| 00:00.09 | *** join/#asterisk CodeWarrior_ (n=kleber@201-74-223-165-am.cpe.vivax.com.br) |
| 00:00.14 | ShamusNY | hi guys |
| 00:00.36 | ShamusNY | has anyone tried recording a conversation in wireshark? |
| 00:00.49 | ShamusNY | i have 99% of one, but i am missing the INVITE |
| 00:00.49 | hardwire | like.. can't he jsut remove waitexten? |
| 00:00.54 | hardwire | then loop? |
| 00:00.57 | sdaniels | Ahhhhh, i think i understand.... so just because someone is in a context doesnt mean that that is the only place it will try to match the regex, it goes though all contexts that are included? |
| 00:01.05 | ShamusNY | so wireshark won't decode the RTP/UDP stream |
| 00:01.07 | ShamusNY | any ideas? |
| 00:01.15 | ShamusNY | or suggestions for a channel that might help? |
| 00:01.15 | hardwire | sorry.. I know.. wrong tree right. |
| 00:01.18 | CodeWarrior_ | hello folks, I'm totally newbie about Asterisk, but I have many doubts about how to starting, is there any good soul that can introduce me ? |
| 00:01.20 | hardwire | boo |
| 00:01.33 | lesouvage | ~book |
| 00:01.34 | jbot | methinks book is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 00:01.42 | *** join/#asterisk ganeshix (n=chatzill@cpe-24-29-46-121.nycap.res.rr.com) |
| 00:01.56 | hardwire | sdaniels: add ||ivr after m |
| 00:01.57 | lesouvage | CodeWarrior: welcome to the crowd |
| 00:01.57 | hardwire | inn Background |
| 00:02.07 | aiksa[LV] | sdaniels: pretty close |
| 00:02.17 | CodeWarrior_ | lesouvage: hehehe |
| 00:02.32 | aiksa[LV] | it will wait to see if you wanted to "go further" |
| 00:02.54 | ManxPower | sdaniels: Your current dialplan would allow anyone that could get to the IVR to dialout your lines |
| 00:02.56 | aiksa[LV] | lets say you extensions 1,2,200,300,4 |
| 00:03.03 | ManxPower | this. is. bad. |
| 00:03.11 | aiksa[LV] | ManxPower: you spoild the triumph |
| 00:03.13 | *** join/#asterisk boolean12 (n=boolean1@tandem.uplinktel.com) |
| 00:03.20 | *** join/#asterisk Corydon76-dig (i=four@pdpc/supporter/bronze/Corydon76-home) |
| 00:03.20 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 00:03.20 | lesouvage | Codewarrior: google on "asterisk guru" and you will be pointed to a website with info that helps me a lot. |
| 00:03.22 | [TK]D-Fender | sdaniels: No |
| 00:03.27 | aiksa[LV] | Manx not neceseraly |
| 00:03.33 | aiksa[LV] | he includes ivr in phones |
| 00:04.03 | aiksa[LV] | if he included only ivr in some incomming context |
| 00:04.06 | sdaniels | adding the ||ivr made it work. |
| 00:04.09 | boolean12 | Does anyone know a good way to list all the realtime config families? |
| 00:04.13 | [TK]D-Fender | sdaniels: because you aren't in the [ivr] context. You are in [phones], from a functional standpoint imagine that as being a cope& paste of everything included into ONE mess. |
| 00:04.18 | hardwire | sdaniels: but instead of just doing that |
| 00:04.19 | *** join/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 00:04.19 | *** mode/#asterisk [+o mog] by ChanServ |
| 00:04.28 | hardwire | you should put extension 900 in phones |
| 00:04.37 | hardwire | and have it "goto" ivr,900,1 |
| 00:04.44 | CodeWarrior_ | lesouvage: I'll friend, but at this moment, let me start asking you about something: Do I need to have some extra hardware to try Asterisk, or just install it, setup it and voila, it's working ? |
| 00:05.06 | hardwire | sdaniels: any time you allow a user to dial an extension, you should isolate it as much as possible |
| 00:05.07 | ManxPower | http://www.fnords.org/~eric/dialplan-example-v2.txt |
| 00:05.11 | [TK]D-Fender | sdaniels: In fact your ivr should have been run off the "s" exten and not "900". In your IRV you can dial 900 over and over to loop indefinitely |
| 00:05.13 | ManxPower | that's off the top of my head. |
| 00:05.23 | aiksa[LV] | and the same later on in [incomming] context to make it public teh safe way |
| 00:05.38 | hardwire | [TK]D-Fender: thats a good tip |
| 00:05.41 | ShamusNY | anyone?. |
| 00:05.44 | aiksa[LV] | tzafrir_laptop: ping again. |
| 00:06.33 | sdaniels | [TK]D-Fender: I think I get it... |
| 00:06.51 | ManxPower | sdaniels: contexts (and this is what you have -- a context problem) are one of the hardest things to understand in Asterisk and maybe the most CRITICAL thing to understand. Contexts are the security mechanism of Asterisk. |
| 00:07.12 | sdaniels | [TK]D-Fender: Im only about a week or so into Asterisk. thanks for the info yall. |
| 00:07.13 | hardwire | aiksa[LV]: does xorcom have a support line? |
| 00:07.24 | aiksa[LV] | no just email |
| 00:07.32 | hardwire | are they gone for the day? |
| 00:07.37 | LiNeTuX_Home | aiksa: I know Xorcom. I might be able to help. |
| 00:07.50 | ManxPower | aiksa[LV]: naw, just holler tzanger!!! and he eventually materializes. |
| 00:07.59 | sdaniels | Im going to try and make this config right, I may pop back in to get a proof read of my config if yall dont mind. Thanks again for the info. |
| 00:08.01 | aiksa[LV] | :))) |
| 00:08.06 | [TK]D-Fender | ManxPower: you mean tzafrir_laptop IIRC... |
| 00:08.15 | [TK]D-Fender | sdaniels: np |
| 00:08.17 | ManxPower | yeah, him too! |
| 00:08.25 | aiksa[LV] | LiNeTuX_Home: ok lets see if you can help |
| 00:08.51 | aiksa[LV] | Distrib - Slack. |
| 00:08.57 | aiksa[LV] | 2.6.25.18-smp kernel |
| 00:09.13 | aiksa[LV] | trying to add Astribank BRI (8 ports) |
| 00:09.40 | aiksa[LV] | the bank works just fine from demo CD (so - no HW error). |
| 00:10.06 | [TK]D-Fender | sdaniels: What the hell, I'm feeling generous : http://pastebin.ca/1229815 |
| 00:10.14 | aiksa[LV] | As per Xorcom manual - I have built their distribution of bristuff |
| 00:10.25 | *** join/#asterisk InHisName (n=InHisNam@c-71-225-221-149.hsd1.pa.comcast.net) |
| 00:10.49 | aiksa[LV] | now - as far as i can see from lsusb and zaptel_hardware |
| 00:11.08 | aiksa[LV] | the chan.bank itself has been identified and even reflashed |
| 00:11.35 | *** join/#asterisk devilsoulblack (n=aandaluz@srv.ec-gye.internet.geainternacional.com) |
| 00:11.42 | devilsoulblack | hi |
| 00:11.45 | aiksa[LV] | ./zaptel_hardware returns "usb:001/002 xpp_usb+ e4e4:1142 Astribank-BRI FPGA-firmware |
| 00:12.16 | aiksa[LV] | the point is - i dont see the actual isdn2 spans |
| 00:12.26 | CodeWarrior_ | Do I need to have some extra hardware to try Asterisk, or just install it, setup it and voila, it's working ? |
| 00:12.47 | [TK]D-Fender | CodeWarrior_: No special hardware, just a PC running *NIX |
| 00:13.01 | aiksa[LV] | dmesg is full of NOTICE-xpp: XBUS-00: copy_pcm_tospan: non-existing address (00): RECEIVE PCM |
| 00:13.15 | CodeWarrior_ | but how can I make a call using asterisk ? trough Skype ? |
| 00:13.18 | jaytee | CodeWarrior_, a local network and a workstation you can run a softphone on at minimum |
| 00:13.28 | aiksa[LV] | and under /proc/xpp/XBUS-00 there aind XDB subfolder |
| 00:13.30 | jaytee | Skype no! |
| 00:13.38 | aiksa[LV] | LiNeTuX_Home: - any idea? |
| 00:13.53 | aiksa[LV] | CodeWarrior_: not yet |
| 00:13.54 | jaytee | ~softphones |
| 00:14.05 | jaytee | ~softphone |
| 00:14.06 | 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 |
| 00:14.06 | CodeWarrior_ | ok, no skype, but what other company can provide me the "phone line" ? |
| 00:14.22 | aiksa[LV] | CodeWarrior_: plenty of them |
| 00:14.31 | LiNeTuX_Home | aiksa[LV]: thinking... checking against my wiki notes... |
| 00:14.56 | hardwire | aiksa[LV]: whats /etc/zaptel.conf look like? |
| 00:15.12 | ManxPower | ~itsp |
| 00:15.13 | 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 |
| 00:15.30 | CodeWarrior_ | ah ok, so I have to had some company to provide me the VoIP line, right ? |
| 00:15.43 | jaytee | correct |
| 00:15.48 | hardwire | CodeWarrior_: some company has PSTN access.. and trunks that to VoIP signals and RTP |
| 00:15.55 | hardwire | for joo |
| 00:16.10 | CodeWarrior_ | ok, what are that E1 interfaces and so on ? |
| 00:16.29 | hardwire | CodeWarrior_: legacy inventions left over from egyptian pyramid times. |
| 00:16.34 | ManxPower | CodeWarrior_: now is the time for you to go read The Book |
| 00:16.36 | ManxPower | ~book |
| 00:16.36 | jbot | hmm... book is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 00:16.49 | jaytee | ~101 |
| 00:16.50 | jbot | well, 101 is Telephony 101, which is a good read if you're unfamiliar with traditional TDM telephony. You can download it at http://www.stromcarlson.com/docs/basics/NTtelephony101.pdf |
| 00:17.04 | ManxPower | jaytee: Your jbot-fu is strong. |
| 00:17.12 | hardwire | sdaniels: you make it sexy yet? |
| 00:17.14 | CodeWarrior_ | great ! I'll get the both books |
| 00:17.15 | jaytee | ManxPower, thanks. |
| 00:17.30 | LiNeTuX_Home | aiksa[LV]: do you have /proc/xpp/XBUS-00/summary |
| 00:17.46 | [TK]D-Fender | ~e1 |
| 00:17.47 | jbot | [~E1] E1 is the basic digital telephony circuit used everywhere except the US,Japan,Taiwan & Hong Kong (who use T1 and similar variants). E1 runs at 2.048Mbps. It can be an unstructured channel for data. It can be channelized, to provide 31 time slots of voice or data, each of 64kbps. Time slot 16 is used for D-Chan when used with PRI signalling. |
| 00:17.51 | jaytee | CodeWarrior_, start with the Asterisk book, use the other one for unfamiliar terms you may come across |
| 00:18.05 | unpaidbill | huh, chan_ooh323 appears to be a pain in the ass |
| 00:18.23 | unpaidbill | wont work with t38modem properly |
| 00:18.25 | hardwire | aiksa[LV]: you have an /etc/zaptel.conf.. right? |
| 00:18.26 | jaytee | major pain and not really worth it |
| 00:18.52 | hardwire | unpaidbill: you dropped the t38 bomb? |
| 00:19.04 | CodeWarrior_ | jaytee, ok I will, thanks ! |
| 00:19.10 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 00:20.09 | CodeWarrior_ | another question, just having a computer + asterisk and a VoIP provider, can I have an ACD ? |
| 00:20.30 | jaytee | CodeWarrior_, ACD as in queues? Yes |
| 00:20.32 | hardwire | sure |
| 00:20.37 | hardwire | runs away |
| 00:20.45 | unpaidbill | oops |
| 00:20.46 | unpaidbill | did i say that |
| 00:20.50 | CodeWarrior_ | very nice |
| 00:21.13 | unpaidbill | i meant t triple 8 |
| 00:21.25 | unpaidbill | my name is john connor and the only way to communicate with the future is through h323 |
