |
| 13:53.04 | Mr_BOnD_007 | okie [TK]D-Fender thanks |
| 13:53.53 | Mr_BOnD_007 | one more thing i want to ask that wat's this VICIDIAL ? |
| 13:56.09 | [TK]D-Fender | Mr_BOnD_007: something you can GOOGLE |
| 13:57.00 | Mr_BOnD_007 | okie i got it |
| 13:57.02 | Mr_BOnD_007 | Thanks |
| 13:59.31 | MaliutaLap | covers his butt re mothers b'day |
| 13:59.54 | MaliutaLap | 'specially since they are putting $$$'s in my account for hip surgery related things |
| 13:59.59 | dominic1 | short question: my telephone is using g722 and fallback to alaw, iax trunk alaw. If I dial a number over iax I get the error: Don't know any of 0x6000 formats |
| 14:01.16 | DavidR2008 | anyone have any aastra experience? I can get a soft phone to register, but not my aastra hard phone. |
| 14:01.42 | MaliutaLap | blow me down! g722 is supported |
| 14:02.20 | MaliutaLap | hard phones can be "interesting" to configure if you don't have the right docs to read |
| 14:02.46 | MaliutaLap | or you're dhcp is screwed and they're not hitting your tftp server |
| 14:03.15 | DavidR2008 | I think it's a docs issue, I've got it reading the config from my tftp server. |
| 14:05.41 | MaliutaLap | the right config? ;) |
| 14:05.44 | *** join/#asterisk jetlagmk2 (i=jetlag@pool-70-104-80-249.pskn.east.verizon.net) |
| 14:07.24 | DavidR2008 | I think that might be the million dollar question right there :-) |
| 14:08.06 | MaliutaLap | so strip out all but the one you want it to hit and put in something kinky to see if it picks it up |
| 14:08.15 | MaliutaLap | or read the logs really well |
| 14:08.44 | dominic1 | how can I see which codecs 0xe703 mean? |
| 14:09.02 | MaliutaLap | including matching the filename with the mac address (if they look the same way cisco's do for a file with the MAC in it) |
| 14:09.18 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:09.36 | MaliutaLap | dominic1: core show codecs much? |
| 14:09.46 | DavidR2008 | well I know it's picking up my config (tcpdump verifies that) I just don't know that my config is correct. :-S |
| 14:09.50 | MaliutaLap | ok, so that one was a tad narky |
| 14:10.16 | MaliutaLap | DavidR2008: so RTFM for you? ;) |
| 14:10.33 | MaliutaLap | I can do cisco's, they're easy |
| 14:11.10 | MaliutaLap | haven't got any astra experience though ... nobody will buy me hardware to play with |
| 14:11.32 | MaliutaLap | _thinks_ the the misso's use astras |
| 14:12.01 | MaliutaLap | I should beg one to play with |
| 14:12.05 | DavidR2008 | well I can't find the FM, I was hoping someone might be able to point me to the right FM (I found one via google, but it isa draft and seems to have some errors) or someone might have some experiance and be able to guide me through a very basic config |
| 14:12.52 | MaliutaLap | FMs can be a PITA sometimes, other you end up ROFPML at what they call an FM |
| 14:13.02 | MaliutaLap | has a TLA crisis |
| 14:14.25 | dominic1 | 0xe703 (g723|gsm|g729|speex|ilbc) |
| 14:15.23 | dominic1 | I don't know why asterisk always wants to encode it to these codecs first and then to alaw |
| 14:15.41 | MaliutaLap | dominic1: because you screwed the conf? |
| 14:16.14 | MaliutaLap | codec selection is dependant on many things, and is infact a negotiation with the the other end |
| 14:16.30 | dominic1 | now I have phone to asterisk1 G722 ; asterisk1 -> asterisk2 gsm (slin write format), and asterisk2 -> isdn (alaw) |
| 14:16.35 | MaliutaLap | may have something to do with you allowing something the other person wants higher |
| 14:16.48 | dominic1 | my codec order for iax trunk is alaw;g722;gsm |
| 14:17.00 | dominic1 | codec order for my phones is G722; alaw |
| 14:17.13 | MaliutaLap | so they will try g722 first |
| 14:17.51 | MaliutaLap | any reason you're using g722 on the phones? |
| 14:18.09 | dominic1 | better quality for internal speaking |
| 14:18.44 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 14:18.50 | MaliutaLap | you read http://lists.digium.com/pipermail/asterisk-dev/2004-December/008064.html I take it? |
| 14:21.01 | Gido-E | is there anyreason that callpickup does not work in 1.4.23.1 and worked in 1.4.22? |
| 14:21.20 | MaliutaLap | right, bed time! |
| 14:21.20 | dominic1 | okay I am an idiot |
| 14:21.36 | dominic1 | didn't see bandwidth=low in iax.conf *dong* |
| 14:21.39 | MaliutaLap | dominic1: I wasn't going to say it .... ;) |
| 14:21.58 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 14:21.58 | MaliutaLap | 'leepies! |
| 14:22.00 | dominic1 | MaliutaLap: thank you ;-) |
| 14:23.23 | Ryushin | I've done about 6 asterisk set for businesses using PRI's and analog. I'm thinking of setting up an asterisk box for home use. I'm wondering if a BRI can act like a mini PRI so that I can have multiple DID's? |
| 14:23.23 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 14:23.32 | Ryushin | I'm in the US as well. |
| 14:26.30 | dominic1 | does slin have the best quality? |
| 14:26.32 | *** join/#asterisk bamsefar (i=emj@core.serv.emj.se) |
| 14:26.50 | dominic1 | then g722 and then g711? |
| 14:27.05 | Corydon76-dig | Heh. Considering slin is uncompressed audio, yes |
| 14:27.24 | Mr_BOnD_007 | how to check where GCC compiler is intalled or not ? |
| 14:27.38 | Corydon76-dig | though slin16 is a bit better |
| 14:27.46 | bamsefar | Hi, I need to make a cluster that simply "listens" to incoming calls, to benchmark an application. How would I go about distributing the calls to my asterisk boxes in the best way? |
| 14:28.11 | bamsefar | A simple round-robin seems like a good idea, but do I use Asterisk for this or something else like regular PAT/NAT or SER? |
| 14:28.16 | dominic1 | core show codecs just shows me slin (16 bit Signed Linear PCM). This should be slin16, right? |
| 14:28.28 | Corydon76-dig | Nope, that's slin |
| 14:28.45 | Corydon76-dig | The difference is in rate, not number of bits |
| 14:29.09 | Mr_BOnD_007 | make menuselect Error1 the confugure script must be excuted before running 'make' ? |
| 14:29.10 | Corydon76-dig | slin is 8000Hz, slin16 is 16000Hz |
| 14:29.20 | Corydon76-dig | Mr_BOnD_007: ./configure |
| 14:29.23 | dominic1 | So allow=slin16 should help, correct? |
| 14:29.32 | Mr_BOnD_007 | Corydon76-dig how to do that ? |
| 14:29.53 | Corydon76-dig | dominic1: slin is an internal format that isn't generally used for voip clients |
| 14:30.04 | Corydon76-dig | Mr_BOnD_007: type that |
| 14:30.25 | Mr_BOnD_007 | ? where in same directory ? asterisk ? /usr/src ? |
| 14:30.44 | Corydon76-dig | In the Asterisk source directory, same place as you typed 'make' |
| 14:31.12 | Mr_BOnD_007 | gcc no cc no cl exe no no acceptable c compiler found in $path |
| 14:31.15 | Mr_BOnD_007 | ya done |
| 14:31.29 | Corydon76-dig | What distro? |
| 14:31.50 | [TK]D-Fender | Mr_BOnD_007: go read the WIKI page I gave you, it tells you how to install all of *'s dependencies <--- |
| 14:31.58 | [TK]D-Fender | Mr_BOnD_007: http://www.voip-info.org/wiki/view/CentOS+5+and+Asterisk+1.4.x+installation <------ |
| 14:32.02 | Mr_BOnD_007 | okie |
| 14:33.16 | Ryushin | [TK]D-Fender: How much do you know about BRI's? Can they be used like mini PRI's so that I can have DID's and such? |
| 14:33.16 | Mr_BOnD_007 | i think i have unziped the 686 64Bit 1.6.0.5 something |
| 14:33.43 | [TK]D-Fender | Ryushin: Yes BRI supports DID's in the form of MSN's IIRC |
| 14:33.47 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 14:33.57 | [TK]D-Fender | Ryushin: No direct experience, just bits I've read |
| 14:34.31 | [TK]D-Fender | Mr_BOnD_007: 1.6's core dependencies are the same as 1.4's for the most-part |
| 14:34.46 | Ryushin | I've been meaning to set up a Asterisk box at my house, and I wanted to get something that was cost affective and still provided the same learning potential of a PRI. |
| 14:35.12 | dominic1 | Corydon76-dig: Thank you very much for you help! |
| 14:35.39 | [TK]D-Fender | Ryushin: The concept of "for learning" with * and any kind of hardware, especially for learning as an analogy to ANOTHER tech is worthless. |
| 14:35.56 | [TK]D-Fender | Ryushin: Like all those people who get an X100P to "learn zaptel". |
| 14:36.18 | [TK]D-Fender | Ryushin: Configuring Zaptel is a TINY shit-for-all portion of configuring *. |
| 14:36.56 | [TK]D-Fender | Ryushin: so you "learn" how to make 20 lines of config files..... and then causually go on to just placing calls. |
| 14:37.03 | Ryushin | I know. But I'd rather have the flexibility of having DID's and such, instead of just static analog lines. |
| 14:37.20 | Ryushin | Sangoma's new hybrid card got me thinking about it for home use. |
| 14:37.21 | [TK]D-Fender | Ryushin: Same goes for "I wannt set up an IAX soft-phone" people. |
| 14:37.49 | [TK]D-Fender | Ryushin: Physical lines cost you in monthly service fees and in hardware. For what? Want DID's? VoIP work just fine |
| 14:37.51 | *** join/#asterisk [intra]lanman (n=Raymond@freeswitch/developer/intralanman) |
| 14:38.03 | [TK]D-Fender | Ryushin: Yeah, Sangoma's line is kinda badd-ass these days |
| 14:38.18 | [TK]D-Fender | Ryushin: B600 = awesome value by todays standards |
| 14:38.44 | [TK]D-Fender | Ryushin: and I've seen the BRI/Analog model specs as well... nifty... but a very niche product |
| 14:39.18 | Mr_BOnD_007 | GCC NO cc No el.exe no same error |
| 14:39.44 | Ryushin | yea, it is going to be a niche product. But it was cool enough that I can have BRI's coming in, and the analog for the house and fax machine. |
| 14:39.47 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 14:40.30 | [TK]D-Fender | Ryushin: Well... I'm sure it does its job... if it really fits your needs in one shot, go for it. |
| 14:40.38 | [TK]D-Fender | Ryushin: Just don't say "testing" ok? :) |
| 14:41.01 | [TK]D-Fender | Ryushin: "z0mg a perfect fit!" is a perfectly valid reason... |
| 14:41.22 | [TK]D-Fender | Ryushin: Esp as we know you don't want to cram a bunch of cards in 1 box for that |
| 14:41.31 | Ryushin | Okay, how aobut furthering my education in asterisk in a home environment. |
| 14:41.40 | *** join/#asterisk eric2 (n=ejc@unused-74-51-54-37.vianet.ca) |
| 14:41.42 | [TK]D-Fender | Ryushin: :/ |
| 14:41.52 | [TK]D-Fender | Ryushin: FFS use that thing... lots :) |
| 14:42.06 | Ryushin | I saw Sangoma's new USB analog. It looks like Sangoma is really starting to branch out. |
| 14:42.41 | [TK]D-Fender | Ryushin: Thy are... the B600 pwns the SMB server for analog use... |
| 14:43.07 | Ryushin | I'm going to have to look at that too. First I've seen it. |
| 14:43.59 | Dovid | anyone here work with a Grandstream 286 ? Seems to always send + infront of the number called. I don't see any setting for it. The prefix option is blank |
| 14:44.06 | *** join/#asterisk timeshell (n=chatzill@gw.lusi.on.ca) |
| 14:44.17 | timeshell | Greetings. |
| 14:44.33 | timeshell | Any word on the digium chan_skype release? |
| 14:44.45 | [TK]D-Fender | timeshell: "when its done" |
| 14:44.51 | timeshell | lol |
| 14:44.58 | [TK]D-Fender | timeshell: "Next spring...SHARP!" |
| 14:45.02 | timeshell | I guess that means no |
| 14:45.18 | [TK]D-Fender | timeshell: "arewethereyetAREWETHEREYETarewethereyetAREWETHEREYETarewethereyetAREWETHEREYETarewethereyetAREWETHEREYETarewethereyetAREWETHEREYET" |
| 14:45.24 | timeshell | :D |
| 14:45.37 | timeshell | Hey, I only ask every couple weeks. |
| 14:45.39 | Dovid | haha |
| 14:46.00 | Gido-E | chan_skype would be UBER kewl :-) |
| 14:46.01 | timeshell | I figure the first place that's going to know is here. |
| 14:46.53 | timeshell | And I'm really anxious to get rid of www.chanskype.com's channel. I find it annoying. |
| 14:47.30 | timeshell | Especially since asterisk-gui doesn't work with it. |