| 00:22.10 | jaytee | unpaidbill, if you ever get that comm link working make sure to tell them to cut the red wire, not the blue one. |
| 00:22.48 | CodeWarrior_ | another question, how could I have telephones to use as ramals connected to asterisk ? special hardware ? |
| 00:22.54 | CodeWarrior_ | ATA ? |
| 00:23.36 | jaytee | what's a ramal? |
| 00:24.15 | unpaidbill | perhaps he meant jamals |
| 00:25.14 | aiksa[LV] | LiNeTuX_Home and hardwire sorry just got back. remote desktop session terminated out of the blue |
| 00:25.15 | CodeWarrior_ | internal lines connected to pbx system |
| 00:25.24 | CodeWarrior_ | to transfer calls |
| 00:25.51 | aiksa[LV] | hardwire: yes i have it of course |
| 00:25.57 | jaytee | internal lines? you mean analog phones as internal "extensions"? |
| 00:26.00 | [TK]D-Fender | CodeWarrior_: To use an analog phone with * : Linksys PAP2T-NA for example |
| 00:26.23 | aiksa[LV] | hardwire: if i add even a single span, ztcfg fails that it cannot initialize the device |
| 00:26.25 | CodeWarrior_ | ah right |
| 00:26.27 | [TK]D-Fender | CodeWarrior_: to use LINES, usually we use specialized PCI cards, or sometimes similar gatways |
| 00:26.45 | CodeWarrior_ | or voip phones, right ? |
| 00:27.10 | aiksa[LV] | LiNeTuX_Home: http://www.pastebin.ca/1229821 |
| 00:27.24 | jaytee | CodeWarrior_, the book explains FXO (for analog lines from your telco provider) and FXS for analog phones. |
| 00:27.42 | CodeWarrior_ | great |
| 00:28.07 | CodeWarrior_ | so, I think the basics are answered, right now I need to go to the books |
| 00:28.34 | CodeWarrior_ | thanks a lot guys, I hope soon I can have an Asterisk running fine ;) |
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| 00:29.22 | aiksa[LV] | LiNeTuX_Home: thats the output of XBUS-00/summary |
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| 00:35.14 | dennisharrison | anyone know what fd == -1 in astman_append, should not happen means ? |
| 00:38.35 | jaytee | dennisharrison, ask in asterisk-gui, it's an asterisk-gui specific error: http://lists.digium.com/pipermail/asterisk-gui/2007-September/000745.html |
| 00:38.56 | dennisharrison | jaytee, howdy, asterisk-gui is dead right now :( |
| 00:38.58 | dennisharrison | oh well :) |
| 00:39.00 | dennisharrison | freepbx it is! |
| 00:39.28 | aiksa[LV]_afk | ok. thats it; time to bed. |
| 00:39.45 | dennisharrison | goodnight |
| 00:40.02 | jaytee | it's bedtime in latvia but not here |
| 00:41.32 | jaytee | wonders if glucosamine-chondrotin actually works or if it's just a bunch of hype cuz he's starting to sound like a bowl of Rice Krispies right after you add the milk. |
| 00:41.46 | aiksa[LV]_afk | jaytee: i was just stating the fact, that its the high time to go to bet for me |
| 00:42.13 | jaytee | don't forget to say your prayers |
| 00:42.19 | *** join/#asterisk LiNeTuX (n=LiNeTuX@253.238.95.24.cfl.res.rr.com) |
| 00:42.29 | aiksa[LV]_afk | oh no |
| 00:42.33 | aiksa[LV]_afk | not yet :) |
| 00:42.38 | aiksa[LV]_afk | LiNeTuX: alive? |
| 00:42.56 | LiNeTuX | aiksa[LV]: sorry, battery died! Gotta give my daughter a bath... bbl... |
| 00:43.30 | aiksa[LV]_afk | ok. did you see the pastebin result? |
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| 00:46.24 | aiksa[LV]_afk | ok, LiNeTuX take care. I am off for a sleep now |
| 00:46.37 | aiksa[LV]_afk | 4hrs of sleep left for me:P |
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| 00:56.35 | CodeWarrior_ | bye folks, again, thanks for the answers... see you.. |
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| 01:34.11 | Arnav | hi.. my service provider just switched to freeswitch and my calls in asterisk don't work anymore, please help me... |
| 01:35.32 | Arnav | is the freeswitch better than asterisk??? |
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| 02:12.37 | jaytee | it's quiet...............too quiet |
| 02:13.05 | [TK]D-Fender | goes to hide the last of the bodies |
| 02:13.18 | jblack | That reminds me.... /me checks on dance of the dead |
| 02:13.26 | jaytee | hands [TK]D-Fender a bag of lime to help speed the decay |
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| 02:27.34 | coppice | reducing the time constant will speed the decay |
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| 02:29.19 | jaytee | so you advocate something that could potentially destabilize the space-time continuum? |
| 02:36.01 | coppice | no, I'm not at Cern |
| 02:42.38 | *** join/#asterisk osiris (n=osiris@c-71-205-27-88.hsd1.mi.comcast.net) |
| 02:44.13 | jblack | It's no fair. Why do the theoretical physicists get to end the world? |
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| 02:50.34 | jaytee | it won't be the LHC that ends the world. People forget about the "Butterfly Effect" or how a butterfly flapping it's wings on one side of the world kickstarts a chain reaction that creates a hurricane on the other side of the world. It'll start with the simplest of things, a delivery truck overturns on the freeway and the beer it is carrying never arrives at it's destination, a VFW Hall in New Jersey. From that little event things will start to u |
| 02:50.34 | jaytee | nravel. |
| 02:54.52 | coppice | jaytee Butterfly Effect was a terrible movie |
| 02:55.44 | jaytee | coppice, I was talking about the concept, not the movie. |
| 02:55.58 | coppice | duh! |
| 02:56.22 | jaytee | and yeah, the movie kinda sucked regardless of which ending you watched |
| 02:57.56 | coppice | katrina probably started with a burrito. the owner farted, and step by step the wind increased |
| 02:59.15 | *** join/#asterisk rcy (n=rcy@S01060002553240a8.vc.shawcable.net) |
| 02:59.20 | jaytee | anything's possible :-) |
| 02:59.27 | WimpMan | So that's the master plan wor WW3? Free Burritos for everyone? |
| 02:59.45 | coppice | Burritos - the wind of change |
| 03:04.32 | denon | haha |
| 03:04.34 | denon | that's just wrong coppice |
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| 03:06.53 | jblack | imagines what a burito made by CERN would be like. |
| 03:07.34 | WimpMan | More like a Donut. |
| 03:07.37 | jblack | I suppose it would have 11 layers, require liquid nitrogen, and be so massive that it would swallow the earth up in a black hole. |
| 03:08.10 | coppice | well donuts started out straight in china, so maybe a ring shaped burrito is the next step |
| 03:09.22 | WimpMan | did indeed wonder if a ring Burrito would be a good idea. |
| 03:09.46 | WimpMan | But I fear it wouldn't be fast food any more then. |
| 03:10.42 | jaytee | http://beyondrandom.com/wp-content/gallery/randompics-august/tape.jpg |
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| 03:10.54 | jblack | dance of the dead is great. |
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| 03:25.17 | Kate-o | Hey |
| 03:26.43 | Nugget | jaytee: http://macnugget.org/photos/tboicey_comics/welders |
| 03:27.36 | jaytee | hehe |
| 03:28.21 | mog | <PROTECTED> |
| 03:28.28 | Qwell | huh? |
| 03:28.37 | Qwell | d'oh |
| 03:29.03 | jaytee | jabber collision? |
| 03:29.06 | Qwell | that better? |
| 03:29.11 | mog | hopefully |
| 03:29.26 | Qwell | sorry, just brought my desktop back up.. not sure how laptop had /Home |
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| 03:30.23 | lolipops | I just installed asterisk 1.6 on my ubuntu box. /var/run/asterisk.ctl doesn't get created anymore. other than that, everything works fine. any pointers? |
| 03:30.39 | Qwell | lolipops: installed from a package? |
| 03:30.47 | lolipops | Qwell, no, from source. |
| 03:30.56 | Qwell | then that file should exist |
| 03:31.05 | jblack | on ubuntu, it should be /var/run/asterisk/... |
| 03:31.12 | Qwell | jblack: hence the question above. |
| 03:31.21 | lolipops | its not under /var/run/asterisk either. |
| 03:31.31 | lolipops | my understanding was that the socket was created at launch? |
| 03:31.37 | Qwell | it is |
| 03:31.52 | lolipops | well it seems that it's not happening right now... :/ |
| 03:32.01 | lolipops | i looked under both /var/run and /var/run/asterisk. |
| 03:32.14 | Qwell | have you verified that it's 1.6 running, and not some odd mix-n-match? |
| 03:32.33 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 03:32.42 | lolipops | asterisk -V returns Asterisk 1.6.0.1 |
| 03:33.00 | Qwell | that shows the version of the client you just launched |
| 03:38.03 | [TK]D-Fender | lolipops: How did you start *? can you confirm that its running? |
| 03:38.38 | lolipops | [TK]D-Fender, i started with some /etc/init.d script. it is running; i can place and receive calls. |
| 03:39.32 | [TK]D-Fender | lolipops: I'd verify the script you used. |
| 03:39.58 | lolipops | [TK]D-Fender, it worked as advertised with 1.4; should it make a difference? |
| 03:40.51 | [TK]D-Fender | lolipops: Try stopping * and starting it as a daemon yourself |
| 03:41.30 | lolipops | .. apparently it works as advertised as root, too. |
| 03:41.36 | lolipops | i guess its a permission issue or something.. |
| 03:42.04 | [TK]D-Fender | lolipops: Quite likely |
| 03:43.09 | lolipops | is there an easy way to tell asterisk to create the file somewhere else? |
| 03:44.27 | [TK]D-Fender | lolipops: you'd have to look inside your init scrip |
| 03:44.43 | lolipops | i see. well that makes sense. thanks. |
| 03:44.58 | lolipops | to you too, Qwell. |
| 03:56.40 | *** join/#asterisk ShamusNY (i=dolphin@pool-173-68-122-204.nycmny.fios.verizon.net) |
| 03:56.50 | *** part/#asterisk ShamusNY (i=dolphin@pool-173-68-122-204.nycmny.fios.verizon.net) |
| 03:56.53 | *** join/#asterisk ShamusNY (i=dolphin@pool-173-68-122-204.nycmny.fios.verizon.net) |
| 03:58.22 | ShamusNY | I tried recording a conversation in WireShark/Ethereal. Unfortunately, I have 99% of a the packets from a personal call (that I need to to hear) but, I am missing the INVITE message. The symptom is that Wireshark cannot "discover" the RTP/UDP Stream! Do you have any ideas or suggestions to solve this? |
| 04:13.44 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 04:18.55 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 04:28.20 | hesco | how much should I reload after editing voicemail.conf? |
| 04:28.33 | lolipops | [TK]D-Fender, I figured my problem. |
| 04:29.29 | lolipops | [TK]D-Fender, the /etc/asterisk/asterisk.conf that ships with ubuntu hardy is broken; it defines directories under [global] instead of [directories] |
| 04:30.34 | [TK]D-Fender | lolipops: And earlier you said you installed from source... |
| 04:30.55 | lolipops | [TK]D-Fender, over an old 1.4 install from ubuntu |
| 04:33.03 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
| 04:33.47 | hesco | I'm trying to register this server to a DID provider. Can anyone advise how I can inspect whether I've done that successfully from the console? Is there some show command which might help me here? |
| 04:34.36 | hesco | iax2 show registry, I guess might be what I'm looking for |
| 05:01.23 | drmessano | hmm |
| 05:08.02 | hesco | I'm pretty sure in my tests of a new inbound DID number, that I'm reaching the right server with my call. But the voicemail seems to be broken on the sample. When I tried to use extension 1234, it left me with dead air, on which I finally hung up. But then the console seems to have documented the interaction. What are the most obvious places to look for issues with the voice mail? |
| 05:08.33 | *** join/#asterisk SQLDarkly (n=nospam@p4-66.dsl.ecentral.com) |
| 05:09.08 | [TK]D-Fender | hesco: Look at whats actually going on in CLI |
| 05:10.06 | *** join/#asterisk BBHoss (n=bbhoss@c-68-62-170-33.hsd1.al.comcast.net) |
| 05:10.15 | BBHoss | ~book |
| 05:10.16 | jbot | extra, extra, read all about it, book is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 05:10.59 | murdock_ut | waiting for 3rd edition now |
| 05:13.43 | drmessano | hmm |
| 05:14.10 | drmessano | I keep getting a 100% hung system on 1.6.0SVN |
| 05:20.11 | drmessano | Question: |
| 05:21.27 | [TK]D-Fender | drmessano: yes.... that's "well hung" ;) |
| 05:21.43 | drmessano | <SIP/111-092a3908>AGI Rx << VERBOSE "ExtensionState: 0" 4 |
| 05:21.44 | drmessano | <PROTECTED> |
| 05:22.03 | drmessano | After a LOOOOOOONG pause when dialing a call, thats the next thing I get on the CLI doing a debug |
| 05:22.11 | drmessano | That appears to be what was being "waited on" |
| 05:22.31 | drmessano | After 12-15 seconds |
| 05:23.02 | drmessano | Happens with any extension.. |
| 05:23.38 | drmessano | Any thoughts on "ExtensionState" lookup? |
| 05:24.25 | [TK]D-Fender | drmessano: Look at the full debug, and examine where that occurs in the AGI |
| 05:24.59 | drmessano | where that occurs in the AGI <-- in the dialparties.agi itself? |
| 05:25.09 | hesco | ~paste |
| 05:25.10 | jbot | well, paste is http://rafb.net/paste/, or see also pb |
| 05:25.55 | drmessano | ~pb is better |
| 05:25.56 | jbot | ...but pb is already something else... |
| 05:25.59 | drmessano | ~pb |
| 05:26.00 | 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 |
| 05:28.00 | hesco | http://rafb.net/p/s9M07H70.html <-- I'm commenting out those out of freq sound files until I can get those fixed and trying again. |
| 05:28.09 | *** join/#asterisk DonX (i=don@another.lostserver.net) |
| 05:28.12 | SQLDarkly | I have 8 servers. 2 openSER 6 Asterisk 1.4.22(but considering 1.6) 1 ser server is redundant as well as 3 redundant asterisk boxes with QoS being handled by a cisco call manager. I am trying to find out a way to balance meetme across this cluster. Any ideas how I can acomplish this without a massive loading of modules or third party apps. |
| 05:28.47 | DonX | Hi all...Is it possible to play a sound (.wav file) over an established call. |
| 05:29.05 | SQLDarkly | Boxes are also using linux ha with nic bonding. So this configuration must have room for a disaster recovery plan |
| 05:29.25 | [TK]D-Fender | hesco: SCrew the console, use a soft-phone |
| 05:29.46 | SQLDarkly | Servers are HP Proliant DL360 g4p with SLES |
| 05:29.57 | [TK]D-Fender | DonX: Triggered how? |
| 05:30.07 | DonX | command line maybe? |
| 05:30.11 | DonX | I'm not sure |
| 05:30.50 | DonX | I know that I can drop a call into /var/spool/asterisk/... but havent found any way to play over an existing chan |
| 05:31.03 | SQLDarkly | its my test lab right now so its not in any sort of production at the moment. |
| 05:31.51 | SQLDarkly | things can break as I have version control handled... So If anyone has any ideas on meetme clustering / balancing I would appreciate you rinput |
| 05:32.09 | [TK]D-Fender | DonX: I was asking what would cause this to happen in your situation... |
| 05:32.15 | DonX | ohhh |
| 05:32.16 | DonX | sorry |
| 05:32.46 | DonX | It would be nice to send the equiv of a wall message to everyone on my system |
| 05:34.09 | [TK]D-Fender | DonX: huh? |
| 05:34.38 | DonX | On a unix box you can send a "wall" message to all active users on your system |
| 05:34.50 | [TK]D-Fender | DonX: And what does this magic term of yours mean? |
| 05:34.56 | DonX | it would be nice to be able to playa recording over all established calls in asterisk |
| 05:35.12 | hesco | ok, here is one which isolates the vm issue, I think: http://rafb.net/p/MKedJU13.html |
| 05:35.18 | [TK]D-Fender | DonX: Page + Chanspy w/ Whisper |
| 05:35.27 | DonX | okay thanks |
| 05:35.27 | DonX | :) |
| 05:35.31 | DonX | I'll look into those |
| 05:36.12 | [TK]D-Fender | hesco: I see no call to voicemail in there |
| 05:36.36 | [TK]D-Fender | hesco: And plenty of signs to stop using CHAN_OSS |
| 05:36.46 | [TK]D-Fender | (console) |
| 05:38.17 | hesco | I assume I don't need chan_oss, then. How would I turn it off? |
| 05:38.17 | WimpMan | Hmm. The idea of a "v-wall" looks interesting. Might be nice if e.g. you run out of power. But I'm not sure if I understand the page thing. |
| 05:39.11 | hesco | using the sample config supplied, I asked for extension 1234, which I though was suppose to transfer to voicemail. Did I miss something here? |
| 05:40.52 | hesco | And can you say more about what this call on console means? I see the logs there. I understand its a command line to try things out. I get the sense folks are interacting directly with calls there as well. What is that about? |
| 05:42.04 | [TK]D-Fender | hesco: Stop calling console/dsp |
| 05:42.08 | hesco | ok, I did no_load in modules.conf, reloading server now |
| 05:42.10 | [TK]D-Fender | hesco: And set up a soft-phone |
| 05:42.36 | [TK]D-Fender | hesco: Sample configs are useless trash. |
| 05:42.43 | hesco | This server is across town, not here. I access it by ssh. |
| 05:43.07 | [TK]D-Fender | hesco: So? |
| 05:43.20 | SQLDarkly | Also has anyone noticed when using realtime sip they do not show in the CLI but still register? Is this an asterisk bug or am I misssing something |
| 05:43.55 | hesco | so you are suggesting that I set up a softphone on my desktop, that somehow interacts with this server across town? |
| 05:49.37 | WimpMan | [TK]D-Fender: Could you give me a hint on how Page would be usefull for that wall-type thing? |
| 05:49.46 | hesco | [TK]D-Fender: can you pls point me to documentation for setting up this soft-phone you advise? I have ekiga and kphone installed locally. How do I get them registered with the server so they get the incoming calls? |
| 05:50.50 | [TK]D-Fender | WimpMan: spawn a script that will call page targeting every active channel. |
| 05:51.16 | [TK]D-Fender | hesco: setting up a basic SIP device is * 101. Go look at any of the million guides out there. |
| 05:51.44 | [TK]D-Fender | hesco: And they would get the incoming calls when you go and tell your dialplan to call them. |
| 05:51.47 | WimpMan | You can page active channels? /me takes a closer look... |
| 05:52.28 | [TK]D-Fender | WimpMan: page multiple local channels that will chanspy each non-local one in progress |
| 05:53.12 | WimpMan | *ding* Yes. That makes sense. |
| 05:58.14 | *** join/#asterisk freakazoid0223 (n=mattc@pool-71-242-207-169.phlapa.east.verizon.net) |
| 06:17.25 | [TK]D-Fender | checkout time. Later all |
| 06:21.00 | drmessano | I guess I am going back to 1.4 |
| 06:21.03 | drmessano | This is bullshit |
| 06:23.18 | ShamusNY | I tried recording a conversation in WireShark/Ethereal. Unfortunately, I have 99% of a the packets from a personal call (that I need to to hear) but, I am missing the INVITE message. The symptom is that Wireshark cannot "discover" the RTP/UDP Stream! Do you have any ideas or suggestions to solve this? |
| 06:25.22 | drmessano | Use the built in recording facilites next time and not wireshark? |
| 06:32.45 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 06:45.00 | *** join/#asterisk jicksta_ (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net) |
| 06:46.53 | *** join/#asterisk wiseguy_ (n=chivilis@infospalvos.lt) |
| 06:47.17 | wiseguy_ | hello |
| 06:47.30 | wiseguy_ | anybody using cisco bri with zaphfc? |
| 06:48.29 | Maliuta | no, but i think I use actual crisco oil to deep fry :) |
| 06:48.40 | wiseguy_ | :D |
| 06:48.49 | wiseguy_ | don't use oil |
| 06:48.53 | wiseguy_ | use gasoline :) |
| 06:52.14 | Maliuta | you mean gasolene |
| 06:52.45 | Maliuta | like you mean aluminium |
| 06:53.08 | Maliuta | people really need to learn to speak properer english |
| 06:53.47 | wiseguy_ | i don't speak english |
| 06:53.51 | wiseguy_ | i use globish |
| 06:53.54 | wiseguy_ | ;-) |
| 06:54.34 | Maliuta | what you speak trollish? |
| 06:54.45 | coppice | the proper English word is petrol |
| 06:55.54 | Maliuta | petroleum |
| 06:56.00 | Maliuta | distillate |
| 06:56.22 | Maliuta | so ... a troll and a flamer meet on a forum |
| 06:56.30 | Maliuta | it'd be funny but is true |
| 06:57.03 | Maliuta | coppice: and in proper english there is the term gasolene |
| 06:58.14 | wiseguy_ | ;) |
| 06:59.12 | coppice | coppice: in England the term gasoline is recognised, but only as a foreign word |
| 06:59.56 | coppice | though I believe in the 19th century the term gasoline was used for lighting fuel |
| 07:00.10 | Maliuta | coppice: yeah, a fuel for lamps |
| 07:00.18 | Maliuta | as it was/is here in .au |
| 07:00.41 | coppice | but its spelled gasoline, not gasolene |
| 07:01.00 | Maliuta | coppice: I think you'll find it's gasolene |
| 07:01.20 | coppice | I think I just checked a dictionary to avoid looking foolish :-) |
| 07:01.24 | Maliuta | it's how I learnt to spell it and what all my dictionaries say |
| 07:01.42 | Maliuta | coppice: an english dictionary or an american one? |
| 07:01.44 | *** join/#asterisk sudhir492 (n=sudhir@adsl-154-183-78.mco.bellsouth.net) |
| 07:01.49 | sudhir492 | Hi all |