| 14:47.53 | *** join/#asterisk JayTee52 (n=jforde@unaffiliated/jaytee) |
| 14:48.08 | [TK]D-Fender | timeshell: Considering that its taken forever for it to work with DAHDI.... I wouldn't get my hopes up about it... |
| 14:48.40 | timeshell | Well, that's not very positive. If one has no hope, nothing gets done. :D |
| 14:49.18 | timeshell | I guess I'll get around to adding support for it in the gui someday. |
| 14:49.22 | timeshell | I just don't have time. |
| 14:50.03 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 14:52.45 | *** join/#asterisk adam000 (n=adam@c-76-97-76-93.hsd1.ga.comcast.net) |
| 14:53.00 | *** join/#asterisk Nasra (n=maxshipp@99.244.127.8) |
| 14:53.14 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 14:55.04 | *** join/#asterisk DarylVOIP (n=daryl@75.147.121.177) |
| 14:59.47 | *** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-bccf72b386862e9d) |
| 14:59.47 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:00.19 | *** join/#asterisk sack (n=sack@1.Red-81-34-163.dynamicIP.rima-tde.net) |
| 15:01.41 | dominic1 | is there any possibility to change the callerid of the person I called in my display? A few months ago I saw there is a feature in the sip protocol to do this |
| 15:03.29 | *** join/#asterisk naitram (n=naitram@12.105.199.38) |
| 15:04.32 | *** join/#asterisk mnicholson_ (n=matthew@72.146.43.239) |
| 15:04.51 | naitram | is there an archive of older deb packages, looking for 1.2.20 deb |
| 15:06.30 | rob0 | Probably someone here will know, but isn't that a #debian question? |
| 15:07.49 | rob0 | dom, core show function CALLERID (not sure if that's the best way) |
| 15:08.01 | naitram | rob0: will try there |
| 15:08.30 | xrmx__ | naitram, see http://svn.debian.org/wsvn/pkg-voip/ |
| 15:10.26 | *** join/#asterisk deadpigeon (n=deadpige@office.xpressamerica.net) |
| 15:10.53 | dominic1 | rob0: But I thought with the callerid function I just can change the callerid of the caller and not the callerid in my own display to see who am I calling |
| 15:11.09 | [TK]D-Fender | dominic1: --> |
| 15:11.09 | [TK]D-Fender | ~cpid |
| 15:11.10 | jbot | [~cpid] Called-Party ID is possible with * using patches on Mantis. See : http://bugs.digium.com/view.php?id=8824 |
| 15:12.02 | rob0 | I want to set up a site-to-site connection via SIP, so each site can call the other's extensions. Static IP via VPN, so NAT is not an issue. Is a "type=friend" what I need, or should I do separate user & peer? |
| 15:12.21 | dominic1 | cool looks like it's already in 1.6 |
| 15:13.02 | rob0 | BTW each site has easily distinguished extensions, _6XXX and _7XXX |
| 15:13.30 | rob0 | and each is 1.6.x |
| 15:15.25 | rob0 | I don't see a good example of this in the sample sip.conf |
| 15:15.46 | [TK]D-Fender | rob0: "friend" was all but phased out in 1.4 |
| 15:16.09 | [TK]D-Fender | rob0: Its little different than setting up any other ITSP |
| 15:16.15 | [TK]D-Fender | rob0: SIP is SIP.... |
| 15:16.47 | rob0 | so set a peer up for inbound from the other site, and a user for outbound to the other site |
| 15:17.15 | mort_gib | rob0: What's wrong with IAX |
| 15:17.47 | dominic1 | okay, the function seems not to be available in 1.6.0.5 :-( |
| 15:17.53 | [TK]D-Fender | rob0: both "type=peer" |
| 15:18.02 | [TK]D-Fender | rob0: but yes, 2 accounts |
| 15:18.25 | rob0 | Would IAX make it simpler in any way? I'm already using SIP, don't otherwise need to add another protocol. |
| 15:19.11 | [TK]D-Fender | rob0: No |
| 15:20.12 | mort_gib | I though that IAX would perform slightly better in trunk mode.... |
| 15:20.39 | [TK]D-Fender | mort_gib: if you NEED the BW |
| 15:20.51 | [TK]D-Fender | mort_gib: Otherwise its trouble you jsut don't need |
| 15:21.06 | mort_gib | Hey, you ALWAYS need the bw |
| 15:21.08 | mort_gib | :-) |
| 15:21.50 | rob0 | how much difference, roughly, are we talking about? Both sites are at least 256Kbps up. |
| 15:22.27 | mort_gib | rob0: And the bw is reserved for SIP?? |
| 15:22.28 | rob0 | probably won't matter, since there's only one phone at one of the sites, won't be multiple active calls at once :) |
| 15:23.33 | [TK]D-Fender | rob0: NO point then :) |
| 15:25.14 | DarylVOIP | Does anyone know a reasonable string to sent to an Asterisk box to see if IAX2 is working? I'm trying to check it with an F5 BigIP rule and can send <something> and need to get back some response that I can predict at least a portion of. |
| 15:26.52 | *** join/#asterisk _Roman (n=roman@92.39.196.250) |
| 15:27.43 | *** join/#asterisk icebrew54 (i=proxy@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 15:28.16 | *** join/#asterisk sasargen (n=chatzill@72-58-224-209.pools.spcsdns.net) |
| 15:28.35 | _Roman | Hello, I was looking at asterisk a while ago, I found a command line tool that let me look at the status of a PSTN line (was it on/off hook, etc) for diagnostic purposes. Does anyone know what that command was? |
| 15:29.21 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 15:29.41 | *** join/#asterisk riddlebox (n=user@mscitspubwlgw.wustl.edu) |
| 15:29.54 | jameswf | I should look at tieing this google latitude thing in to asterisk.... |
| 15:31.28 | rob0 | So, I need two type peer, secret (won't hurt), host, and context. The context would be something like [site1-in] and [site1-out]. I use Dial(SIP/${EXTEN}@site1-out) to get to site1 from site2, and that hits [site2-in] context on site2. |
| 15:32.09 | rob0 | um, I didn't mean the [] as context, but those would be the sections in sip.conf |
| 15:33.38 | *** join/#asterisk seanmh (n=johndoe@abq-216-31-109-157.dsl.zianet.com) |
| 15:35.40 | rob0 | [site1-in] will go to context=to-site1 ; [site1-out] context=to-site1 (sounds clearer to me) |
| 15:37.37 | *** join/#asterisk Odd_Bloke (n=oddbloke@daniel-watkins.co.uk) |
| 15:41.06 | Odd_Bloke | Hello all. I'm looking to replace one of the stock Asterisk sounds with one of my own. Is it possible to do this without actually recording over the file on my filesystem? (i.e. what search path does Asterisk use for sounds?) |
| 15:41.09 | rob0 | maybe this is simple, or maybe I am confused :) ... about to try it |
| 15:41.49 | [TK]D-Fender | Odd_Bloke: copy the original somewhere else. |
| 15:41.56 | rob0 | by default sounds are in /var/lib/asterisk/sounds, paths are relative under that |
| 15:42.25 | *** join/#asterisk killown (n=Yamato@unaffiliated/killown) |
| 15:44.06 | *** join/#asterisk ghenry (n=ghenry@ghenry.plus.com) |
| 15:44.43 | ghenry | what's the cause again for not accepting a password at voicemail or DISA? The password is right, but I've hit this before when amodule wasn't loaded for some reason. Anyone remember? |
| 15:45.53 | [TK]D-Fender | ghenry: "core show application voicemailmain" , "core show application disa" |
| 15:46.05 | Odd_Bloke | [TK]D-Fender: OK, thanks. :) |
| 15:46.34 | [TK]D-Fender | ghenry: Neither of these should refuse a correct PW. |
| 15:46.46 | ghenry | I'm not sure. |
| 15:46.51 | [TK]D-Fender | ghenry: Meetme can report back a bad PW if no Zaptel timer is available |
| 15:46.52 | ghenry | it's on the tip of my toungue |
| 15:46.59 | ghenry | yeha, that was it |
| 15:47.05 | ghenry | that's what I was thinking of |
| 15:47.09 | [TK]D-Fender | ghenry: But neither of the others has any dependency |
| 15:47.10 | ghenry | will check console |
| 15:47.13 | ghenry | yeah |
| 15:49.11 | *** join/#asterisk flujan (n=flujan@internet.nube.com.br) |
| 15:49.18 | flujan | hello guys. |
| 15:49.24 | flujan | I have the two asterisk box |
| 15:49.39 | flujan | on the fist box, i have two "sip trunks" that connets to the second box. |
| 15:49.46 | flujan | here is the sip.confs |
| 15:49.56 | flujan | box1 http://pastie.org/379493 |
| 15:50.10 | flujan | box2 http://pastie.org/379494 |
| 15:50.22 | flujan | the problem is with the incoming calls on box1 |
| 15:50.32 | flujan | the calls enters box2 and dial to the sip of box1 |
| 15:51.13 | flujan | a show channel commands just show channels from the first sip trunk ... no matter what asterisk consider all calls comming from box 2 to box 1 as comming from the first sip trunk that registers |
| 15:51.20 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 15:51.23 | flujan | is it a bug or i need to do additional configuration? |
| 15:51.51 | flujan | box1 is registering at machine two |
| 15:53.19 | [TK]D-Fender | flujan: insecure=very <------- |
| 15:53.40 | flujan | hey [TK]D-Fender but on box one or box two? |
| 15:53.48 | [TK]D-Fender | flujan: First come... first served. |
| 15:54.42 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 15:54.52 | Dovid | has anyone used stream_file in an agi ? no matter what I try the value of the input comes back as 50 |
| 15:55.08 | [TK]D-Fender | flujan: http://pastie.org/379493 <-- between lines 45-47... notice something missing? |
| 15:55.46 | flujan | [TK]D-Fender: didn't pastie a extension that i use... no problems with that |
| 15:55.55 | flujan | i will remove insecure and give it a try. |
| 15:58.20 | Dovid | upto how long can an extension be in asterisk ? |
| 15:58.46 | loather | as many as you want |
| 15:59.00 | Dovid | so it can be 2000 if i want ? |
| 15:59.11 | *** part/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 15:59.21 | Faustov | less than 2000 can get lost in dtmf, i'd use 3000 or more |
| 15:59.31 | Dovid | lol |
| 15:59.41 | loather | well, there's probably a practical limit, but it's likely high enougn that you'd never hit it under normal circumstances |
| 15:59.42 | Dovid | because I am trying to use an AGI to get a logn string of digits |
| 15:59.48 | *** part/#asterisk bamsefar (i=emj@core.serv.emj.se) |
| 16:00.20 | flujan | [TK]D-Fender: removing the insecure=very from the sip.conf i still have the same problem all calls from box2 to box1 shows the first sip trunk... not showing calls on the second one with show channels command. |
| 16:00.27 | *** join/#asterisk jasonwoot (n=some@bookit-dev.com) |
| 16:00.34 | Dovid | but i am not getting back what i need. if i use stream_file then i do not get back the correct value if I use get_data then it does not record the # sign |
| 16:00.37 | *** join/#asterisk dlewis (i=c7340d66@about/security/staff/dlewis) |
| 16:00.45 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 16:01.09 | [TK]D-Fender | flujan: Complete PB's please |
| 16:01.18 | flujan | [TK]D-Fender: ok |
| 16:01.43 | [TK]D-Fender | Dovid: In AGI don't read an exten <- |
| 16:01.51 | [TK]D-Fender | Dovid: this isn't a pattern match. |
| 16:02.16 | [TK]D-Fender | Dovid: And don't forget * has a var length and dialplan line lenth limit.... |
| 16:02.18 | Dovid | TK: That I know. i am not trying to get an extension. I want some one to enter a string of numbers along with * and # |
| 16:02.34 | [TK]D-Fender | Dovid: Fine then collect 1 char at a time YOURSELF in AGI |
| 16:02.39 | flujan | [TK]D-Fender: box1 updated http://pastie.org/379493 |
| 16:02.55 | Dovid | TK: I just have a loop that gets it all |
| 16:03.05 | Dovid | but my issue is that i cant seem to get # |
| 16:03.15 | Dovid | and with stream_file its sending me what i put in |
| 16:03.47 | *** join/#asterisk af_ (n=getsmart@88-149-230-108.dynamic.ngi.it) |
| 16:03.57 | *** join/#asterisk _Roman (n=roman@87.254.77.116) |
| 16:04.06 | [TK]D-Fender | Dovid: ?? |
| 16:04.31 | [TK]D-Fender | Dovid: And WTF are you using "stream_file" for? That doesn't say "read DTMF" to me... |
| 16:04.49 | rob0 | drat, first I have to get the dahdi FXS working. dahdi-genconf generated a dahdi-channels.conf, but it's not being parsed, do I need an include in chan-dahdi.conf? |
| 16:05.25 | Dovid | then i understood escape_string wrong. thought it would put that value in to the variable. |