| 07:01.59 | coppice | english. I don't speak american |
| 07:02.04 | Maliuta | is not high |
| 07:02.18 | Maliuta | um-er-ick-an |
| 07:02.50 | Maliuta | as it was famously done in a comedy sketch here (done by a yank who had his tv career in .au) |
| 07:02.55 | sudhir492 | I dont speak um-er-ick-an either. Just know a few words :-) |
| 07:03.08 | coppice | though the 16 volume OED has many spelling for many words, and will probably allow almost any misspelling you try :-) |
| 07:03.12 | sudhir492 | only write american |
| 07:03.18 | Maliuta | coppice: hardda taulk um-er-ick-an |
| 07:03.50 | sudhir492 | Anyone using asterisk queues here? |
| 07:04.49 | wiseguy_ | sudhir492: yes |
| 07:04.50 | wiseguy_ | ;-) |
| 07:06.50 | sudhir492 | There are three SIP phones which are shared by 8 agents. Is there a way that an agent logs in from an extension with his agent id, and becomes part of the queue and the reports will show which agent number took calls |
| 07:06.57 | Maliuta | we are all queued up for our turn on asterisk |
| 07:07.09 | ShamusNY | do you know of another app that can use pcap and output the conversaion? |
| 07:07.18 | ShamusNY | -> drmessano |
| 07:07.38 | wiseguy_ | sudhir492: yes, it is possible |
| 07:07.40 | Maliuta | ShamusNY: why do you want to do it with pcap? |
| 07:07.51 | sudhir492 | wiseguy_, how |
| 07:07.54 | Maliuta | ShamusNY: this is what the inbuilt recording features are for |
| 07:07.54 | wiseguy_ | sudhir492: just create agents, password |
| 07:08.06 | drmessano | Use the built in recording function |
| 07:08.06 | wiseguy_ | sudhir492: make them members of queue |
| 07:08.08 | drmessano | I said that |
| 07:08.12 | sudhir492 | ok, in agents.conf |
| 07:08.18 | Maliuta | ShamusNY: unless you are trying to tap other people conversations |
| 07:08.24 | wiseguy_ | wiseguy_: and make login in extensions.conf |
| 07:08.26 | ShamusNY | no |
| 07:08.29 | ShamusNY | i have vonage |
| 07:08.37 | drmessano | ROFL |
| 07:08.39 | ShamusNY | so i piped the vonage to my 2nd nic |
| 07:08.56 | ShamusNY | and recorded the traffic on the NIC to pcap |
| 07:08.57 | wiseguy_ | sudhir492: for statistics I use queuemetrics, I think it will be enough for you |
| 07:09.02 | Maliuta | ShamusNY: if you have a * box terminating the sip (or in the chain with a noreinvite) use * to record it |
| 07:09.03 | drmessano | Welcome to #asterisk, the home of Vonage support |
| 07:09.13 | wiseguy_ | ;-) |
| 07:09.15 | sudhir492 | wiseguy_, for login, use agentLogin? |
| 07:09.16 | Maliuta | vonwho? |
| 07:09.21 | ShamusNY | well it's just VoIP |
| 07:09.22 | drmessano | Exactly |
| 07:09.26 | *** join/#asterisk axisys (n=axisys@117.18.230.49) |
| 07:09.27 | ShamusNY | g721 |
| 07:09.43 | ShamusNY | if anyone is wellversed, this is a good time to show off |
| 07:09.53 | ShamusNY | can I forge a packet for INVITE into the stream>? |
| 07:09.58 | drmessano | LOL |
| 07:10.08 | ShamusNY | you arent going to help |
| 07:10.10 | ShamusNY | thanks |
| 07:10.26 | wiseguy_ | sudhir492: exten => 1001,1,AgentCallbackLogin(||${CALLERID(num)}@sip) |
| 07:10.33 | drmessano | No, this is a good time for us who know real VoIP to laugh out loud at someone try to con some help recording Vonage calls |
| 07:10.51 | sudhir492 | But I saw that AgentCallbackLogin is deprecated |
| 07:11.03 | Maliuta | ShamusNY: so this has what to do with *? |
| 07:11.18 | drmessano | Maliuta: It's all VoIP, yanno |
| 07:11.34 | drmessano | Maliuta: If we knew anything about VoIP, we would show off and help him |
| 07:11.48 | drmessano | THAT doesn't sound self serving |
| 07:11.49 | ShamusNY | it's a voip RTP/UDP conversation recorded via PCAP |
| 07:11.52 | ShamusNY | thats all it is |
| 07:11.59 | ShamusNY | yea i was baiting you |
| 07:12.03 | ShamusNY | but you were being condescending |
| 07:12.04 | sudhir492 | wiseguy_, also when the agent calls out from that extension, report will not have a record of his agent id. |
| 07:12.08 | drmessano | Oh, even better |
| 07:12.42 | ShamusNY | how about this: have you ever used pcap to record/diagnose a VoIP call? |
| 07:12.46 | drmessano | I wasn't being condescending.. that implies some level of passive aggression. I was being outright insulting. |
| 07:13.02 | wiseguy_ | sudhir492: for outbound calls I use queuemetrics make "virtual" outbound queue |
| 07:13.07 | ShamusNY | you're an ass, and I've seen tons of you over the years in expert channels |
| 07:13.10 | ShamusNY | thanks |
| 07:13.16 | drmessano | You're welcome |
| 07:13.36 | drmessano | Try Vonage support.. 1-VonageHelp |
| 07:13.55 | wiseguy_ | does anyone uses cisco router with zaphfc ? |
| 07:14.06 | ShamusNY | is there a vonage channel |
| 07:14.18 | ShamusNY | i would do fine if you just helped me find the right direction |
| 07:14.31 | ShamusNY | this is more about VoIP and SIP in general |
| 07:14.32 | drmessano | #google |
| 07:14.44 | WimpMan | wiseguy_: What's the relationship between cisco and zaphfc? |
| 07:14.59 | wiseguy_ | WimpMan: isdn bri |
| 07:15.51 | WimpMan | wiseguy_: Ok, but I guess you're not trying to install * on a cisco, are you? |
| 07:16.34 | wiseguy_ | WimpMan: no, really not. I want to peer cisco isdn interface with asterisk server with hfc isdn card |
| 07:17.36 | WimpMan | Talking what? Voice? ppp? |
| 07:17.46 | wiseguy_ | voice |
| 07:18.30 | sudhir492 | wiseguy_, thanks for your suggestion. I will look into queuemetrics. |
| 07:18.52 | sudhir492 | However, what to do about deprecated function. Is there any other way? |
| 07:19.34 | WimpMan | Ok, unfortunaletly my crystall ball is away for service. So do you try to connect them locally or what? |
| 07:19.59 | wiseguy_ | sudhir492: it should be alternative, check voip-info.org and I am sure you will find solution |
| 07:20.18 | wiseguy_ | WimpMan: yes, locally, via ISDN BRI |
| 07:21.03 | WimpMan | Ok, so now we know the setup. And what's the trouble? |
| 07:21.17 | wiseguy_ | L1 (physical layer) is active |
| 07:21.23 | wiseguy_ | but L2 and L3 fails |
| 07:21.36 | wiseguy_ | cisco is nt side |
| 07:21.43 | sudhir492 | thats what I am checking right now |
| 07:22.29 | WimpMan | Do they talk the same protocoll? Any debug output from either side? |
| 07:23.24 | WimpMan | Is it only me or does Cisco in NT mode sound scary to others as well? |
| 07:24.46 | wiseguy_ | :-) |
| 07:25.10 | wiseguy_ | hfc isdn card is passive, it doesnt sound good to make it NT |
| 07:25.19 | wiseguy_ | yes, WimpMan, i have debug information |
| 07:29.51 | WimpMan | Looks like the right time to get back to bed. |
| 07:33.54 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 07:38.51 | *** join/#asterisk Samrat (i=be348cee@gateway/web/ajax/mibbit.com/x-403ad68ee3403033) |
| 07:44.58 | cvnet | hi |
| 07:45.07 | cvnet | has anyone here installed a2billing ? |
| 07:55.39 | drmessano | THE DEVIL HAS |
| 07:55.40 | drmessano | ok no |
| 07:55.51 | Samrat | anyone here tried freeswitch? |
| 08:00.05 | drmessano | why |
| 08:00.30 | drmessano | There is a very nice #freeswitch channel which offers lots of helpful support |
| 08:00.40 | Samrat | just wonder |
| 08:08.41 | Samrat | drmessano: thanks |
| 08:09.59 | drmessano | no probs |
| 08:18.44 | mort_gib | Samrat: People here prefer vanilla, you will find all kinds of compilations of "usefull" combinations. |
| 08:19.30 | mort_gib | In my humble opinion they are really good, until you have to do x, where x is the function the companies behind didn't expect |
| 08:20.09 | mort_gib | -And if you "just run the CD" you end up not understanding what you are supporting |
| 08:20.34 | mort_gib | And that attitude should be left to the average MS Windows support technician |
| 08:21.42 | Samrat | mort_gib: ok |
| 08:22.05 | drmessano | What does that have to do with Freeswitch? |
| 08:22.28 | cvnet | has anyone here installed a2billing ? |
| 08:22.53 | mort_gib | drmessano: Get off Coffee |
| 08:23.11 | drmessano | He said freeswitch, you idiot, not freepbx |
| 08:23.17 | drmessano | Get off "stupid" |
| 08:25.20 | mort_gib | drmessano: piss off |
| 08:25.32 | drmessano | mort_gib: learn to read |
| 08:25.49 | mort_gib | <PROTECTED> |
| 08:31.17 | mvanbaak | mort_gib: be nice |
| 08:31.30 | mort_gib | Ok, sorry |
| 08:31.56 | mort_gib | Saturday, I'm in the office working and meeting clients in 20 minutes |
| 08:32.22 | mvanbaak | mort_gib: ugh, that sux |
| 08:32.26 | mvanbaak | it should be weekend |
| 08:32.39 | mort_gib | And hanging out in here I see the same shit, why do we need this crap: I wasn't being condescending.. that implies some level of passive aggression. I was being outright insulting. |
| 08:32.52 | mort_gib | Yes that sucks! |
| 08:33.16 | drmessano | It's called having a sense of humor, you stuck up, pompous jerk |
| 08:33.45 | drmessano | Stop taking IRC seriously and get some sun |
| 08:33.56 | mort_gib | Still, this was not a comment to you drmessano.... |
| 08:34.06 | drmessano | I dont care |
| 08:34.36 | *** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net) |
| 08:34.36 | mort_gib | You can quite happily piss off (sigh) |
| 08:35.00 | drmessano | You've told me that 3 times.... yet, not pissing off |
| 08:35.03 | drmessano | Maybe you should give up |
| 08:35.34 | drmessano | Fact is, he asked about Freeswitch.. I kindly directed him to the proper location, you went on some rant about vanilla asterisk |
| 08:35.40 | mvanbaak | shows drmessano a picture of this 'lady' I saw on the street yesterday and watches him running off |
| 08:35.43 | drmessano | Stop being so rude to people |
| 08:36.18 | mvanbaak | she was beyond ugly |
| 08:36.18 | drmessano | mvanbaak: Is she single? |
| 08:36.33 | drmessano | lol |
| 08:36.37 | mvanbaak | like, 250kg, 1.90m and growing a beard |
| 08:37.52 | drmessano | ouch |
| 08:38.19 | drmessano | Does she have a 220v fondue pot? |
| 08:40.37 | mvanbaak | it was not in her shoppingcart. |
| 08:40.52 | drmessano | HAW |
| 08:40.59 | mvanbaak | and I think everything she has was in that cart |
| 08:41.44 | mvanbaak | she had clothes (or something she likes to call clothes) and some booze in there but that's pretty much it |
| 08:42.49 | drmessano | Probably wild turkey |
| 08:46.11 | drmessano | Anyone know the latest spandsp that works with 1.6? |
| 08:46.23 | *** join/#asterisk axisys (n=axisys@bbgw10.bdcom.net) |