| 16:05.33 | *** join/#asterisk BBHoss (n=bbhoss@c-68-62-168-242.hsd1.al.comcast.net) |
| 16:05.37 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 16:05.38 | [TK]D-Fender | Dovid: It does. |
| 16:05.48 | [TK]D-Fender | Dovid: You just don't understand what the VALUE IS |
| 16:05.57 | Dovid | $foo[value] |
| 16:06.31 | [TK]D-Fender | Dovid: You pressed "2" |
| 16:06.34 | *** join/#asterisk moy (n=chatzill@bas1-unionville55-1177733953.dsl.bell.ca) |
| 16:06.59 | JayTee52 | flujan, what version of Asterisk are you running? 1.2? or higher? |
| 16:07.07 | Dovid | TK: If I have $foo = $agi->stream_file('enter-data', $escape_digits='1234567890*#', $offset=0); |
| 16:07.22 | Dovid | if i verbose $foo[result] i get 50 when I press # or 1 or 2 |
| 16:07.24 | [TK]D-Fender | Dovid: Yes, and it escaped on "2" |
| 16:07.44 | Dovid | ok. so escaped on 2 meaning ? |
| 16:07.52 | [TK]D-Fender | DoYOU PRESSED 2 DAMMIT |
| 16:08.05 | [TK]D-Fender | Dovid: Please.... go caffeinate! |
| 16:08.18 | *** join/#asterisk sasargen_ (n=chatzill@72-58-224-209.pools.spcsdns.net) |
| 16:08.20 | [TK]D-Fender | grabs his ClueBat (tm) |
| 16:08.33 | JayTee52 | it's mad cow I tell ya! it's infected almost everyone. |
| 16:08.56 | [TK]D-Fender | prepares for a mass-purge |
| 16:09.05 | Dovid | lol |
| 16:09.10 | Dovid | ok so 1 was 49 2 was 5 |
| 16:09.21 | Dovid | i dont understand why is that. |
| 16:09.27 | [TK]D-Fender | *sigh* |
| 16:09.43 | [TK]D-Fender | I've seen larvae with greater deductive capabilities |
| 16:09.57 | [TK]D-Fender | goes to feed his maggot-farm |
| 16:10.03 | Dovid | haha |
| 16:10.06 | loather | maggots are nasty |
| 16:10.14 | [TK]D-Fender | disposes of the rest of the last newbs personal effects |
| 16:10.55 | loather | Dovid: find your nearest handy ASCII code chart and look up the decimal values for characters 49 and 50. Then wait for the lightbulb. |
| 16:11.05 | [TK]D-Fender | loather : Seen a great video of them used medically to remove infected material from patients. |
| 16:11.47 | Dovid | loather: THANK YOU !!!!!!!!!!!!!! |
| 16:11.52 | [TK]D-Fender | loather : amggots only eat necrotized flesh |
| 16:11.58 | *** join/#asterisk genin (i=phrame@ANice-252-1-70-22.w83-201.abo.wanadoo.fr) |
| 16:12.02 | *** join/#asterisk BipBip (n=BipBip@194.65.5.235) |
| 16:12.02 | genin | allo |
| 16:12.13 | rob0 | Oh duh, it's right there in the comments of the generated file. |
| 16:12.14 | loather | I've heard of that. It's actually an old folk remedy. But seeing as I loathe the creatures more than just about anything, I think i'd complain quite loudly if some were introduced into my festering wounds. |
| 16:12.41 | genin | anyone know anything about 3gp video streaming using asterisk and a T1 card |
| 16:12.52 | [TK]D-Fender | loather : Along with my resolute acceptance of death is the means by which life can be preserved. |
| 16:13.05 | [TK]D-Fender | loather : Helps when in the dentist's chair as well :) |
| 16:13.17 | BBHoss | yeah they used the maggots on a house episode once |
| 16:13.26 | loather | to be honest the dentist never really bothered me that much |
| 16:13.36 | Faustov | [TK]D-Fender: can they play chess? |
| 16:13.38 | [TK]D-Fender | BBHoss: Wouldn't put it past them... Only seen 2 eps of it personally. |
| 16:14.00 | BBHoss | [TK]D-Fender: i like the show, but the story can get repetitive at times |
| 16:14.03 | Faustov | BBHoss: true, poor kid |
| 16:14.14 | [TK]D-Fender | BBHoss: Hasn't been an original thought since 1969 :) |
| 16:14.27 | BBHoss | heh |
| 16:14.42 | loather | blame the hippies. |
| 16:14.55 | [TK]D-Fender | loather : or lack thereof |
| 16:17.16 | rob0 | [Feb 4 10:16:58] ERROR[27562]: config.c:1093 process_text_line: The file '= /etc/asterisk/dahdi-channels.conf' was listed as a #include but it does not exist. |
| 16:17.27 | rob0 | oh haha it's the = |
| 16:17.53 | *** join/#asterisk mrsci (n=sq@ppp-70-251-250-110.dsl.rcsntx.swbell.net) |
| 16:17.56 | rob0 | [Feb 4 10:17:40] ERROR[27562]: chan_dahdi.c:8394 mkintf: Signalling requested on channel 1 is FXO Loopstart but line is in FXO Kewlstart signalling |
| 16:18.44 | loather | if it can detect that then why bother configuring it? |
| 16:18.49 | genin | or better yet anyone know a chan on freenode where people talk about video streaming solutions |
| 16:18.50 | genin | ? |
| 16:18.55 | loather | (sorry, rhetorical). |
| 16:19.14 | *** join/#asterisk rue_mohr (n=rue@24.207.122.10) |
| 16:20.05 | *** join/#asterisk jsolis (n=jimmy@190.41.153.85) |
| 16:20.46 | flujan | JayTee52: 1.4.22 |
| 16:21.04 | flujan | [TK]D-Fender: any hint besides the insecure removal? |
| 16:21.28 | loather | why do you need two trunks? |
| 16:21.30 | JayTee52 | insecure=very is deprecated in 1.4 you want to use insecure=port,invite |
| 16:21.35 | [TK]D-Fender | flujan: I didn't get the COMPLETE picture like I asked. |
| 16:21.48 | [TK]D-Fender | JayTee52: And he shouldn't be using EITHER |
| 16:22.29 | flujan | [TK]D-Fender: http://pastie.org/379536 and the http://pastie.org/379493 sip.conf from box1 without the insecure |
| 16:22.38 | rob0 | So does dahdi-genconf get the signalling on FXS/FXO reversed? |
| 16:23.06 | flujan | loather: each trunk is associated with a E1 link on box 2. I need to separate them. |
| 16:24.20 | *** join/#asterisk slima (i=slima@unaffiliated/slima) |
| 16:24.38 | loather | then they each are going to need separate contexts. when a call arrives on the first e1, have it dial extensions in a context pertaining to the first trunk. when a call arrives on the second e1, have it dial an extension in the other context pertaining to the other trunk. |
| 16:25.53 | flujan | loather: they are... here is the sip.conf from box2 |
| 16:26.43 | loather | pastebin both the sip.conf and extensions.conf from the two machines and we'll take a look |
| 16:28.24 | loather | and the zaptel/dahdi conf from the box with the e1 spans |
| 16:28.42 | *** join/#asterisk disposable (i=disposab@blackhole.sk) |
| 16:29.31 | *** join/#asterisk ta^3 (n=tacvbo@189.146.171.23) |
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| 16:34.19 | flujan | loather: http://pastie.org/379549 |
| 16:34.56 | *** join/#asterisk cesau2 (n=cesau@66.94.94.66) |
| 16:34.57 | disposable | I've got static conference rooms for which i would like to be able to change PINs using a phone. has anyone built this into their dialplan before so that i don't have to reinvent the wheel? (call an extension, enter a conference room number, enter new pin, have it read back, reload asterisk, hangup) |
| 16:35.51 | flujan | loather: zaptel and zapata.conf are working, do you wanna see zapata.conf right? zaptel is just driver stuff. |
| 16:36.37 | cesau2 | if cli> show odbc ==> "Connection 1: connected" -- and yet i still get "Realtime mapping for 'sippeers' found to engine 'odbc', but the engine is not available" -- where can i do next to debug the problem? |
| 16:37.27 | loather | flujan: yeah, zapata. |
| 16:39.29 | flujan | loather: http://pastie.org/379558 |
| 16:40.08 | loather | ok, i'm stumped. it should do what you want it to do. |
| 16:40.51 | flujan | loather: my config is right? |
| 16:41.17 | flujan | loather: it is doing that but showing always that all calls comes from the e1 link |
| 16:41.37 | flujan | [TK]D-Fender: any tips? |
| 16:41.56 | loather | unless i'm missing something glaringly obvious, yeah. i'd expect it to do what you describe: use the first sip trunk for calls into the first span, and the second for the second span |
| 16:42.04 | [TK]D-Fender | disposable: Read new PIN and reload *? Why bother. Pure dialplan with AstDB <- |
| 16:42.11 | [TK]D-Fender | disposable: And yeah.. jsut code it yourself |
| 16:42.21 | *** join/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-100abccbbc48404f) |
| 16:42.22 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 16:43.27 | rob0 | rerunning dahdi_cfg seems to have helped, but still no dial tone :( |
| 16:43.44 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:44.01 | [TK]D-Fender | RobPlugged in the molex? |
| 16:44.54 | flujan | loather: yeap asterisk is missing the registry and is not reading the right sip trunk on box2 when it receives a call |
| 16:44.55 | rob0 | yup, how do I get debug to the console for dahdi? |
| 16:45.08 | rob0 | like to show offhook/onhook changes? |
| 16:45.11 | flujan | loather: i will try to ping the bug channel before openning one |
| 16:45.40 | rob0 | maybe I'll try a different phone too :) |
| 16:45.46 | rob0 | but I think this one works |
| 16:46.09 | rob0 | unfortunately no telco line to plug into |
| 16:47.28 | *** join/#asterisk aksyn (n=aksyn@94-193-98-124.zone7.bethere.co.uk) |
| 16:56.20 | elred | rob0: what kind of debug for dahdi do you want ? |
| 16:56.23 | prg3 | [TK]D-Fender: Sangoma has a 64-bit 4Gb ram option that is needed to be set for the cards I have.. set that, and my voice issues seem to be gone. need more testing to be sure.. next, I have to sort out my dialplans |
| 16:57.32 | elred | rob0: if you run asterisk with -dd option you will have lot of message from chan_dahdi.so |
| 16:57.49 | elred | rob0: otherwise you can load your drivers module with debug=1 and see in dmesg what appears |
| 16:58.05 | elred | or you can also use ztmonitor [channel] -v to follow the life of a channel |
| 16:58.08 | rob0 | core debug is 5, does that matter? |
| 16:58.20 | *** join/#asterisk erth64net (n=erth64ne@69-30-67-191.dq1sn.easystreet.com) |
| 16:58.39 | elred | I don't have in mind what core debug you need to do that. I personally run asterisk with -cvvvvvvvvvvvvv when I want debug output |
| 17:00.36 | *** join/#asterisk angler_ (n=angler@nat/digium/x-f85694e252f7ee41) |
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| 17:01.47 | rob0 | Well, something is working on the FXO. The inactive (but powered) telco line being unplugged caused a red alarm, plugging in again cleared it. |
| 17:01.51 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 17:01.58 | rob0 | but I need to get the FXS working :) |
| 17:03.02 | dominic1 | is there any support for g.772 in misdn or dahdi? |
| 17:03.08 | *** join/#asterisk fish-bulb (n=cstewart@nat/digium/x-5b6637b571f6d897) |
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| 17:03.24 | dominic1 | g.722,sorry |
| 17:03.45 | rob0 | The phone seems to work as best I can tell; being plugged into the telco line it generates tones when buttons are pushed. But nothing, when in the FXS. :( |
| 17:05.04 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
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| 17:09.16 | elred | rob0: did you ztcfg -vv properly before launching asterisk ? Is your /etc/zaptel.conf well configured ? In etc/asterisk/indications.conf do you have the rigth country= tag ? etc |
| 17:09.48 | rob0 | It's all dahdi, I'll start making a pastebin |
| 17:10.31 | rob0 | the FXS module has a lit LED, I'm plugged in right, that's for sure. |
| 17:13.38 | *** join/#asterisk NovceGuru (i=novcegur@server1.jsreedinc.com) |
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| 17:16.35 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 17:17.10 | demiv | hello there... WARNING[14299]: file.c:1162 waitstream_core: Unexpected control subclass '20' |
| 17:18.20 | demiv | that is an error by bandwitdh ? |
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| 17:41.07 | rob0 | http://pastebin.ca/1327246 summary of my dahdi woes |
| 17:41.10 | rob0 | elred: ^^ |
| 17:41.43 | *** join/#asterisk Qwell (i=north@pdpc/sponsor/digium/Qwell) |
| 17:41.43 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 17:41.59 | rob0 | BTW this FXS used to work, back when I had a phone line on the FXO. |