| 08:46.48 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 08:51.34 | *** join/#asterisk ibm2 (n=Administ@196.203.192.179) |
| 08:53.37 | ibm2 | it's possible to activate xmpp in my switchvox |
| 08:55.19 | mort_gib | drmessano: I mistook Freeswitch with one of the "compilations" your right, but did you ever have a look at it?? |
| 08:55.37 | drmessano | I know what Freeswitch is... |
| 08:55.47 | mvanbaak | drmessano: 0.5 |
| 08:55.58 | mvanbaak | drmessano: I think there's a patch in trunk to support 0.6 |
| 08:55.59 | drmessano | mvanbaak: TY, 0.6 is fail |
| 08:56.02 | drmessano | oh.. |
| 08:56.10 | mvanbaak | so 1.6.2 will suppot 0.6 |
| 08:56.14 | drmessano | I see |
| 08:56.20 | drmessano | Im ok with 0.5 |
| 08:56.33 | mort_gib | Did you try it out?? |
| 08:57.18 | mort_gib | Asterisk fork, yes but is it any good?? |
| 08:57.25 | drmessano | Its not an asterisk fork |
| 08:58.11 | *** join/#asterisk piparkuka (n=igor@fw.wan.co.il) |
| 08:59.17 | mort_gib | That's what this links says: http://www.freeswitch.org/node/117 |
| 09:00.23 | drmessano | Thats not at all what that says, and its not a fork |
| 09:01.23 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 09:01.33 | mort_gib | I know, but did you try it out?? |
| 09:01.59 | drmessano | A long time ago |
| 09:02.13 | mort_gib | -Any good?? |
| 09:03.30 | drmessano | Dunno.. I use Asterisk.. Looked interesting, I gave it a once over. Wasn't interested in learning another platform. I also have a problem with how the product is positioned and the ANTI-asterisk sentiment that seems to come from the project. |
| 09:04.48 | mort_gib | I have head little about the project, I'm just curious, which is why I started looking at * |
| 09:04.48 | drmessano | Generally I don't see a lot of "Freeswitch is great, use it!".. I see "Asterisk sucks ass, ZOMG DIGIUM R TEH MICROSUK ZOMG, use this instead!!!11!!!" |
| 09:05.11 | mort_gib | Ok, but it's NOT like * is perfect |
| 09:05.12 | drmessano | So I could care less about it at this point |
| 09:05.20 | drmessano | Nothing is perfect |
| 09:05.30 | mort_gib | :-) true |
| 09:05.59 | drmessano | But IMO, if Freeswitch was so great.. they could walk around exposing their giant hairy balls for all to see, not having to win people over who have problems with Asterisk |
| 09:06.12 | drmessano | That wreaks of "not good enough" |
| 09:06.32 | mort_gib | Like I have some 150+ * users on 8 different systems, and on ONE, only ONE has in frequent issues with dropped calls.... |
| 09:06.50 | mort_gib | Yeah, but the OpenSource camp is much like that |
| 09:07.26 | mort_gib | I use OpenBSD for some things, and it's really good, but damn those guys are slagging off EVERYBODY! |
| 09:07.45 | drmessano | I dont see anywhere on Postgresql's site proclaiming "100 reasons you should use this instead of MySQL because MySQL sucks so hard, lemmetellya" |
| 09:08.01 | drmessano | BSD users are zealots |
| 09:08.03 | drmessano | Thats different |
| 09:08.05 | mort_gib | Freeswitch actually has that?? |
| 09:08.14 | mort_gib | I know, I still use OpenBSD |
| 09:09.43 | drmessano | No, but finding wholly qualified claims of Freeswitches superiority without being stacked against something in Asterisk are just about nonexistant. I mean, I know "Asterisk" is the "ex-wife" to some of the devs.. But if you keep talking about how much of a bitch your ex-wife is, you're not gonnaa get a lot of dates |
| 09:10.55 | mort_gib | LOL, no that true! |
| 09:13.25 | *** join/#asterisk kerx (n=prepro@adsl-69-104-18-9.dsl.irvnca.pacbell.net) |
| 09:14.03 | kerx | Hi, is anyone around that knows about wholesale voip for broadcast dialing? |
| 09:14.54 | mort_gib | Kerx: like Spamming |
| 09:15.11 | kerx | Is it considered spamming? |
| 09:16.01 | mort_gib | Sounds a lot like spamming to me.... |
| 09:17.36 | kerx | Well, I won't spend time trying to explain otherwise to you |
| 09:17.47 | mort_gib | No don't :-) |
| 09:17.49 | kerx | I need VoiP minutes, do you know where I can go? |
| 09:18.08 | mort_gib | Well I use Voipon.co.uk |
| 09:18.16 | mort_gib | And others... |
| 09:20.11 | kerx | Ok, I'll check it out thanks |
| 09:20.38 | mort_gib | They accept IAX too, which is handy |
| 09:20.39 | *** join/#asterisk Provito (n=Provito@pdpc/supporter/sustaining/Provito) |
| 09:20.43 | drmessano | mvanbaak: 0.5 FTW |
| 09:20.44 | mort_gib | So you can use them in IAX Trunk mode |
| 09:26.03 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-ce802d8dbe24a108) |
| 09:30.16 | mvanbaak | drmessano: :) |
| 09:30.23 | mvanbaak | drmessano: OpenBSD FTW |
| 09:30.25 | mvanbaak | hides |
| 09:30.39 | drmessano | zealot |
| 09:31.05 | drmessano | I am going to write some code for 1.6 |
| 09:31.19 | drmessano | I wanna see if I can get app_fondue added to 1.6.3 |
| 09:31.43 | mvanbaak | gheh |
| 09:32.48 | ibm2 | can anyone tell me how i can activate xmpp standard in my asterisk |
| 09:35.20 | drmessano | What are you trying to do, ibm2? |
| 09:35.58 | ibm2 | i try to activate IM between 2 bria |
| 09:37.15 | drmessano | Bria uses XMPP? |
| 09:37.28 | ibm2 | yes |
| 09:37.36 | drmessano | You need an XMPP server then |
| 09:38.35 | mvanbaak | ejabberd or something like that |
| 09:38.35 | drmessano | http://www.igniterealtime.org |
| 09:38.35 | drmessano | Get openfire |
| 09:38.43 | mvanbaak | but beware, the asterisk plugin is buggy at the moment |
| 09:38.53 | drmessano | Yes |
| 09:38.55 | mvanbaak | cant get the phonemappings page |
| 09:39.14 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 09:39.14 | mvanbaak | if I add an asterisk server it wont show up |
| 09:39.14 | drmessano | With 1.6? |
| 09:39.16 | mvanbaak | but I do see it connecting to the asterisk AMI |
| 09:39.17 | drmessano | oh |
| 09:39.26 | drmessano | I have a patch for that |
| 09:39.33 | mvanbaak | where is it ? |
| 09:39.45 | drmessano | hang on |
| 09:41.28 | drmessano | http://www.2l2o.com/asterisk/asterisk-im-WORKING.jar |
| 09:41.45 | drmessano | Thats the latest release with the db scripts fixed per a forum post |
| 09:42.57 | ibm2 | yes i already installed openfire |
| 09:43.09 | mvanbaak | drmessano: cool, thanks |
| 09:43.10 | drmessano | ok, and? |
| 09:43.45 | ibm2 | but i have some problem with him |
| 09:44.06 | drmessano | If you cant get Bria talking to Openfire, you're wasting your time in here |
| 09:45.44 | contactdq | I need a dial plan that logs an agent in/out in 1.4. It would first prompt agent for agent no/pin. If agent was logged into that particular phone, it would log him out. If he was not logged into that phone, it would log agent out any extensions, and then log him into that phone. Anyone know where I could find something like this. I would be willing to pay a reasonable fee for this. It needs to be queuemetrics compatible and play the |
| 09:45.55 | contactdq | thanks |
| 09:46.36 | drmessano | play the thanks? |
| 09:46.44 | drmessano | I R not understand |
| 09:46.49 | drmessano | :( meh |
| 09:47.45 | contactdq | lol |
| 09:47.58 | contactdq | no, agent logged in, agent logged off audio files :-) |
| 10:05.37 | *** join/#asterisk qdk (n=qdk@87.54.6.235) |
| 10:31.32 | *** join/#asterisk Ericounet (n=Ericoune@ACaen-151-1-14-54.w86-215.abo.wanadoo.fr) |
| 10:36.18 | contactdq | anyone available to help figure out where I went wrong with a dialplan? |
| 10:36.55 | mvanbaak | just pastebin the dialplan and a CLI log of when it fails |
| 10:41.24 | contactdq | http://pastebin.com/m140dc812 |
| 10:42.31 | contactdq | Playing 'cannot-complete-as-dialed' in the CLI is all I get |
| 10:43.11 | BBHoss | contactdq: set debug 10 |
| 10:43.15 | BBHoss | set verbose 10 |
| 10:43.22 | contactdq | ok |
| 10:43.32 | BBHoss | then post cli, should be much more |
| 10:44.20 | contactdq | http://pastebin.com/m7a8f0436 |
| 10:44.50 | BBHoss | what are you dialing? 2001? |
| 10:44.51 | *** join/#asterisk var1 (n=james@92-236-96-224.cable.ubr20.edin.blueyonder.co.uk) |
| 10:45.28 | contactdq | yes |
| 10:45.34 | BBHoss | it looks like you're using freepbx, so i really can't tell you 100% |
| 10:46.09 | BBHoss | if you were using asterisk, you'd need to make sure that the [macro-custom-agent-inout] is included in the from-internal context |
| 10:46.48 | BBHoss | you might try making an custom extension with the dial string of LOCAL/2001@macro-custom-agent-inout |
| 10:47.01 | BBHoss | then it would work from internal |
| 10:47.01 | contactdq | ok |
| 10:47.04 | var1 | hello, I'm about to download and compile Asterisk 1.6.0.1 if there is any reason I shouldn't speak now. ;) |
| 10:47.19 | BBHoss | var1: oh no wait, no!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!! |
| 10:47.27 | BBHoss | var1: j/k its fine :) |
| 10:47.30 | var1 | lol |
| 10:47.43 | var1 | thx BBHoss |
| 10:47.48 | BBHoss | welcome |
| 10:48.36 | BBHoss | you should get hossterisk 2.0 though, 100% erlang code with high avaliability and clustering built right into it from the ground up, all with gen_server/1 |
| 10:49.00 | BBHoss | dialplans stored in mnesia etc |
| 10:49.25 | BBHoss | var1: don't you want a link!?!?! |
| 10:49.31 | var1 | yeah sure |
| 10:49.39 | *** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net) |
| 10:50.06 | *** join/#asterisk mvanbaak_ (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 10:50.07 | BBHoss | heh, its just a dream right now, do you take dreammagnet links? |
| 10:50.24 | var1 | I'm a bit overwhelmed with it all but I still take it in my stride :) |
| 10:50.40 | BBHoss | haha |
| 10:50.55 | BBHoss | contactdq: did that work? |
| 10:52.37 | contactdq | nope |
| 10:52.44 | contactdq | unfortuantely |
| 10:53.57 | contactdq | i'm trying to avoid agent_callbacklogin |
| 10:54.01 | BBHoss | what did that log say with the custom extension, remember you must dial the custom extension number rather than 2001, unless you set the extension to 2001 |
| 10:54.31 | BBHoss | well i really don't know jack shit about agents, much less in freepbx and its cronies |
| 10:57.02 | contactdq | what customn extension numbeR? |
| 10:57.21 | contactdq | sorry...i've always been able to bypass freepbx by using extensions_custom.conf etc |
| 10:57.34 | BBHoss | then why the hell use it in the first place |
| 10:57.45 | BBHoss | sucks anyways, too bloated |
| 10:58.32 | BBHoss | how do you expect to be able to get to 2001 in the macro that you posted? |