| 17:42.20 | *** join/#asterisk ScarEye (n=scareye@12.27.87.66) |
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| 17:46.49 | rob0 | Sorry for the repeat so soon, but being that there are new folks since then, including two Digiummy ... http://pastebin.ca/1327246 summary of my dahdi woes |
| 17:47.38 | *** join/#asterisk myselfhimself (i=5bc7062c@gateway/web/ajax/mibbit.com/x-f24b60581beb9e6d) |
| 17:47.43 | myselfhimself | hi !! |
| 17:48.18 | *** join/#asterisk jayrod422 (n=jayrod42@node2.164.136.64.1dial.com) |
| 17:48.29 | rob0 | I guess the plural of Digium is Digia. |
| 17:49.00 | *** join/#asterisk jeffgus (n=jeffgus@green.zimage.com) |
| 17:49.36 | *** join/#asterisk emrahpbx (i=emrah@sip.slimvoip.nl) |
| 17:50.41 | myselfhimself | in function Dial() |
| 17:50.46 | myselfhimself | what does the @ stand for ? |
| 17:53.10 | *** join/#asterisk murdock_ut (n=chatzill@mail.kimballequipment.com) |
| 17:53.17 | Qwell | myselfhimself: You're going to need to give a little more context. |
| 17:55.18 | rob0 | Frightened himself! |
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| 18:19.18 | *** join/#asterisk Ritzerisk (n=Ritztech@nv-65-40-156-46.sta.embarqhsd.net) |
| 18:19.34 | Ritzerisk | anyone know if asterisknow is embedded with asterisk gui 2.0 |
| 18:20.02 | hardwire | damnit |
| 18:20.06 | hardwire | where'd all the hawaii peeps go |
| 18:20.12 | hardwire | waits an hour or so. |
| 18:28.39 | [TK]D-Fender | Ritzerisk: look at the release dates |
| 18:31.28 | rue_mohr | the hwec arrived! |
| 18:32.02 | rue_mohr | I thought the tms320 was obsolete? |
| 18:33.22 | kannan | does the asteriosk +iaxmodem+hylafax still require spandsp lib and udptl set in sip.conf? |
| 18:33.37 | kannan | for sip to sip faxing |
| 18:34.40 | *** join/#asterisk ingenius (n=alektro@69.90.72.173) |
| 18:43.17 | eric2 | is there a quick and dirty way to have the phone go to vm when one is alredy on the phone instead of having it ring for the 20 seconds before going to vm? |
| 18:46.47 | rob0 | sounds like call waiting, try disabling it? |
| 18:48.42 | lmadsen | eric2: in sip.conf you could try setting the call-limit=1, or there is a dialplan application you could use to check the status (I can't remember off the top of my head as it's been so long, so I'm checking) |
| 18:49.02 | lmadsen | ChanIsAvail() |
| 18:49.18 | lmadsen | with option 's' |
| 18:49.57 | lmadsen | then you could check on the variable, and use a GotoIf($["${AVAILSTATUS}" = "BUSY"]?voicemail,s,1) or something like that |
| 18:50.13 | lmadsen | check on what ${AVAILSTATUS} actually returns... might be a number or something... I haven't used it in a while |
| 18:50.57 | *** join/#asterisk sasargen (n=chatzill@68-245-179-195.pools.spcsdns.net) |
| 18:55.57 | eric2 | ok, I'll look at that... tx |
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| 18:58.05 | rue_mohr | hmm ok |
| 18:58.12 | rue_mohr | it takes a second to train eh? |
| 18:58.28 | rue_mohr | should my audio cut out the incomming audio? |
| 18:58.34 | *** join/#asterisk CrazyTux (n=brandon@ip68-111-67-4.oc.oc.cox.net) |
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| 19:06.10 | TommyBJ | What is the CLI command to show how much CPU a codec conversion "costs"? |
| 19:08.00 | rue_mohr | ok, well |
| 19:08.13 | rue_mohr | make a few adjustments |
| 19:09.54 | TommyBJ | Never mind... Found it. For interested viewers, it's core show translation |
| 19:09.59 | *** join/#asterisk adr|an (n=xpl@unaffiliated/adrianxxx) |
| 19:10.51 | oh2gma | How can I enable debugging on 1.6.1-rc1? core set debug 1-10 doesn't give me any debug messages. |
| 19:11.42 | rue_mohr | you know, I never asked about paging, we dont need it... |
| 19:12.27 | DavidR2008 | is there a SIP expert (or at least more expert then I :-) ) that can answer the following question? In a REGISTER message is it ok for the From: header to be "name" <sip:user@> instead of "name" <sip:user@192.168.0.11> {for example} |
| 19:12.49 | rue_mohr | pretty sure the ip needs to be in there |
| 19:13.15 | *** join/#asterisk martyn-dev (n=admin@190.24.134.154) |
| 19:13.19 | martyn-dev | Hi |
| 19:13.35 | martyn-dev | I need update the date of a grandstream bt100 and bt200 .. how can I do it ? |
| 19:13.40 | martyn-dev | some help ? |
| 19:13.45 | DavidR2008 | ok, that's the only thing I've been able to find that is different between my softphone which registers and my aastra hardphone which doesn't |
| 19:14.52 | *** join/#asterisk joesuffceren (n=chatzill@96.14.29.74) |
| 19:18.33 | *** join/#asterisk Gopher_77 (n=Jim@cpe-71-72-19-206.neo.res.rr.com) |
| 19:18.38 | Gopher_77 | ~monkeys |
| 19:18.39 | jbot | This problem, like many others in the computer industry, can be solved by the application of monkeys. |
| 19:18.48 | Gopher_77 | ~monkey |
| 19:18.48 | jbot | This problem, like many others in the computer industry, can be solved by the application of monkeys. |
| 19:19.13 | Gopher_77 | anybody know the number for monkeys? |
| 19:19.31 | rob0 | 42 |
| 19:20.19 | Gopher_77 | or any other number leading to audio that will verify an SIP trunk connection |
| 19:20.26 | joesuffceren | my telco has recently started offering sip termination. I currently have a PRI with them and wanted to try their sip implementation to gauge quality and see if it would make sense for me to think about switching. they say that they "support asterisk as long as it is switchvox, but won't support open source asterisk." And, but"won't support" they don't just mean they won't help me configure... |
| 19:20.27 | *** join/#asterisk Ritzerisk (n=Ritztech@nv-65-40-156-46.sta.embarqhsd.net) |
| 19:20.28 | joesuffceren | ...it. they actually won't sell me sip because "a hacker could put malicious code in the source for asterisk and use our customers' systems to attack our network" |
| 19:20.54 | joesuffceren | any good references showing the similarities between asterisk and switchvox codebase |
| 19:21.14 | Gopher_77 | geez, they can inspect the code themselves |
| 19:21.39 | Ritzerisk | soo tkD i had to go into a vpn so it dropped my connection but anyways is the asterisk gui 2.0 not released on an iso or do i have too look up the yum install for the gui portion |
| 19:21.41 | Gopher_77 | or you can argue that you can |
| 19:22.17 | joesuffceren | yeah. I mean, switchvox runs on centos, correct? |
| 19:22.26 | joesuffceren | so there's still open-source-ness going on there. lol |
| 19:22.48 | Gopher_77 | you could argue that a hacker could slip something in switchvox code |
| 19:23.15 | rob0 | Wow, Joe, they're real smart. No one can slip malware into prorietary/closed source crap. Sony never happened. |
| 19:23.21 | Gopher_77 | or microsoft windows for that matter; but we know that would never happen ;) |
| 19:23.37 | Gopher_77 | exactly |
| 19:23.38 | [TK]D-Fender | Ritzerisk: You have to and LOOK at the release date of the ISO and the release date of the GUI version and COMPARE |
| 19:24.00 | Gopher_77 | just call back and see if you get somebody else who has some sense |
| 19:24.15 | Ritzerisk | k so would i look at the asterisk.org for taht one and the asterisknow.org for the iso .... |
| 19:24.19 | Ritzerisk | Thanks |
| 19:27.56 | hardwire | http://failblog.org/2009/02/04/verizon-math-fail/ |
| 19:28.33 | joesuffceren | I asked to have the project manager give me a call, so hopefully I can talk some sense into her |
| 19:33.05 | bmoraca | Ritzerisk: asterisk-gui is no longer included in asterisknow. they use freepbx as the default. you have to manually install it (yum install asterisk-gui or some such) |
| 19:35.04 | Ritzerisk | is it not a good gui .... |
| 19:35.17 | Ritzerisk | i just liked it because i saw that it does BLf functionality |
| 19:39.00 | Gopher_77 | When I try to connect to my SIP provider (voipuser) I always get a busy/congested message with status of 'CHANUNAVAIL'. I've never successfully made a call. Can someone point me closer to a possible solution? |
| 19:39.43 | *** join/#asterisk protocols (n=protocol@ip-88-153-196-33.unitymediagroup.de) |
| 19:39.53 | Gopher_77 | joesuffceren: I would have them call digium, since they sponsor the software ;) |
| 19:40.02 | rue_mohr | [TK]D-Fender, you answering today? |
| 19:40.10 | *** join/#asterisk ElSonico (n=tav@dondo.ampiainen.net) |
| 19:40.16 | Gopher_77 | rue_mohr: ~Hello |
| 19:40.23 | rue_mohr | hello |
| 19:40.28 | rue_mohr | how are you with polycom phones |
| 19:40.55 | Gopher_77 | rue_mohr: Me? I've used them, but never set up * for them |
| 19:41.34 | Gopher_77 | ~hello |
| 19:41.35 | jbot | Howdy Bub |
| 19:41.40 | Gopher_77 | there it is :) |
| 19:41.55 | rue_mohr | k, in my system you have to dial 2 digits to get a particular line, so, how can (or can I at all) have a speed dial append its digits to the current call |
| 19:42.56 | Gopher_77 | rue_mohr: to the current call? Doesn't that mean you'd change the configuration of * in the middle of the call? |
| 19:43.10 | Gopher_77 | rue_mohr: or do you mean transfer the call to a different line? |
| 19:43.30 | rue_mohr | no, I need to dial 2 digits to select the line, then I want to have the speed dial send more digits to the "call I'm already on" |
| 19:43.53 | rue_mohr | right now It tries to use the speed dial as a whole new call, which dosn't work cause the line needs to be selected first |
| 19:43.57 | pfn | it's surprising how many big companies use asterisk now |
| 19:44.31 | pfn | selecting the line... such a pita |
| 19:44.37 | pfn | I do that now (using appearances on the phone) |
| 19:44.50 | rue_mohr | 4 businesses in one office with 2 people working for all 4 |
| 19:44.56 | *** join/#asterisk infinity1 (n=brendon@li6-32.members.linode.com) |
| 19:45.11 | pfn | easiest thing is to use a sip phone with line appearances |
| 19:45.16 | infinity1 | do polycom 501's have paging functionality? |
| 19:45.22 | pfn | 4 line appearances will handle that just fine |
| 19:45.23 | rue_mohr | but I cant use appearances on the phone because no voip phone can do presence to say if the line is busy |
| 19:45.29 | Gopher_77 | rue_mohr: I haven't done this sort of thing with a polycom system before, but I would think that you could set one speed dial for the line and another for the number |
| 19:45.52 | [TK]D-Fender | rue_mohr: Somewhat |
| 19:45.54 | Gopher_77 | rue_mohr: or just put all the numbers in one speed dial, and have * parse it |
| 19:45.55 | rue_mohr | yea, I have speed dials with buddy watch for all the lines, so they can tell which ones are busy |
| 19:45.58 | pfn | rue_mohr, isn't there a hack in * for that? max channels or something on the SIP channel |
| 19:46.09 | pfn | so if you set maxchannels=1 then any calls more than 1 will result in busy? |
| 19:46.13 | pfn | or something like that |
| 19:46.14 | Gopher_77 | rue_mohr: never heard of buddy watch |
| 19:46.35 | [TK]D-Fender | Gopher_77: Presence |
| 19:46.38 | rue_mohr | but in this office the users have to select which line their going out on cause each has a different call display on the pots |
| 19:46.43 | pfn | rue_mohr, in any case, if you can't be bothered to figure it out, a dialplan using prefixes should work easily |
| 19:46.49 | Gopher_77 | [TK]D-Fender: haven't heard of that either |
| 19:47.01 | rue_mohr | the users have to select which line their call goes out on |
| 19:47.16 | rue_mohr | cause the call has to be made on the line for the business its relivent for |
| 19:47.22 | pfn | rue_mohr, _91NXXXXXX, _92NXXXXXX, _93NXXXXXX, _94NXXXXXX and have each send to a different line |
| 19:47.47 | rue_mohr | yea, but that means having them program 4 speeddials for each number they want to speeddial |
| 19:47.56 | [TK]D-Fender | pfn: He's whoring himself to key-system junkies |
| 19:47.59 | pfn | rue_mohr, use a macro |
| 19:48.09 | pfn | I guess |
| 19:48.10 | rue_mohr | [TK]D-Fender, I'm using your 'its not keyd |
| 19:48.12 | rue_mohr | system |
| 19:48.28 | rue_mohr | but... |
| 19:48.30 | pfn | oh, speeddial is on the phone itself |
| 19:48.31 | pfn | suck |
| 19:48.38 | pfn | use a phone with 4 actual lines then |
| 19:48.38 | rue_mohr | why can nobody understand this office |
| 19:48.56 | rue_mohr | I cant do that because prescence dosn't work |
| 19:48.56 | [TK]D-Fender | pfn: then he loses the PRESENCE info for occupancy <- |
| 19:49.10 | [TK]D-Fender | watches rue_mohr build a new house of cards.... |
| 19:49.18 | pfn | even on a phone with 4 pots lines or key-system type phones? |
| 19:49.23 | [TK]D-Fender | bangs the table |
| 19:49.32 | rue_mohr | the design of this office is killing me, I so look forward to doing an asterisk system for a normal office |
| 19:49.35 | Gopher_77 | lol |
| 19:49.35 | [TK]D-Fender | CRISIS! Bailout time! |
| 19:50.04 | rue_mohr | pfn, right now everyone has 4 pots phones and 4 call display modules on their desk |
| 19:50.06 | Gopher_77 | rue_mohr: so basically, they're trying to account for the calls that go out for each business |
| 19:50.40 | rue_mohr | yea, if they are making a call for olson electric, the obd development line cant be used |
| 19:50.51 | pfn | is confused why sip presence doesn't work |
| 19:51.06 | rue_mohr | cause the customer would get the wrong call display data |
| 19:51.29 | rue_mohr | prescence dosn't work on lines, it only works for watching other extensions |
| 19:51.52 | pfn | how is this a problem? |
| 19:52.09 | rue_mohr | so each dahdi line has an extension so that prescence can be used to tell if the line is busy |
| 19:52.37 | rue_mohr | there is only 1 line for each business, with 1 of the lines having distinctive ring for the other business |
| 19:52.45 | rue_mohr | your head spinning yet? |
| 19:52.57 | pfn | so you mean you want the handset to show if a given line is busy... |
| 19:53.00 | *** join/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 19:53.00 | *** mode/#asterisk [+o russellb] by ChanServ |
| 19:53.14 | pfn | that's kind of silly |
| 19:53.18 | [TK]D-Fender | pfn: Trust me... its complicated. You can't assign presence to a "line appearance line", only a speed-dial |
| 19:53.24 | rue_mohr | it has to, you cant have people have to try dialing their number 4 times if all the lines are busy |
| 19:53.42 | pfn | if you've got 2 people, add more lines :p |
| 19:53.47 | rue_mohr | and the speed dial dosn't seem to work with pre-started calls |
| 19:53.50 | [TK]D-Fender | pfn: I spent 2 weeks watch rue_mohr run in circles over it :) |
| 19:54.04 | Gopher_77 | rue_mohr: I see |
| 19:54.11 | rue_mohr | its 4 main desks, 4 businesses on 3 lines plus a fax line |
| 19:54.27 | rue_mohr | with everyone at each desk working for all 4 businesses |
| 19:54.58 | rue_mohr | no they wont get anymore lines |
| 19:55.29 | rue_mohr | no a T1 at $1000/mo isn't an option |
| 19:55.42 | Gopher_77 | rue_mohr: not very efficient |
| 19:55.45 | *** join/#asterisk adr|an (n=xpl@unaffiliated/adrianxxx) |
| 19:55.47 | rue_mohr | and a isdn isn't available |
| 19:56.06 | rue_mohr | (you cant belive how much I wish it were) |
| 19:56.07 | Gopher_77 | rue_mohr: could get an SIP provider and screw the lines |
| 19:56.14 | rue_mohr | not here |
| 19:56.40 | rue_mohr | also, our network provider isn't reliable enough for sip lines |
| 19:56.50 | Gopher_77 | rue_mohr: that sucks |
| 19:57.12 | rue_mohr | what I need, is for the speeddial to provide a way to enter a line |
| 19:57.23 | *** part/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 19:57.30 | Gopher_77 | rue_mohr: sounds like you need phone macros |
| 19:57.33 | rue_mohr | and the polycom 601 starts a new call when you use the speed dial |
| 19:57.35 | Gopher_77 | rue_mohr: but they probably don't exist |
| 19:59.57 | Gopher_77 | rue_mohr: so the presence doesn't work without speed dial? |
| 20:00.17 | rue_mohr | the only other thing I can think of is to have asterisk manage the speed dials somehow |
| 20:00.30 | rue_mohr | prescence dosnt work on lines |
| 20:00.32 | Gopher_77 | rue_mohr: define an extension for each commonly used number? |
| 20:00.44 | rue_mohr | just extensions, so I made the lines into extensions |
| 20:00.45 | Gopher_77 | rue_mohr: but each line has an extension right? |
| 20:00.54 | rue_mohr | yea |
| 20:02.21 | joesuffceren | http://pastebin.com/d101c3f21 <--my response to my telco if anyone is interested. hopefully I'm not reaching too far in this response... |
| 20:02.22 | [TK]D-Fender | rue_mohr: You cannot have a speed-dial act upon an active channel |
| 20:02.34 | [TK]D-Fender | rue_mohr: Aastra's can do this however |
| 20:02.37 | rue_mohr | its frustrating the problem is that the speed dials are too smart, they dont just blurt digits |
| 20:02.51 | rue_mohr | hmm ok |
| 20:03.17 | rue_mohr | I'm seriously looking at using aastra for 'the first client' |
| 20:03.28 | rue_mohr | their dumbness is their advantage |
| 20:03.30 | Gopher_77 | rue_mohr: the proof-of-concept? |
| 20:03.39 | rue_mohr | and the main office |
| 20:03.53 | rue_mohr | like I say right now everyone has 4 phones on their desk |
| 20:04.09 | Gopher_77 | rue_mohr: really excessive |
| 20:04.11 | rue_mohr | 1 of them is cordless |
| 20:04.21 | rue_mohr | other are mixed brands |
| 20:04.25 | Gopher_77 | rue_mohr: cool, so he can pass it around |
| 20:04.39 | Gopher_77 | :) |
| 20:04.43 | rue_mohr | no its a 4 set cordless |
| 20:04.51 | Gopher_77 | oh |
| 20:05.09 | *** join/#asterisk docelmo (n=vircuser@pool-70-110-114-243.lyn.east.verizon.net) |
| 20:05.11 | rue_mohr | [TK]D-Fender, I need ideas |
| 20:05.12 | Gopher_77 | is the cordless dumb? |
| 20:05.25 | rue_mohr | no, it can rings its buddies for 'transfers' |
| 20:05.36 | Gopher_77 | I mean for speed dial |
| 20:05.43 | rue_mohr | they dont have speed dial |
| 20:05.51 | Gopher_77 | even better :) |
| 20:06.00 | rue_mohr | the other desk phones have the speed dial |
| 20:06.09 | [TK]D-Fender | rue_mohr: make a web-panel for their speed-dials that picks the line they want to ID as |
| 20:06.20 | rue_mohr | nods |
| 20:06.33 | rue_mohr | for use on the phones? |
| 20:06.52 | rue_mohr | or from their pc's? |
| 20:06.53 | [TK]D-Fender | rue_mohr: On their PC. |
| 20:07.04 | [TK]D-Fender | rue_mohr: You could use the microbrowser if you wanted... |
| 20:07.08 | rue_mohr | k, how would that work |
| 20:07.10 | [TK]D-Fender | rue_mohr: more painful of course |
| 20:07.30 | rue_mohr | it would use meeting to make a call between the line and their phone? |
| 20:07.35 | [TK]D-Fender | rue_mohr: You would actually be well serverd to use a common back end and make a front-end for each |
| 20:07.56 | [TK]D-Fender | rue_mohr: "AMI originate" , "call file" <- |
| 20:08.15 | rue_mohr | takes a deep breath |
| 20:09.40 | rue_mohr | how about I just give each person on their speed dial list a voip set? :) |
| 20:09.57 | rue_mohr | ok, .... |
| 20:10.14 | rue_mohr | goes back to page 1 of the asterisk book |
| 20:10.35 | *** join/#asterisk docelm0 (n=vircuser@pool-151-199-175-28.lyn.east.verizon.net) |
| 20:10.53 | Gopher_77 | rue_mohr: the silver lining to a simple but imperfect solution would be that you have more phones to experiment with :) |
| 20:10.59 | *** join/#asterisk obnauticus (n=lol@about/windows/regular/obnauticus) |
| 20:11.10 | [TK]D-Fender | holds the book and points for rue_mohr to look closer, then slams it shut on his face |
| 20:11.13 | [TK]D-Fender | *WHAM* |
| 20:11.19 | [TK]D-Fender | TRABAJO |
| 20:11.32 | rue_mohr | be nice |
| 20:11.38 | [TK]D-Fender | channels a little more eppigy |
| 20:12.08 | rue_mohr | if you were doing this system, you would just insist they get a t1 wouldn't you? |
| 20:12.22 | [TK]D-Fender | rue_mohr: No... totally not cost effective |
| 20:12.41 | rue_mohr | well, I'd love to hear how you would do this office |
| 20:12.44 | [TK]D-Fender | rue_mohr: maybe get them on an ITSP instead |
| 20:12.59 | Gopher_77 | ~itsp |
| 20:12.59 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 20:13.11 | [TK]D-Fender | rue_mohr: the real problem you have is getting people to deal with the user interface |
| 20:13.26 | rue_mohr | there are none here, you might find one in vancouver, the data network here is really unstable though |
| 20:13.35 | [TK]D-Fender | rue_mohr: it isn't the hardware... its the constant ass-kissing and concessions being made |
| 20:13.56 | [TK]D-Fender | rue_mohr: Analog it is. The problem is your USERS |
| 20:13.57 | Gopher_77 | rue_mohr: do you use satellite? |
| 20:14.00 | rue_mohr | you back to having a system that dosn't work cause of isp problems |
| 20:14.09 | rue_mohr | hah, too much delay |
| 20:14.24 | Gopher_77 | rue_mohr: exactly |
| 20:15.01 | Gopher_77 | rue_mohr: maybe a reason to get a redundant internet provider |
| 20:15.10 | rue_mohr | there are none |
| 20:15.15 | [TK]D-Fender | Gopher_77: Diminshing returns |
| 20:15.33 | Gopher_77 | [TK]D-Fender: possibly, but they would save on 4 lines |
| 20:15.34 | [TK]D-Fender | rue_mohr: Analog is fine... its all this user interface crap. |
| 20:15.59 | Gopher_77 | rue_mohr: yep, the most difficult part of IT: users |
| 20:16.06 | rue_mohr | can we not have this system emulate 4 analog circuits? |
| 20:16.08 | [TK]D-Fender | Gopher_77: BS, how much do redundant internet connections / ISPs / etc add up when they have 4 lines? |
| 20:16.30 | *** join/#asterisk mog (n=mog@nat/digium/x-8426092ab8911dc5) |
| 20:16.31 | *** mode/#asterisk [+o mog] by ChanServ |
| 20:16.32 | [TK]D-Fender | rue_mohr: You already ahve the answer to that |
| 20:16.40 | [TK]D-Fender | rue_mohr: Yes... and it works SHITTY |
| 20:17.04 | rue_mohr | current answer is "no, asterisk cannot emulate 4 phone circuits" |
| 20:17.41 | rue_mohr | its all working fine, this is just a little hurtle, speed dial |
| 20:18.54 | *** join/#asterisk Greek-Boy (n=greek@41.222.89.77) |
| 20:19.16 | rue_mohr | think I could call polycom on this? see if they have a magic switch? |
| 20:19.20 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 20:20.10 | pfn | rue_mohr, you can use my same suggestion from earlier, _91..., _92, _93, etc for speed dials |
| 20:20.26 | pfn | rue_mohr, only program the speed dials into asterisk |
| 20:20.31 | rue_mohr | I alreayd have it accept its 25, 26, 27, 28 |
| 20:20.47 | pfn | so call 2501 for speed dial 1, 2502 for speed dial 2, etc. |
| 20:20.58 | pfn | use a web interface to manage speed dials, or an ivr |
| 20:21.03 | rue_mohr | and how are the users to know whats what? |
| 20:21.05 | rue_mohr | hmm |
| 20:21.09 | pfn | use a web interface to manage speed dials, or an ivr |
| 20:21.18 | Gopher_77 | ~ivr |
| 20:21.19 | jbot | i heard ivr is Interactive Voice Response |
| 20:21.24 | rue_mohr | no I got ya |
| 20:21.25 | pfn | 2601 would call the same speed dial on 2501, except on line 26 |
| 20:21.27 | rue_mohr | I'm thining |
| 20:21.49 | pfn | no more phone speed dials, which kinda sucks, but at least there's a web interface or ivr and it's somewhat simple |
| 20:22.16 | pfn | it could be made more advanced, as [TK]D-Fender said, use click-to-call on the web interface, even |
| 20:22.22 | Gopher_77 | and a central database of some sort so the users can store it there instead of sticky notes on the desk |
| 20:22.24 | pfn | for people that can't be bothered to punch in 4 digits |
| 20:22.26 | rue_mohr | almost use a messaging interface for it "please say your name and dial your extension" |