| 11:01.58 | contactdq | i don't know..... |
| 11:02.34 | BBHoss | contactdq: you either need a custom extension, or you need to configure a number you can dial that will drop you in the macro that you posted |
| 11:02.48 | contactdq | ok |
| 11:03.34 | contactdq | let me create a custom extension |
| 11:07.59 | *** join/#asterisk Dr-Linux|home (n=Nothing@119.63.130.34) |
| 11:08.38 | Dr-Linux|home | I want to discuss something about Queue application func... anybody ? |
| 11:12.32 | Dr-Linux|home | i want once a caller is waiting in the queue, moh should be played but once an agent/extension is chosen by the queue then caller should listen to ring. how i can do that? |
| 11:13.28 | BBHoss | goodnight all |
| 11:14.35 | Dr-Linux|home | good noon to me |
| 11:15.43 | BBHoss | well technically its goo 6:15AM for me |
| 11:20.58 | var1 | night BBHoss! |
| 11:39.20 | Dr-Linux|home | var1: around? |
| 11:40.51 | var1 | around where, not here it's 12:40pm . so it's good afternoon really. |
| 11:40.57 | *** join/#asterisk jks (n=jks@193.189.93.254) |
| 11:42.16 | var1 | I don't normally hang out in irc but I will stick *around* here until I can't take anymore ;) |
| 11:42.35 | var1 | that was a joke |
| 11:43.43 | slingr | lol |
| 11:44.02 | slingr | i still haven gone to bed |
| 11:44.09 | slingr | its 7:40am for m |
| 11:44.10 | slingr | e |
| 11:53.12 | *** join/#asterisk Vale-ICS (n=vale@user-54460330.lns1-c12.dsl.pol.co.uk) |
| 11:57.20 | *** join/#asterisk steliosk (n=Stelios@91.140.105.61) |
| 11:57.44 | *** join/#asterisk kerx (n=prepro@adsl-69-104-18-9.dsl.irvnca.pacbell.net) |
| 11:58.06 | kerx | hi, can somebody please refer to me a good documentation to modify asterisk + astguiclient/vicidial from a predictive dialer to a survey based auto dialer? |
| 12:30.59 | *** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net) |
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| 12:53.07 | Dr-Linux|home | i want once a caller is waiting in the queue, moh should be played but once an agent/extension is chosen by the queue then caller should listen to ring. how i can do that? |
| 12:54.45 | *** join/#asterisk coppice (n=chatzill@33.155.17.210.dyn.pacific.net.hk) |
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| 12:59.29 | *** join/#asterisk johnakabean (n=none@pool-72-82-106-153.nrflva.east.verizon.net) |
| 12:59.57 | johnakabean | anyone know of anyway to spoof a provider to get unlimited channels |
| 13:02.46 | johnakabean | come on, sip isn't that secure lol |
| 13:06.51 | johnakabean | ok, using Ulaw, how much bandwidth is needed up and down.......90 kbps?? |
| 13:10.20 | *** part/#asterisk johnakabean (n=none@pool-72-82-106-153.nrflva.east.verizon.net) |
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| 13:23.49 | *** join/#asterisk donnib (n=mihai@0x555281d0.adsl.cybercity.dk) |
| 13:24.04 | donnib | hi. what does 407 Proxy Authentication Required mean ? |
| 13:24.15 | donnib | i can't get inbound call to work |
| 13:29.30 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 13:33.59 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 13:36.40 | lmadsen | donnib: it means the other end needs to authenticate, and that 407 has the nonce required to generate the MD5 hash in the 2nd INVITE from the device |
| 13:36.49 | lmadsen | INVITE (without auth) ---> |
| 13:36.56 | lmadsen | <-- 407 Proxy Auth Required |
| 13:37.03 | lmadsen | INVITE (with authorization) --> |
| 13:37.12 | *** join/#asterisk [gnubie] (n=[gnubie]@cm99.omega113.maxonline.com.sg) |
| 13:37.44 | [gnubie] | waves |
| 13:39.41 | Dr-Linux|home | i want once a caller is waiting in the queue, moh should be played but once an agent/extension is chosen by the queue then caller should listen to ring. how i can do that? |
| 13:41.43 | [TK]D-Fender | Dr-Linux|home: this is documented in the sample configs... go read them... |
| 13:42.12 | Dr-Linux|home | [TK]D-Fender: tried but could't find |
| 13:42.35 | [TK]D-Fender | Dr-Linux|home: Try again. |
| 13:42.57 | Dr-Linux|home | [TK]D-Fender: lemme open WIKI again |
| 13:44.35 | [TK]D-Fender | Dr-Linux|home: "core show application queue" <- |
| 13:44.46 | [TK]D-Fender | Dr-Linux|home: WIKI is the last place to look. |
| 13:44.52 | Dr-Linux|home | ok |
| 13:46.24 | Dr-Linux|home | [TK]D-Fender: it says: 'r' -- ring instead of playing MOH |
| 13:47.30 | Dr-Linux|home | but that "rings" the queue from start unless call is assigned to an agent/ext but i want it should play music but once call is assigned to an agent/ext then I should start ringing |
| 14:02.28 | *** join/#asterisk Great_Anta_Baka (i=c4d1fb03@gateway/web/ajax/mibbit.com/x-9746e39e62a1d397) |
| 14:04.41 | *** part/#asterisk donnib (n=mihai@0x555281d0.adsl.cybercity.dk) |
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| 14:57.31 | *** join/#asterisk zchaos (n=none@CPE0080c828f609-CM0019474d28a8.cpe.net.cable.rogers.com) |
| 14:58.39 | zchaos | hey guys... i wanted to ask you what type of internet connection do i need to run a VOIP network? rogers has the following connections.... http://pastebin.com/m488ea235 |
| 15:02.18 | eliel | zchaos: it depends at least on the number of calls simultaneously that you want to pass throu the link and the codec that you will be using |
| 15:03.27 | [TK]D-Fender | zchaos: Express minimum, keeping in mind Rogers sucks.... |
| 15:08.04 | *** join/#asterisk Dr-Linux|home (n=Nothing@119.63.130.34) |
| 15:08.15 | zchaos | this ins't a call centre |
| 15:08.20 | zchaos | its only going to be a call here or there |
| 15:08.22 | zchaos | the odd call |
| 15:08.26 | zchaos | there wont be many calls at all.... |
| 15:09.05 | zchaos | that should change the answer |
| 15:11.29 | eliel | zchaos: there is a magnitude of 8 aprox. between using g729 (8kbps) and g711 (64kbps) so, it depends on your voip provider or the other end codec capabilities |
| 15:13.19 | [gnubie] | how do you usually interface or integrate existing legacy branded pabx to an asterisk box? |
| 15:13.22 | *** join/#asterisk MalMen (n=Mal@bl8-126-62.dsl.telepac.pt) |
| 15:13.29 | coppice | the bit rate ratio is more like 3:1 when you add in the overheads |
| 15:14.05 | MalMen | where can i find a list whith the best rated voip companys ? |
| 15:15.28 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
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| 15:18.47 | *** join/#asterisk edwin_quijada (n=macaruch@190.80.159.93) |
| 15:20.20 | edwin_quijada | i am trying to compile zaptel and I get an error |
| 15:20.30 | edwin_quijada | The configure script was just executed, so 'make' needs to be |
| 15:20.30 | edwin_quijada | **** restarted. |
| 15:20.45 | seanbright | did you run ./configure first? |
| 15:20.49 | edwin_quijada | and I get 1 error [config.status] |
| 15:20.55 | edwin_quijada | seanbright: yes |
| 15:21.11 | edwin_quijada | I get this when I do make menuselect or make |
| 15:21.33 | seanbright | what version of zaptel? |
| 15:21.35 | edwin_quijada | how can I unload a module ? |
| 15:21.41 | edwin_quijada | seanbright: 1.4.12.1 |
| 15:22.00 | edwin_quijada | i had a old version from zaptel running before |
| 15:22.49 | seanbright | can you pastebin your config.log? |
| 15:22.52 | seanbright | ~pb |
| 15:22.53 | 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 |
| 15:24.27 | edwin_quijada | seanbright:ok |
| 15:27.52 | *** join/#asterisk murdock_ut (n=chatzill@c-98-202-154-226.hsd1.ut.comcast.net) |
| 15:28.57 | edwin_quijada | http://pastebin.com/m270d067a |
| 15:31.27 | seanbright | what version of gcc do you have? |
| 15:32.36 | edwin_quijada | gcc version 4.1.2 20061115 (prerelease) (Debian 4.1.1-21) |
| 15:32.44 | seanbright | hm. strange. |
| 15:33.34 | edwin_quijada | what does mean? |
| 15:33.44 | seanbright | not sure. |
| 15:33.50 | edwin_quijada | it is the first time i get this |
| 15:33.56 | RypPn | try make clean ?? |
| 15:34.05 | RypPn | then ./configure |
| 15:34.10 | edwin_quijada | i did it |
| 15:34.14 | edwin_quijada | get the same |
| 15:34.35 | seanbright | make distclean |
| 15:34.40 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 15:35.03 | tzafrir_laptop | make clean? that voodoo is not needed |
| 15:35.12 | seanbright | annnnd here we go |
| 15:35.14 | tzafrir_laptop | just: ./configure; make |
| 15:35.23 | RypPn | # |
| 15:35.23 | RypPn | gcc: '-V' option must have argument |
| 15:35.40 | RypPn | maybe the Makefile is wrong and it should be gcc -v |
| 15:35.52 | seanbright | that's in configure, not the Makefile |
| 15:36.05 | RypPn | well, wherever it gets that from is wrong |
| 15:36.17 | edwin_quijada | tzafrir_laptop: i did it too |
| 15:36.39 | tzafrir_laptop | edwin_quijada, did the configure script end successfully? |
| 15:36.50 | edwin_quijada | tzafrir_laptop: yes |
| 15:37.03 | tzafrir_laptop | what's the output of: make |
| 15:37.13 | tzafrir_laptop | could you pastebin it? |
| 15:37.16 | edwin_quijada | ok |
| 15:37.22 | edwin_quijada | i reboot the pc |
| 15:37.28 | edwin_quijada | give me a second |
| 15:41.10 | edwin_quijada | did |
| 15:41.17 | edwin_quijada | i paste the last part |
| 15:41.24 | edwin_quijada | do u need everything? |
| 15:42.16 | seanbright | pastebin all the output |
| 15:43.04 | edwin_quijada | how can I unload all modules from asterisk? |
| 15:43.32 | seanbright | one thing at a time |
| 15:43.41 | seanbright | 'make' is not affected by what is running in asterisk |
| 15:43.50 | seanbright | just pastebin all the output when you run 'make' |
| 15:44.59 | edwin_quijada | seanbright: I solved the problem.. dont kill me!! |
| 15:45.07 | seanbright | i have no desire to kill you |
| 15:45.17 | edwin_quijada | just the date was in the past |
| 15:45.39 | edwin_quijada | so make launch error for the date .... |
| 15:45.55 | edwin_quijada | uhmmm... must say something about that?? |
| 15:46.07 | seanbright | yes... fix your system date |
| 15:46.08 | edwin_quijada | WWell, thks! everyone! |
| 15:46.11 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 15:46.26 | edwin_quijada | i fixed the date now |
| 15:46.32 | edwin_quijada | and compile fine!! |
| 15:46.49 | seanbright | super. |
| 15:56.54 | *** join/#asterisk `Sauron (n=sauron@dsl001-130-033.aus1.dsl.speakeasy.net) |
| 15:57.26 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 16:05.41 | *** join/#asterisk Thorn (n=thorn@unaffiliated/thorn) |