| 20:22.31 | icebrew54 | click2call is fucking sweet |
| 20:22.37 | icebrew54 | just as a random interjection... |
| 20:22.41 | rue_mohr | I'm working on the click to call thing |
| 20:22.47 | icebrew54 | using this nojeeclick 2 dial |
| 20:22.50 | icebrew54 | firefox extension.... |
| 20:22.54 | rue_mohr | wait is there already a system set up for that? |
| 20:22.55 | beek | who is sweet, and why is click2call fucking him/her? |
| 20:23.25 | [TK]D-Fender | rue_mohr: Better option : Your "line" SD's call DAHDI directly. STOP. Make an IVR with dialtone backgrounded where they can dial out or use a secondary SD |
| 20:23.26 | icebrew54 | rue_mohr: http://www.noojee.com.au/Page/NoojeeClick-Installation |
| 20:23.26 | Gopher_77 | lol |
| 20:23.26 | pfn | rue_mohr, dunno, but it can't be very hard to set up a dialplan for it |
| 20:23.47 | [TK]D-Fender | icebrew54: WAY wrong for him... |
| 20:23.56 | [TK]D-Fender | rue_mohr: See above |
| 20:24.08 | [TK]D-Fender | rue_mohr: And that will help with CDR's as well |
| 20:24.30 | icebrew54 | just sending him to the click2call page...virtually anyone who uses asterisk can find it useful |
| 20:24.35 | icebrew54 | well asterisk + firefox |
| 20:24.42 | icebrew54 | wrong advice applies I'm sure... |
| 20:24.56 | Gopher_77 | ~noojee |
| 20:25.16 | icebrew54 | works with our sugarcrm too which is nice |
| 20:25.43 | [TK]D-Fender | icebrew54: its a nice idea, but you don't understand how he has complicated things |
| 20:26.02 | rue_mohr | so ok, now I have 4? leads to follow? |
| 20:26.10 | Gopher_77 | want more? |
| 20:26.22 | rue_mohr | no I want lunch, back in a half hour |
| 20:26.26 | [TK]D-Fender | rue_mohr: Start with mine, its a freebie and fixes some immediate backdraws |
| 20:27.44 | icebrew54 | heh yeah my advice is prolly whack, I'm one of those idiots that started out using asterisk-gui and am regretting it |
| 20:28.06 | icebrew54 | I liked it at first, but now I'm yanking out what it's done and replacing with manual |
| 20:28.21 | icebrew54 | for a simple setup it's great, but I wanted call monitoring, fax2email, etc. |
| 20:30.15 | rue_mohr | [TK]D-Fender, so you say I do a virtual dialtone and ... or the manager interface with the call file? |
| 20:30.18 | *** join/#asterisk ^sandro^ (n=lskdjfls@67.55.0.18) |
| 20:30.51 | [TK]D-Fender | rue_mohr: no AMI, etc for this |
| 20:31.13 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 20:31.20 | [TK]D-Fender | rue_mohr: this replaces the "raw" tone from DAHDI with * generated where you can let them dial out while implenting SD's as well |
| 20:31.39 | Khratos | Ok, I know this is the wrong channel, but... here I go... I'm writting a class in php for the AMI interface, and Writting to the socket used to connect to AMI i get a sudden 'connection reset by peer', does anyone knows about some possible causes of this behavior ? |
| 20:31.55 | pfn | rue_mohr, no, the idea is that you have say 20 speed dial buttons on the phone, 4 will be dedicated to "selecting a line" |
| 20:31.56 | [TK]D-Fender | rue_mohr: so they can press the "line 3" SD, get a tone, enter #24 and dial entry 24 from their speed-dials |
| 20:32.02 | pfn | rue_mohr, the remaining 16 will dial the actual number |
| 20:32.21 | default23434 | hello.. i was wondering if someone could assist me for a sec.... i have noticed that having an ATA with two lines ( Eg. sipura 2100 ) registered sometimes I have a registration problem. Eg. Line 2 continues to register however Line 1 fails after say 2 days. not sure why it doens't keep trying but once I reboot the device it works properly again for a perid of time. Also I am running a realtime server and once I reload configuration i am able once aga |
| 20:35.04 | pfn | so you end up doing something like, exten => 25,1,Set(LINE=Zap/whatever); ...,WaitExten ... |
| 20:35.37 | [TK]D-Fender | pfn: Something like that. |
| 20:35.43 | pfn | yeah, something like that |
| 20:36.01 | pfn | you'll need to switch contexts and stuff to make what I say work right |
| 20:36.13 | pfn | maybe |
| 20:36.19 | [TK]D-Fender | pfn: Yes, optionally. |
| 20:36.34 | [TK]D-Fender | pfn: More if you want to make your dialplan based SD's overlappable, etc |
| 20:36.49 | [TK]D-Fender | pfn: So each div can have 1-100 for instance, etc |
| 20:37.00 | [TK]D-Fender | pfn: Makes the whoel process more brain-dead |
| 20:37.49 | *** join/#asterisk aksyn (n=aksyn@gw.na.nu) |
| 20:39.54 | *** join/#asterisk D0C5i5 (n=caldwell@cpe-71-67-162-81.neo.res.rr.com) |
| 20:43.42 | D0C5i5 | can someone help me get the terminology right/point me in the right direction? i'm just starting with asterisk... if I want to connect one vonage line (via the existing CPE) to an asterisk box, and then at a remote location have a PCI card in a computer that allows me to use a regular phone, what are those pieces of hardware called? (and/or maybe let me know a nice/inexpensive to get them?) :) |
| 20:49.01 | Gopher_77 | D0C5i5: you can look at some equipment at digium.com |
| 20:49.55 | Gopher_77 | D0C5i5: but it's not cheap |
| 20:53.45 | *** join/#asterisk macli (n=macli@nmc.brc.ubc.ca) |
| 20:54.02 | rue_mohr | D0C5i5, [TK]D-Fender heh, another issue was just brought up was that people cant see the digits they have dialed |
| 20:55.06 | D0C5i5 | Gopher_77: yea, that's where i started |
| 20:57.18 | D0C5i5 | i was hoping to get into something for under $200 |
| 20:58.15 | rue_mohr | tdm400 is $500 |
| 20:58.53 | [TK]D-Fender | rue_mohr: overkill, and quite wrong on price :) |
| 20:59.01 | rue_mohr | you can get a tdm100 cheap :) |
| 20:59.31 | [TK]D-Fender | D0C5i5: You should switch to a "softphone account', that that in DIRECT off of Vonage and jsut buy your own ATA. Cost = $50 |
| 20:59.48 | [TK]D-Fender | s/that that/get that/ |
| 21:00.54 | ruben23 | hi.. |
| 21:01.15 | ruben23 | anyone have idea on this error: http://pastebin.com/m5b224108 |
| 21:02.36 | Qwell | psps ax |
| 21:02.48 | rue_mohr | so I split the dialplan where they might dial # |
| 21:02.52 | Qwell | err |
| 21:03.14 | rue_mohr | otherwise use the 10 or 11 digits and send to the dahdi |
| 21:03.19 | [TK]D-Fender | ruben23: "No such extension/context" <- what part of this is not excruciatingly clear? |
| 21:03.56 | rue_mohr | it could say if the problem is a missing extension or context |
| 21:04.21 | *** join/#asterisk VoipForces (n=courchea@67.55.25.221) |
| 21:04.37 | *** join/#asterisk jsolis (n=jimmy@190.41.153.85) |
| 21:04.47 | frogonwheels | ruben23: show dialplan default |
| 21:04.57 | rue_mohr | I'm going to try one thing first and call polycom |
| 21:05.06 | VoipForces | Hi, I have a queue question. Anyone knows a way that a non-queue member can pickup a call from a queue. And I mean a single call. |
| 21:06.25 | jjshoe | VoipForces walk over to the queue phone and pick it up? ;) |
| 21:07.35 | frogonwheels | VoipForces: temporarily add them to the queu perhaps? |
| 21:07.39 | VoipForces | jjshoe: Not an option. |
| 21:08.42 | VoipForces | frogonwheels: well, would have to join for only 1 call then remove that member... |
| 21:10.43 | *** join/#asterisk seanmh (n=johndoe@abq-216-31-109-157.dsl.zianet.com) |
| 21:10.56 | rue_mohr | hmm one of the ladies jsut said they were getting echo again... |
| 21:10.59 | [TK]D-Fender | ruben23: Go look for "No such extension/context 912127775678@default" |
| 21:14.38 | ruben23 | [TK]D-Fender:thanks.... |
| 21:20.01 | *** join/#asterisk hfb (n=hfb@pool-96-247-109-136.lsanca.dsl-w.verizon.net) |
| 21:20.02 | VoipForces | rue_mohr: what telephony card and phone are u using? |
| 21:20.21 | rue_mohr | tdm400 |
| 21:20.38 | VoipForces | rue_mohr: no hardware echo canceler? |
| 21:20.42 | rue_mohr | 2 fxs ch 4 fxo channels with echo hardware |
| 21:21.05 | rue_mohr | it was a call of a few mins, she said that toward the end she was starting to get echo |
| 21:21.08 | frogonwheels | VoipForces: what v of * ? |
| 21:21.18 | VoipForces | frogonwheels: 1.4 |
| 21:21.28 | rue_mohr | Asterisk 1.4.22 |
| 21:21.37 | rue_mohr | erp |
| 21:21.48 | [TK]D-Fender | rue_mohr: TDM400 does not have HWEC |
| 21:22.04 | rue_mohr | hah its an 800! |
| 21:22.14 | VoipForces | rue_mohr: TK is right. Again. |
| 21:22.53 | *** part/#asterisk AndyML (n=quassel@pool-72-78-117-135.phlapa.fios.verizon.net) |
| 21:23.53 | VoipForces | frogonwheels: what I would like it something like Pickup, but instead of pickup a call from a ringing extension in the group, do a pickup of a queue caller... |
| 21:25.02 | frogonwheels | VoipForces: hmm.. only way I can think of (bearing in mind I'm no expert).. is to use AddQueueMember to add a local channel .. and to set up a MeetMe() .. and get the local channel to dial into the meetme. |
| 21:25.19 | VoipForces | rue_mohr: make sure that your rj-11 cables going to your patch panel are the shortest possible. |
| 21:25.22 | default23434 | question.. does someone understand ringback? |
| 21:25.34 | frogonwheels | VoipForces: .... then the local channel can also remove itself from the queue when it's called. |
| 21:25.42 | default23434 | i receive 180/sdp indicating early media ringback and the accompanying RTP packets. |
| 21:25.55 | frogonwheels | VoipForces: does that make sense? |
| 21:26.02 | default23434 | and then send 183/sdp indicating early media ringback but does not send any RTP packets with this ringback. |
| 21:26.09 | default23434 | why am i sending 183?? |
| 21:26.15 | VoipForces | frogonwheels: 1 sec on the phone |
| 21:26.26 | default23434 | the termination party stop the ring because it assumes i am providing it.. this is not the case |
| 21:26.33 | default23434 | i want to send back 180.. what do i haev to do? |
| 21:26.39 | default23434 | anone? |
| 21:28.09 | rue_mohr | hmm polycom support sucks too |
| 21:28.15 | rue_mohr | you know, I think polycom just sucks |
| 21:28.33 | rue_mohr | VoipForces, yea, its properly connected |
| 21:28.52 | [TK]D-Fender | rue_mohr: No.... its just you :) |
| 21:29.01 | [TK]D-Fender | ok, checkout time... later all |
| 21:29.15 | rue_mohr | looks like you have to go tot he distributor with all questions |
| 21:29.34 | rue_mohr | aastra was happy to answer all my questions |
| 21:29.51 | rue_mohr | polycom says get lost |
| 21:30.12 | *** join/#asterisk xacatecas (n=jkroon@dsl-240-175-28.telkomadsl.co.za) |
| 21:32.25 | VoipForces | frogonwheels: wow, will have to think about that one... |
| 21:32.57 | xacatecas | ok, i'm going to get shot for this, but i must try. Using a GS GWX4104 gateway (4-port FXO). Inbound (PSTN -> SIP) calls are quite happy, and quality is fine. However, when I try to place a call outbound it on some numbers connects me, and then shortly after creating the packet bridge asterisk receives a INVITE from the gateway for weird extensions (seems to be correlated, but not exactly always) the dialed number, at which point |
| 21:32.58 | xacatecas | asterisk rejects the INVITE and the RTP stream stops, causing an eventual hangup on the call due to no RTP traffic. |
| 21:33.01 | VoipForces | rue_mohr: how long are your cables going from your TMD800 to telephony provider patch panel? |
| 21:33.02 | xacatecas | any ideas what could be wrong? |
| 21:33.53 | frogonwheels | VoipForces: I know it seems rather convoluted.. but I've been mucking about with connecting two streams.. and you just can't quite do it yet... well you can but it's restricted. |
| 21:34.00 | VoipForces | xacatecas: reinvite=no maybe? |
| 21:34.10 | xacatecas | tries |
| 21:34.35 | xacatecas | voipforces reinvite=no or canreinvite=no ? |
| 21:35.00 | VoipForces | xacatecas: canreinvite is the 1.6 syntax I believe |
| 21:35.19 | xacatecas | ok, running 1.6 |
| 21:35.26 | xacatecas | either way, same problem. |
| 21:35.32 | VoipForces | xacatecas: and I think it's the reverse of reinvite... not sure. I would try it both ways (yes and no() to see. |
| 21:35.47 | VoipForces | xacatecas: you did a reload chan_sip ? |
| 21:35.53 | xacatecas | jip |
| 21:36.26 | xacatecas | ok, multi-homing sucks. |
| 21:36.35 | hardwire | xacatecas: what are you doing? |
| 21:36.48 | VoipForces | xacatecas: The my next step would be to check if you have the latest GS firmware for your device. |
| 21:37.14 | xacatecas | hardwire, i'm not on site, so the phone i've got is connecting to the switches public IP. |
| 21:37.26 | hardwire | what kind of multi-homing? |
| 21:37.35 | xacatecas | with canreinvite=yes it doesn't kill the call but the audio sucks. |
| 21:37.35 | hardwire | knows mh-fu |
| 21:38.01 | rue_mohr | nec makes phone systms eh? |
| 21:38.05 | hardwire | xacatecas: whats the dial string and flags for the GS to dial out on? |
| 21:38.11 | xacatecas | phone --LAN-- NAT GW --DSL-- inet --DSL-- asterisk --LAN-- GWX4104 |
| 21:38.16 | xacatecas | does that make sense? |
| 21:38.29 | hardwire | everything but phone and inet |
| 21:38.32 | hardwire | what are those |
| 21:38.34 | hardwire | <PROTECTED> |
| 21:38.38 | VoipForces | rue_mohr: yes they do or at least did |
| 21:38.51 | xacatecas | GS BT200 (silly piece of junk that happened to be lying around) |
| 21:39.00 | VoipForces | xacatecas: what codec are u using? |
| 21:39.07 | xacatecas | g729 right through. |
| 21:40.14 | toadonsurboard | I'm confused |
| 21:40.28 | xacatecas | ? |
| 21:40.36 | VoipForces | xacatecas: one thing is sure is that you can not use canreinvite=no as this tells asterisk to get out of the RTP path. |
| 21:40.53 | VoipForces | xacatecas: and in your case that kills your call. |
| 21:41.03 | toadonsurboard | wrong, it tells Asterisk to STAY in the RTP path |
| 21:41.58 | xacatecas | is with toad on this one. however, not too much experience with that. |
| 21:42.32 | VoipForces | xacatecas: yeah I got it reversed. the canreinvite is the inverse of reinvite... |
| 21:42.51 | xacatecas | np. |
| 21:43.17 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 21:43.53 | rue_mohr | the nec systems look pretty slick |
| 21:44.13 | xacatecas | http://pastebin.co.za/9356 <-- asterisk config for the sip account. |
| 21:44.22 | toadonsurfboard | canreinvite is just an option. if set to yes it allows the phones to reinvite so that their RTP streams don't go through the Asterisk server. If set to no as in the case of MeetMe it locks the phones from reinviting and keeps the RTP locked to the server. |
| 21:44.52 | xacatecas | it should thus not make a difference? |
| 21:45.57 | rue_mohr | did the people who made these phones ever use a phone before they designed these? |
| 21:46.07 | toadonsurfboard | depends on the nature of the call, from phone to phone on the same network why burden the server with handling the RTP media when the two phones can just manage it. With SIP to ZAP channel going to PSTN you don't have a choice. |
| 21:46.09 | rue_mohr | I swear I'm at 110% frustrated |
| 21:46.22 | toadonsurfboard | rue_mohr, what phones? |
| 21:46.23 | rue_mohr | if the echo can comes apart I think I'm gonna lose it |
| 21:46.53 | rue_mohr | the freaking $2000 in voip equip I bought to show how great asterisk is thats just ruining everything |
| 21:47.09 | xacatecas | rofl |
| 21:47.11 | toadonsurfboard | what brand? |
| 21:47.16 | toadonsurfboard | Grandstream? |
| 21:47.23 | rue_mohr | polycom is my problem |
| 21:47.38 | pfn | polycom is a problem? |
| 21:47.39 | pfn | boggles |
| 21:48.14 | xacatecas | not the easiest phones to initially work with and i bricked one, but that's about the only problems I've had. |
| 21:48.34 | pfn | still happily uses his 7960 |
| 21:48.42 | xacatecas | they worked well after initial struggles, and that was mainly due to me knowing _nothing_ about VoIP at the time. |
| 21:48.55 | toadonsurfboard | really? I've got 92 sip peers on my asterisk server right now. 85 of them are polycom phones, the rest are Linksys ATA's. not had a problem with Polycom. Got 2 port T1 PRI connecting to my PSTN and a Nortel PBX. |
| 21:49.34 | VoipForces | hates polycom, they are a bitch to configure. I will be staying with Aastra for phones |
| 21:49.50 | pfn | how are they a bitch to configure? they don't boot tftp? |
| 21:50.03 | xacatecas | ja ja. i just want to fix this GXW problem. then I'm off to bed. |
| 21:50.16 | toadonsurfboard | so the problem is probably not the equipment but the person blaming the equipment that configured it improperly because they didn't want to read through the documentation or just skimmed it and didn't understand it. |
| 21:50.17 | VoipForces | pfn, they do, but the digimap feature for one thing is a bitch I find. |
| 21:50.27 | pfn | no idea what that is |
| 21:50.37 | pfn | is very out of date on voip handsets |
| 21:50.49 | pfn | stopped looking once he started using the 7960 |
| 21:50.52 | toadonsurfboard | digitmap is the dialplan for the phone itself. it acts like a pattern matching filter |
| 21:50.54 | pfn | and we use the spa942 at work |
| 21:50.58 | VoipForces | pfn: basically allos the phone to only dial predefined digit paterns. |
| 21:51.02 | pfn | oh |
| 21:51.11 | pfn | like dialplan.xml for cisco |
| 21:51.17 | VoipForces | hates phones that think they are more intelligent than the PBX they are connected to. |
| 21:51.46 | VoipForces | I rather have the PBX do the intelligent work. |
| 21:52.00 | pfn | that's not a problem of the phone |
| 21:52.00 | pfn | the phone doesn't send numbers over to the pbx while it's dialing |
| 21:52.01 | toadonsurfboard | with great power comes great responsibility. if can't play with the big boys, stay home! |
| 21:52.27 | pfn | VoipForces, that's not the phone's fault |
| 21:53.11 | xacatecas | http://forums.grandstream.com/node/754 <-- this may explain my issues. |
| 21:53.20 | VoipForces | pfn: well, never had any issues with Aastra. Their sonfiguration files are very straight forward and complete, and their support is just great. |
| 21:53.50 | toadonsurfboard | ~gs |
| 21:53.51 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 21:54.05 | VoipForces | does not want to start a phone war BTW. :-) |
| 21:54.12 | toadonsurfboard | :-) |
| 21:54.26 | toadonsurfboard | "Kill them all! God will know his own!" |
| 21:56.02 | icel | Anyone interested in helping me with a dialplan and tweak an existing * setup? It would probably take a couple of hours. I can pay, let me know if you are interested. |
| 21:56.41 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 21:58.29 | rue_mohr | I need a break, there must be some 4/0 wire that needs to be pulled somewhere |
| 22:01.05 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 22:07.34 | rue_mohr | can I dial up the echo buffer on the hwec? |
| 22:07.45 | [TK]D-Fender | ? |
| 22:08.02 | rue_mohr | aka were still getting some echo |
| 22:08.12 | rue_mohr | how do i make it go away... |
| 22:08.46 | rue_mohr | well it was a call to the states |
| 22:08.57 | rue_mohr | maybe it'll only be a problem on long long distance calls |
| 22:09.10 | rue_mohr | delays are too long |
| 22:10.51 | rue_mohr | pfn your idea dosn't work cause you can press speed dials in succession on a polycom phone, every time you do a speed dial the phone starts a new call |
| 22:11.30 | rue_mohr | this wouldnt be an issue if they had prescence on line keys |
| 22:11.43 | rue_mohr | chases tail |
| 22:12.23 | rue_mohr | [TK]D-Fender, I have no reason to have it give them a fake dialtone |
| 22:12.58 | [TK]D-Fender | rue_mohr: You do. 2 of them. Gain CDR for tracking calls, second to allow for numbered speed-dials. |
| 22:13.53 | rue_mohr | [TK]D-Fender, why even have the line selected first? |
| 22:13.57 | [TK]D-Fender | rue_mohr: Lets add : more control over dial timeout, # manipulation, etc |
| 22:14.17 | rue_mohr | why not have the speed dial dial the outside number and the system wait for you to tell it what line you want after |
| 22:14.24 | *** join/#asterisk SparFux (n=raoul@f050022128.adsl.alicedsl.de) |
| 22:14.50 | rue_mohr | then process the pile and execute the call |
| 22:15.03 | [TK]D-Fender | rue_mohr: My speed dials are OFF HOOK with tone once you've SELECTED the line already |
| 22:15.21 | SparFux | Hi. When somebody calls without callerID and I route the call to an ISDN card with Dial() command, I will get the first MSN of the isdn installation as callerID. How can I have no callerID when there was no callerID transmitted in the initial call? |
| 22:16.34 | rue_mohr | but on this system as it stands you cant select a line without being in a call, but you cant exec speed dial..... arg, my head hurts |
| 22:17.06 | rue_mohr | i wish the damned speed dials on the polycom just sent digit strings |
| 22:18.20 | rue_mohr | I'm trying to work out how to implement what you said |
| 22:19.01 | *** join/#asterisk lore20 (n=lorenzo@unaffiliated/lore20) |
| 22:19.09 | lore20 | hello everybody |
| 22:19.17 | [TK]D-Fender | rue_mohr: Speed dial done through DIALPLAN not a damn LINE KEY |
| 22:19.27 | rue_mohr | a good analogy here is if a person were using speed dial to navigate an external ivr |
| 22:19.40 | [TK]D-Fender | rue_mohr: Punch "line 2", hear tone, DTM #13 for entry 13 |
| 22:19.43 | [TK]D-Fender | DTMF |
| 22:20.09 | rue_mohr | yea |
| 22:20.10 | [TK]D-Fender | rue_mohr: And you aren't using a Polycom SD for anything except starting a new call. |
| 22:20.26 | [TK]D-Fender | rue_mohr: Aastra can do in-line DTMF. Their use of soft-keys is Godly |
| 22:20.39 | rue_mohr | and I get to write a php /postgres webpage for managing speed dials |
| 22:21.02 | rue_mohr | aastra also has tech support you can just phone |
| 22:21.16 | rue_mohr | and they have more freely programmable keys |
| 22:21.19 | rue_mohr | and a better manual |
| 22:21.55 | lore20 | I have a network with one asterisk server and 5 sip client; everything works between client and from pstn to server; now i'm trying to pair my server with SIP Broker, i configured sipbroker peer in sip.conf, i forward any * entry to sipbroker peer in extension.conf, and now i'm trying to call SIP welcome number from a sip client, call is established correctly but i can't hear anything |
| 22:22.02 | [TK]D-Fender | rue_mohr: Dunno about the manual part.... |
| 22:22.05 | *** join/#asterisk talirk81 (i=434e2716@gateway/web/ajax/mibbit.com/x-6d5361d902cd6208) |
| 22:22.15 | rue_mohr | the aastra manual is 1200 pages, the polycom is like 300 |
| 22:22.34 | rue_mohr | the aastra web interface is better to |
| 22:22.54 | lore20 | i think i need to set port forwarding on my router... could you help me? |
| 22:22.57 | rue_mohr | its just a shame the aastras look like junk beside the polycom |
| 22:23.14 | talirk81 | Is there an agi command for playing a sound file similar to Background() |
| 22:24.51 | lore20 | anybody? |
| 22:24.54 | Gopher_77 | how do I set up * to connect to a softphone? |
| 22:25.12 | lore20 | Gopher_77: sip |
| 22:25.28 | talirk81 | stream file looks to be the onlything close, but its not exactly the same right? |
| 22:25.37 | Gopher_77 | rue_mohr: isn't the point to have only one telephone at each desk? |
| 22:25.57 | Gopher_77 | lore20: of course, but I don't have any login information or anything |
| 22:26.05 | [TK]D-Fender | losrQuick guess, server behind NAT? |
| 22:26.21 | [TK]D-Fender | lore20: Quick guess, server behind NAT? |
| 22:26.24 | lore20 | [TK]D-Fender: if you are talking with me.. yes |
| 22:26.30 | lore20 | i'm behind a router nat |
| 22:26.30 | [TK]D-Fender | ~sipnat |