| 16:11.04 | *** join/#asterisk LARefugee (n=chatzill@c-76-104-191-194.hsd1.wa.comcast.net) |
| 16:13.00 | *** part/#asterisk LARefugee (n=chatzill@c-76-104-191-194.hsd1.wa.comcast.net) |
| 16:22.58 | *** join/#asterisk zuez (n=sf@catalyst.httpd.org) |
| 16:23.03 | *** join/#asterisk xacatecas (n=jkroon@196.46.173.212) |
| 16:23.37 | zuez | Hey folks, is there any particular reason the following is not logging the userfield portion of my CDR? (CDR writing to database works wonderful otherwise, and the Verbose statement I put verifies ${PHONE_ID} is being populated: http://pastie.textmate.org/295315 |
| 16:24.12 | xacatecas | hi all, hope all is well. astdb is persistent accross restarts, is there something similar which is not? |
| 16:26.06 | xacatecas | in particular I'm in need of some "storage" to store temporary presence information. |
| 16:26.19 | [netman] | global variables? |
| 16:26.53 | xacatecas | that might work. |
| 16:27.52 | xacatecas | is it possible to construct those variables names dynamically? in other words, if I have one channel variable that contains a string like "agent5", use that as an index into an array of sorts, or generate a variables name like "pres_agent5" from that? |
| 16:28.13 | [netman] | zuez: try using Set(CDR(userfield) in the h extension ... |
| 16:29.29 | zuez | so instead of exten => _., use exten => h, ? |
| 16:29.52 | [netman] | xacatecas: you can Set(pres_agent5,g) in any point of your dialplan |
| 16:30.21 | [netman] | yes zuez, that It was I said |
| 16:32.36 | xacatecas | netman, agent5 is in a variable, so I can do Set(pres_${foo},g) ? |
| 16:32.51 | *** join/#asterisk ManxPower (n=manxpowe@208.sub-75-200-201.myvzw.com) |
| 16:33.17 | zuez | [netman] that doesn't appear to do it either, oh well. |
| 16:33.50 | seanbright | zuez: CDR backend are you using? |
| 16:34.06 | zuez | seanbright: cdr_addon.mysql.so |
| 16:34.22 | zuez | Everything is getting written to properly except userfield |
| 16:34.26 | seanbright | what version? |
| 16:34.52 | zuez | seanbright: Asterisk 1.4.17~dfsg-2ubuntu1 |
| 16:35.13 | [netman] | zuez: for that works, you first have to customize a cdr.conf setting, I remember |
| 16:35.26 | seanbright | pastebin your cdr_mysql.conf file |
| 16:35.32 | seanbright | masking passwords where necessary |
| 16:35.38 | zuez | heh ok |
| 16:35.44 | [netman] | endbeforehexten=yes |
| 16:37.40 | ManxPower | Could you have CDR batching enabled? |
| 16:38.25 | seanbright | zuez: you need to add 'userfield=1' in the [global] section of cdr_mysql.conf |
| 16:38.37 | seanbright | zuez: that will fix you up. |
| 16:39.05 | zuez | seanbright: thanks man. Let me try that. is reloading just the cdr_addon_mysql.so module after modifying the config adequate for it to re-read the config into memory? |
| 16:39.16 | seanbright | zuez: yes |
| 16:40.39 | zuez | seanbright: That did it, thanks. |
| 16:40.44 | zuez | [netman]: Thanks for the help as well! |
| 16:40.50 | seanbright | zuez: no sweat. |
| 16:40.50 | [netman] | zuez: :) |
| 16:40.55 | [netman] | I forget that :( |
| 16:41.46 | zuez | I should update some of the wikis I've been finding by googling that don't make note of that, although it's probably expected of the user to read the sample configs and figure it out on their own, which I failed miserably at. :-) |
| 16:43.28 | *** join/#asterisk Grnd-Wire (i=user@75.147.178.170) |
| 16:43.46 | Grnd-Wire | good morning everyone! |
| 16:44.16 | seanbright | zuez: any expectation that the user will read anything is full hearty. |
| 16:44.25 | seanbright | err |
| 16:44.28 | seanbright | fool* |
| 16:46.01 | *** join/#asterisk redder86 (n=lee@gateway.howardsilvan.com) |
| 16:46.56 | ManxPower | zuez: see, if you had read the official docs none of this would have happened. |
| 16:47.24 | *** join/#asterisk loca|host (n=tux@196.203.53.221) |
| 16:47.27 | ManxPower | zuez: now is the time to go into your asterisk source dir, look in configs and doc |
| 16:47.44 | seanbright | and the one in -addons as well |
| 16:47.54 | ManxPower | that too! |
| 16:48.03 | Grnd-Wire | ooh, what did I miss? |
| 16:48.19 | seanbright | nothing |
| 16:48.40 | redder86 | After upgrading from zaptel-1.4.7 / asterisk-1.4.17 to zaptel-1.4.12.1 / asterisk-1.4.22 now all I get is a yellow alarm on my wcte1xxp. Any ideas? |
| 16:49.07 | Grnd-Wire | Did you update libpri to current as well? |
| 16:49.07 | ManxPower | redder86: upgrading should not cause a yellow alarm. |
| 16:49.31 | redder86 | Grnd-Wire: libpri 1.4.7 |
| 16:49.40 | Grnd-Wire | ok |
| 16:49.42 | ManxPower | redder86: I assume you are using zttool to see the card status? |
| 16:50.24 | redder86 | ManxPower: I was just saying what the CLI says from zap show status |
| 16:50.45 | ManxPower | redder86: use zttool and have asterisk stopped. |
| 16:51.03 | ManxPower | your card should go green if you just have zaptel loaded. |
| 16:52.00 | ManxPower | that way you eliminate any issue with Asterisk or libpri and you can concentrate on zaptel |
| 16:52.37 | redder86 | ok, I'm building zaptel with zttool now |
| 16:54.25 | *** join/#asterisk coppice (n=chatzill@33.155.17.210.dyn.pacific.net.hk) |
| 16:54.33 | redder86 | Alarms Span │ |
| 16:54.33 | redder86 | <PROTECTED> |
| 16:54.37 | redder86 | coppice: hey |
| 16:54.48 | redder86 | ManxPower: still yellow alarm |
| 16:54.52 | ManxPower | redder86: have you tried a total reboot? |
| 16:55.04 | redder86 | no, but I can do that now |
| 16:55.41 | ManxPower | redder86: before you do that put a copy of your /etc/zaptel.conf on pastebin.ca |
| 16:56.26 | ManxPower | redder86: are you physically near the box? |
| 16:56.30 | redder86 | while it's rebooting... in full-disclosure it *is* an OpenVox D110PG. zaptel is patched with their wcte1xxp.c changes, however |
| 16:56.44 | coppice | redder86: hi |
| 16:56.45 | redder86 | ManxPower: no, I'm 1000 miles away over an ocean. |
| 16:56.49 | ManxPower | Well thank you for wasting my time. |
| 16:57.02 | ManxPower | Contact your hardware vendor. |
| 16:57.37 | redder86 | ManxPower: ? |
| 16:58.45 | seanbright | i thought it was pretty clear... :) |
| 16:58.52 | ManxPower | redder86: I know nothing about that card, what patches it does, or how to diagnose problems with the card. Go contact your hardware vendor for support. |
| 16:59.24 | ManxPower | Now if you had a TE1xxP card, it would be a different thing. |
| 16:59.41 | redder86 | sure, I was just hoping that someone here was willing to help troubleshoot regardless of the hardware |
| 17:00.40 | Grnd-Wire | redder86: I would try and get someone to physically unplug your T1 cable, let it go red.. wait.. and plug it back in.. |
| 17:00.53 | Grnd-Wire | You can't do that over software. |
| 17:01.07 | ManxPower | Grnd-Wire: I was going to suggest that if he was close to the system |
| 17:01.30 | Grnd-Wire | ManxPower: yeah - but you can find anyone who will follow instructions to do it.. Remote control monkey. :P |
| 17:01.35 | redder86 | Grnd-Wire: I did that yesterday when someone was on-site |
| 17:01.48 | Grnd-Wire | redder86:ok, well then I am out of ideas |
| 17:02.31 | ManxPower | redder86: in the future mention you have a non-Digium, non-Sangoma card right at the start. |
| 17:02.53 | seanbright | that way people can ignore you right from the get-go ;) |
| 17:02.59 | seanbright | (kidding) |
| 17:03.00 | ManxPower | Exactly! |
| 17:03.05 | Grnd-Wire | ManxPower: err.. Don't forget to mention Rhino |
| 17:03.29 | ManxPower | seanbright: I'm not aware of even a single person using that card. I doubt anyone here can help him other than the very generic stuff. |
| 17:04.11 | seanbright | points to redder86 |
| 17:04.15 | seanbright | there's a single person |
| 17:04.19 | seanbright | :) |
| 17:04.20 | Grnd-Wire | laughs |
| 17:04.23 | redder86 | I've used the card in a number of places. |
| 17:04.39 | redder86 | I also have used many Sangoma cards |
| 17:04.46 | redder86 | I've also used many Digium cards |
| 17:04.58 | redder86 | I tend to be fairly open with what hardware I'll use. |
| 17:05.16 | redder86 | I have had good support from all hardware vendors, including OpenVOX, it's just outside business hours right now. |
| 17:05.23 | Grnd-Wire | Once I find what works, I don't deviate.. |
| 17:05.39 | seanbright | clone hardware + custom patches = virtually impossible to diagnose what the problem is |
| 17:06.23 | ManxPower | I'm here to answer *easy* questions, not here to answer difficult ones. I answer difficult questions for money. |
| 17:06.29 | drmessano | I was gonna buy a nxtvox card to play with.. the 4 port and 8 port base card are the same price... difference is the 8 has a patched Zaptel |
| 17:06.35 | drmessano | Not worth it |
| 17:06.40 | redder86 | seanbright: if you were to look at the patches you'd see that it's fairly straight-forward ... not much to be afraid of |
| 17:06.48 | ManxPower | I pick a card vendor and stick with it. Easier to keep spares around for one thing. |
| 17:07.02 | seanbright | redder86: i'm not afraid |
| 17:07.10 | drmessano | Besides, if you're using 8 analog lines, you're doing it wrong |
| 17:07.20 | Grnd-Wire | seanbright: oh yes you are - you're shaking in your boots |
| 17:07.25 | seanbright | maybe a little |
| 17:07.31 | drmessano | seanbright: What is your name? |
| 17:07.34 | drmessano | seanbright: What is your quest? |
| 17:07.35 | redder86 | if I could standardize on Sangoma I would... however, some customers just can't stomach the price .... and I have to support hardware decisions made years and years ago that maybe I disagree with now |
| 17:07.54 | ManxPower | redder86: then it sucks to be you. |
| 17:08.03 | drmessano | seanbright: What is the zaptel timing ration of a disconnected TE100P? |
| 17:08.05 | ManxPower | In any case, BEST of luck with your card. |
| 17:08.09 | Grnd-Wire | drmessano: I will never again use FXO on a new phone system. Too much risk of echo.. Gotta love the 4wire interface. :) |
| 17:08.32 | drmessano | lol |
| 17:08.35 | ManxPower | Grnd-Wire: you realize echo comes from the FAR end analog loop, right? |
| 17:09.04 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 17:09.07 | redder86 | ManxPower: I'm not afraid of spending money for support either... but your hostility is a total deal breaker. |