| 22:26.31 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 22:26.32 | [TK]D-Fender | ^^^^^^^^^^^^^^ |
| 22:26.35 | [TK]D-Fender | lore20: read up |
| 22:26.44 | rue_mohr | Gopher_77, yea, I'm testing mostly on one desk |
| 22:26.55 | emrahpbx | hello all |
| 22:26.55 | lore20 | i already forwarded 5060 e rdp |
| 22:27.12 | [TK]D-Fender | lore20: takes a hell of a lot more than that. READ |
| 22:27.33 | [TK]D-Fender | talirk81: Yes, Stream File is pretty much "background" + more |
| 22:27.48 | lore20 | could i set externip to a dynamic dns? |
| 22:28.36 | Gopher_77 | ~softphone |
| 22:28.36 | jbot | [~softphone] A soft-phone is a program that lets your computer use a compatible audio interface (usually a speaker&mic, or headset), and optionally a webcam, to place VoIP calls with a given protocol (SIP/IAX/H.323/MGCP/etc). For links enter ~zoiper , ~xlite , ~twinkle , ~bria , ~eyebeam , ~ekiga |
| 22:30.49 | *** join/#asterisk harry_v (n=pcsuppor@S0106001d7e52cc78.vs.shawcable.net) |
| 22:31.07 | rue_mohr | [TK]D-Fender, all the outside lines start with 2, I can use that as a prefilter to look for a speeddial or not |
| 22:34.38 | rue_mohr | will asterisk be able to link fax calls though the dahdi card ok? |
| 22:34.47 | *** join/#asterisk McUrex (n=aurlov@aurlov.astel.ru) |
| 22:35.25 | *** join/#asterisk bird_of_Luck (n=melifaro@77.73.232.13) |
| 22:35.39 | [TK]D-Fender | rue_mohr: Doesn't let you select the LINE its going out unless you embed the # in it |
| 22:36.28 | rue_mohr | as an alternate stratagey? |
| 22:36.29 | *** part/#asterisk cheriff (n=davidm@58.96.27.155) |
| 22:36.31 | [TK]D-Fender | lore20: "externhost" + "externrefres" |
| 22:36.41 | [TK]D-Fender | lore20: "externhost" + "externrefresh" |
| 22:36.52 | lore20 | yes... i'm seeing it now |
| 22:38.51 | SparFux | If I Goto() a different context to standard extension, will the CallerID be dropped? |
| 22:39.53 | [TK]D-Fender | SparFux: no |
| 22:39.58 | *** join/#asterisk saftsack (n=oliver@g227066073.adsl.alicedsl.de) |
| 22:40.32 | SparFux | Ok. |
| 22:45.03 | rue_mohr | so if a line pool is full on a large asterisk system, you dont find out till after you dial all your digits? |
| 22:45.44 | [TK]D-Fender | rue_mohr: Holy crap drop the Norstar lingo! |
| 22:45.55 | [TK]D-Fender | rue_mohr: And what a psycho mess it is! |
| 22:46.20 | [TK]D-Fender | rue_mohr: Please be EXTREMELY clear about what is in control, at which point.. the PHONE, or ASTERISK |
| 22:46.26 | rue_mohr | so if all channels are occupied on a large asterisk system, you dont find out till after you dial all your digits? |
| 22:46.48 | [TK]D-Fender | rue_mohr: depends how you dial. |
| 22:46.52 | rue_mohr | standard implemenation |
| 22:47.00 | *** join/#asterisk path_ (n=path@pc-15-190-86-200.cm.vtr.net) |
| 22:47.03 | [TK]D-Fender | rue_mohr: NO SUCH THING |
| 22:47.11 | [TK]D-Fender | reaches for his ClueBat |
| 22:47.26 | [TK]D-Fender | rue_mohr: Push for that 3rd strike! |
| 22:47.27 | rue_mohr | well, as I understand a standard implementation the phone wont even connect till you have dialed all your digits, as per the dialplan |
| 22:47.46 | [TK]D-Fender | rue_mohr: WHOSE dialplan? |
| 22:47.51 | [TK]D-Fender | rue_mohr: YOU'RE OUT! |
| 22:47.57 | [TK]D-Fender | starts swinging |
| 22:48.03 | jaytee | he's whining about the Polycoms again |
| 22:48.04 | [TK]D-Fender | ~cluebat rue_mohr |
| 22:48.05 | jbot | ACTION pulls out a ClueBat (tm) and thwaps rue_mohr. |
| 22:48.08 | rue_mohr | the phones come with the (whatever its called north america unifed dialing plan) built in |
| 22:48.36 | rue_mohr | I'm trying to understand 'normal' so I can better understand how this is 'not normal' |
| 22:50.23 | jaytee | it's easy! normal is how other people configure their stuff. not normal is how you've done it! |
| 22:50.49 | [TK]D-Fender | ... |
| 22:50.50 | [TK]D-Fender | PWNED |
| 22:51.05 | rue_mohr | ok they use *98 to get to the voicemail, so that works |
| 22:51.34 | jaytee | weird, Linksys ATA's use *98 to do transfers |
| 22:51.47 | rue_mohr | there has to be a 'its desgned to work like this' it'd be nice if it were written somewhere |
| 22:52.05 | *** join/#asterisk joobie (n=joobie@mx01.anric.com.au) |
| 22:52.06 | jaytee | they call them manuals or at least they did back in the day |
| 22:52.24 | rue_mohr | there is onyl only manual for asterisk and its a book |
| 22:52.38 | jaytee | Polycom even has this thing called a SIP Admin Guide. Go figure! |
| 22:52.45 | rue_mohr | have it |
| 22:52.53 | rue_mohr | and the user guide |
| 22:53.03 | jaytee | oh, you mean "the book"? |
| 22:53.08 | jaytee | I have 3 copies |
| 22:53.12 | jaytee | in print |
| 22:53.19 | rue_mohr | which promptly ended one of my users trying to make a speed dial that brought down the system |
| 22:53.25 | jaytee | one right here and two at work |
| 22:53.40 | rue_mohr | casue the polycom phone is too smart and tries to start a new call every time you use the speed dial |
| 22:55.37 | rue_mohr | [TK]D-Fender, if I split it after the 2, use the 2nd digit to determine the line, and if the 3rd digit is a # then I go with speed dial, otherwise connect to dahdi, dump the 1 digit and pass over to let the rest of the digits fall through |
| 22:56.28 | rue_mohr | http://eds.dyndns.org/~ircjunk/not_public_dont_open/phonesys/asterisk/extensions.conf |
| 22:57.25 | rue_mohr | I'm gonna have to brush up on my extensions programming |
| 22:57.46 | rue_mohr | I have to go take a look at a installation for a new client, see ya tommorow |
| 23:00.59 | *** join/#asterisk path_ (n=path@pc-15-190-86-200.cm.vtr.net) |
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| 23:06.56 | *** mode/#asterisk [+o russellb] by ChanServ |
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| 23:31.24 | Gopher_77 | I'm trying to use voipuser for an SIP provider, but I keep getting a congestion response from the server. I've confirmed that voipuser registers, but I still get the congestion response. Here are my sip.conf and debug info from a ping: http://nopaste.com/p/axn1NdkQpb Can someone help me get calls through? |
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| 23:32.02 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
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| 23:35.12 | cesau2 | if cli> show odbc ==> "Connection 1: connected" -- and yet i still get "Realtime mapping for 'sippeers' found to engine 'odbc', but the engine is not available" -- where can i do next to debug the problem? |
| 23:35.13 | [TK]D-Fender | Gopher_77: Meaningless. Look at the SIP debug of a CALL. |
| 23:35.31 | [TK]D-Fender | cesau2: Go look at all of your ODBC configs |
| 23:36.21 | cesau2 | i can sqsh and isql to the database using the same dsn -- infact, i can see that im logged in on the sql server... |
| 23:36.39 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 23:36.41 | cesau2 | (from asterisk) |
| 23:37.01 | [TK]D-Fender | cesau2: PASTEBIN is your friend... |
| 23:37.17 | cesau2 | =) |
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| 23:40.28 | edibrac | i've been battling intermittant, random HDLC errors 2-3 times per day with my Digium TE121 T1 card for several months now... I switch to Sangoma and all looks good.. is there a technical explanation for this? |
| 23:40.39 | *** join/#asterisk StanManCan (n=stan_man@S010600195b3059b4.gv.shawcable.net) |
| 23:40.45 | StanManCan | I'm getting an error |
| 23:40.49 | edibrac | that somehow Samgoma A101 cards are more resilient against low-level errors of some sort |
| 23:40.57 | [TK]D-Fender | edibrac: Superior board design :) |
| 23:41.04 | StanManCan | Rejected connection attempt from **IP** request 'NUMBER@mycontext' does not excist |
| 23:41.08 | [TK]D-Fender | edibrac: And a common reason people pick Sangoma. |
| 23:41.58 | edibrac | [TK]D-Fender: i am a Believer now :) i was quite skeptical before.. I previously was thinking it might be just good marketing. |
| 23:41.59 | [TK]D-Fender | StanManCan: Then it likely doesn't |
| 23:42.21 | StanManCan | Fender: Well what do I need to change/fix ? |
| 23:42.38 | StanManCan | is it something in my iax.conf or extensions.conf |
| 23:43.07 | [TK]D-Fender | StanManCan: Yes. |
| 23:43.24 | StanManCan | Fender: which one. |
| 23:43.35 | [TK]D-Fender | StanManCan: Maybe one, maybe both. |
| 23:43.50 | StanManCan | Fender: aka. I'm on my own ? |
| 23:43.54 | [TK]D-Fender | StanManCan: Show an ACTUAL error, with ACTUAL configs and you'll get a DEFINITE answer. |
| 23:44.15 | [TK]D-Fender | StanManCan: this whole "vague" thing isn't going to get you very far |
| 23:44.59 | cesau2 | [TK]D-Fender: config @ http://pastebin.com/m1f857bed |
| 23:49.37 | ruben23 | hi any idea on what is SVN trunk..? |
| 23:50.29 | StanManCan | Fender: |
| 23:50.30 | StanManCan | ERROR = http://pastebin.com/d3967eab0 |
| 23:50.30 | StanManCan | iax.conf = http://pastebin.com/d237e2437 |
| 23:50.30 | StanManCan | extensions.conf = http://pastebin.com/d3bcc7651 |
| 23:52.40 | [TK]D-Fender | cesau2: dsn => Principal-Asterisk <- try "asterisk" |
| 23:52.54 | *** join/#asterisk voxter (n=voxter@S0106001c1025ca09.vc.shawcable.net) |
| 23:52.56 | cesau2 | will do, thanks! |
| 23:53.00 | [TK]D-Fender | cesau2: res_odbc.ini: <- This SHOULD be ".conf", not ".ini" |
| 23:53.14 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 23:53.50 | [TK]D-Fender | StanManCan: Well from what we can see.... nothing. |
| 23:53.54 | cesau2 | aye, it is, good catch (just a typo) |
| 23:54.07 | voxter | does anyone know a way to do more than 3 user conferencing on a polycom phone? |
| 23:54.17 | voxter | I feel like they added many user conferencing to recent firmware |
| 23:54.38 | [8none1] | voxter, you have to buy a license for that feature. |
| 23:54.49 | voxter | [8none1]: but it does work? |
| 23:54.50 | Gopher_77 | [TK]D-Fender: there isn't any debug for my SIP calls |
| 23:54.54 | [TK]D-Fender | voxter: Its a licensed add-on for the models that support it |
| 23:54.58 | StanManCan | Fender: what else do you want. those are my full iax.conf and extenions.conf _and_ errors |
| 23:55.09 | [8none1] | Just use a Asterisk MeetMe conference. |
| 23:55.20 | [TK]D-Fender | Gopher_77: Go place a call and pastebin the entire attempt. |
| 23:55.47 | voxter | This one client of mine is too inept to change from a key system and to transfer people they want into a conference room |
| 23:55.48 | [TK]D-Fender | StanManCan: Exact names & numbers matter and you're masking everything. I trust none of what you've shown. |
| 23:56.08 | voxter | I cant think of an easier way to suggest either call, blind xfer to meetme, or set up DID for the meetme. |
| 23:56.40 | [TK]D-Fender | voxter: Ineptitute is par for the course these days... |
| 23:56.58 | StanManCan | Fender: why would i want to provide you with my account numbers, phone numbers, user names and passwords |
| 23:57.07 | voxter | [TK]D-Fender: yep. and when you're dealing with movie production studios they dont take "re learn it" as an answer |
| 23:57.40 | [TK]D-Fender | StanManCan: Do think I need to know the passwords? No, THAT you can mask. Please show some intelligence here... |
| 23:58.26 | StanManCan | the actual accounts and numbers are redundant |
| 23:59.18 | [TK]D-Fender | StanManCan: We can dance around in circles forever on this, but until you should all of the exact info to match the full inbound request (which you SHOULD have provided with full IAX debug) you simply aren't going to get anywhere |
| 23:59.37 | [TK]D-Fender | StanManCan: Numbers matter.Something doesn't match and you're hiding the evidence |
| 23:59.41 | [TK]D-Fender | StanManCan: Not too bright |