| 17:09.29 | drmessano | ManxPower: I think the point was you should have spent money on a better card |
| 17:09.32 | drmessano | Errr |
| 17:09.32 | ManxPower | redder86: I'd not take money from you anyway -- I'd not be able to do a good job of helping you. |
| 17:09.36 | seanbright | that's not hostility... it's charm |
| 17:09.40 | drmessano | redder86: I think the point was you should have spent money on a better card |
| 17:09.45 | Grnd-Wire | ManxPower: err.. In the cases that I've had the problem, it appeared to be an impedance mismatch with my local loop. I had 6 lines, and each one performed differently. |
| 17:10.00 | ManxPower | drmessano: No, my point is that if you use a non-Digium/non-Sangoma card then you should not expect the community to be ABLE to help you. |
| 17:10.17 | drmessano | Same difference lol |
| 17:10.17 | ManxPower | Grnd-Wire: that is a possible cause of echo I guess. |
| 17:10.22 | redder86 | sigh |
| 17:11.05 | redder86 | nevermind. I honestly was just trying to brainstorm here with people. |
| 17:11.07 | Grnd-Wire | ManxPower: yeah - You know, that whole hybrid matching, sidetone bullshit.. Which is why it's T1/PRI all the way for me now. :) |
| 17:11.08 | ManxPower | hugs his Tellabs 96 channel T-2 echo canceller |
| 17:11.24 | ManxPower | T-1 of course, not T-2 |
| 17:11.29 | Grnd-Wire | heh.. |
| 17:11.50 | drmessano | redder86: IMO, being a tweaker and an experimenter, if you can MAKE something work, go for it.. if it's going 1000 miles away, spend money so the shit never crashes, pack it in a black box with no buttons, switches, or knobs, fill it with epoxy so it cant be fucked with, and send it off |
| 17:12.21 | seanbright | that is exactly how my russian bride was delivered. |
| 17:12.26 | redder86 | drmessano: agreed 100% ... but the hardware was a decision made years ago under a different mindset |
| 17:12.36 | *** join/#asterisk StephenF[W] (n=StephenF@198.144.197.28) |
| 17:12.54 | drmessano | redder86: Sounds like deprecation time.. Load them up with a new card, take that card for something local |
| 17:13.05 | xacatecas | [netman], thanks, Set(GLOBAL(pres_${foo})=TRUE) will work. |
| 17:13.19 | drmessano | Pay a local tech to swap the card with a digium, config it remotely |
| 17:13.19 | redder86 | drmessano: well, I'll just regress to the old versions |
| 17:13.22 | seanbright | redder86: where is the box? |
| 17:13.25 | ManxPower | drmessano: or just get a customer that is not a cheap ass,. |
| 17:13.47 | redder86 | seanbright: Hawaii |
| 17:13.57 | seanbright | redder86: that's far from me. |
| 17:13.59 | ManxPower | I do *NOT* recommend switching the card now. That will just make things even MORE complicated. |
| 17:14.01 | drmessano | ManxPower: Even if they are, demand a new card.. demand something that somewhat sensible to work remotely |
| 17:14.22 | redder86 | I won't be switching the card anyway. I'll be just regressing versions. |
| 17:14.26 | ManxPower | drmessano: I enjoy doing tech stuff, not convincing the customer they are wrong. |
| 17:15.03 | seanbright | redder86: is that patch known to work with 1.4.21.1? |
| 17:15.05 | drmessano | ManxPower: convincing? Nah.. "If you want this fixed, this is what it will take. Otherwise, it will remain broken." Period. |
| 17:15.36 | ManxPower | drmessano: *nod* Why not just not accept the customer in the first place. |
| 17:15.37 | drmessano | Cause and effect.. You paid me to fix it, I evaluated it, now here is my parts list |
| 17:15.58 | drmessano | ManxPower: If it was a problem from the onset, then set, I am 100% with you |
| 17:16.03 | redder86 | seanbright: not sure there... the patch was made against 1.4.11 ... to which I'll be regressing, and then back to 1.4.7 |
| 17:16.07 | drmessano | s/set/yes |
| 17:16.11 | ManxPower | BTW, does anyone know of most recycling places accept old computers and monitors? |
| 17:16.26 | seanbright | redder86: ahh. lots has changed between 1.4.11 and 1.4.21.1... the patch applies cleanly though? |
| 17:16.32 | ManxPower | <PROTECTED> |
| 17:16.38 | redder86 | 1.4.12.1 you mean? |
| 17:16.43 | seanbright | yes, thank you |
| 17:16.47 | redder86 | seanbright: yeah, the patch was clean |
| 17:16.49 | drmessano | ManxPower: No, but if you call your local county IT dept, they can likely point you to a place that does |
| 17:16.58 | seanbright | baltimore county does. woo. |
| 17:17.04 | seanbright | redder86: strange. |
| 17:17.05 | drmessano | They are most likely to know the responsible disposal path |
| 17:17.10 | seanbright | redder86: might just be a bug in zaptel. |
| 17:17.11 | ManxPower | drmessano: I have an entire pickup load of old cases w/motherboards |
| 17:17.36 | ManxPower | drmessano: there's a dumpster on the property if all else fails they can go there. |
| 17:17.46 | redder86 | seanbright: of course... which is why I kind of came here... but I'll backtrack versions now. |
| 17:17.51 | drmessano | Call the county.. Should be easy enough |
| 17:18.15 | seanbright | redder86: alrighty. might want to check bugs.digium.com as well (if you care). the bugs are tracked there. |
| 17:18.20 | ManxPower | drmessano: The country here doesn't even require proper disposal of used engine oil. they say "soak it up with cat litter then throw it in the trash. |
| 17:18.28 | drmessano | lol |
| 17:18.37 | redder86 | seanbright: bugs.digium.com is a joke as soon as you mention OpenVOX |
| 17:18.52 | ManxPower | The county has NO curb side pickup of recycleables |
| 17:19.08 | seanbright | redder86: i wasn't suggesting opening a bug. just taking a look to see if others are having yellow alarm problems with the latest and greatest. |
| 17:20.01 | redder86 | seanbright: yeah, I may look into that if I have trouble regressing versions |
| 17:20.27 | redder86 | seanbright: thanks for your open-mindedness |
| 17:20.45 | seanbright | nods |
| 17:22.02 | *** join/#asterisk xuser (n=JJ@unaffiliated/xuser) |
| 17:22.24 | *** join/#asterisk StephenF[W] (n=none@198.144.197.28) |
| 17:27.42 | loca|host | i have a big latency between the moment asterisk receive the call and the moment my sip phones (ring_all group) ring, some 5 seconds and that's a huge delay ... on the the flash panel, i can see the call is ringing on the trunk but transfered to the group after some time ... how to fix it ? same thing when the caller hangup, my sip phones continue ringing after that for some 5 seconds |
| 17:28.22 | *** join/#asterisk VJFROMGT (n=vjfromgt@user-12lcpfg.cable.mindspring.com) |
| 17:28.34 | VJFROMGT | <PROTECTED> |
| 17:34.07 | drmessano | #mysql ? |
| 17:34.10 | *** join/#asterisk StephenF[W] (n=none@198.144.197.28) |
| 17:40.55 | loca|host | anyone ? |
| 17:43.52 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 17:44.05 | jaytee | beuller? |
| 17:49.04 | ManxPower | loca|host: delays in incoming calls via analog FXO are usually caused by having asterisk configured to expect callerid info, but the telco is not sending the callerid info. |
| 17:56.38 | *** join/#asterisk TOrrIeri (n=torrieri@nelug/crew/torrieri) |
| 18:01.01 | zchaos | hey guys... i wanted to ask you what type of internet connection do i need to run a VOIP network? rogers has the following connections.... http://pastebin.com/m488ea235 |
| 18:01.28 | zchaos | it isn't going to be a call centre , i just might get the odd call here and there... maybe 1 or 2 at a time at MOST ... but it wont happen often just the odd call |
| 18:02.37 | drmessano | express |
| 18:02.44 | [gnubie] | how do you usually interface or integrate existing legacy branded pabx to an asterisk box? |
| 18:02.50 | drmessano | Light MAY cut it, but not with any internet comms |
| 18:03.08 | drmessano | [gnubie]: T1 seems to be most common |
| 18:03.18 | drmessano | [gnubie]: T1 cards + T1 crossover |
| 18:04.22 | [gnubie] | drmessano: i see.. so the common legacy pabx out there comes with a t1 port already? |
| 18:06.20 | drmessano | Nope |
| 18:06.20 | tzafrir_laptop | what can I do if I have an IAX client that connects from behind a NAT and can't keep a steady port? |
| 18:06.55 | drmessano | tzafrir_laptop: Sounds like the router isn't handling NAT very well |
| 18:07.30 | [gnubie] | drmessano: so you mean, you have to add a t1 module on a legacy pabx first? |
| 18:07.50 | drmessano | [gnubie]: Lets do the math here.. A legacy PBX will have T1 or FXO, right? |
| 18:08.08 | [gnubie] | drmessano: yes |
| 18:08.20 | drmessano | [gnubie]: So those are your interface options.. Most serious PBX installing will be using PRI |
| 18:08.36 | drmessano | [gnubie]: IF you need to go FXO <> FXS you can, but man.. messy |
| 18:08.42 | drmessano | Analog = blah |
| 18:08.57 | tzafrir_laptop | drmessano, some routers are known not to work very well. Which is why we have insecure=port in sip.conf |
| 18:08.59 | drmessano | How big is this OLD pbx install? |
| 18:09.10 | [gnubie] | drmessano: yes.. thanks.. |
| 18:09.44 | [gnubie] | drmessano: i actually don't have an old pbx.. i just want to know how you integrate them |
| 18:09.50 | [gnubie] | thanks.. ;) |
| 18:09.52 | tzafrir_laptop | NAT routers in this case (two different cases) - sonicwall, fortinet |
| 18:09.52 | drmessano | tzafrir_laptop: If it's having that sort of problem with IAX, I dont see much you can do with it, other than replace the router |
| 18:10.26 | drmessano | hmmm |
| 18:10.39 | drmessano | Which sonicwall? |
| 18:10.54 | drmessano | [gnubie]: How big is the PBX install? |
| 18:11.17 | tzafrir_laptop | drmessano, I'll have to check that (this is a case I recall from the past. and was not under my control, anyway) |
| 18:12.19 | drmessano | If its a TZ170, you can try new firmware.. SW is awesome about adding features, fixing bugs.. |
| 18:12.20 | *** join/#asterisk maxxim (n=maxxxim@93-96-96-106.zone4.bethere.co.uk) |
| 18:12.46 | maxxim | hi. how can i create a varibale with a value in externsion file, in order to pass it to AGI script? |
| 18:13.35 | maxxim | i was using: exten => s,n,Set(DNID=91239123) |
| 18:13.39 | maxxim | but no effect... |
| 18:17.22 | jaytee | maxxim, try Set(CALLERID(dnid)=91238123) |
| 18:18.21 | [gnubie] | drmessano: i don't have a legacy pbx |
| 18:18.33 | jaytee | I do |
| 18:18.46 | jaytee | I even have a legacy car |
| 18:19.07 | jaytee | Subaru Legacy 1995 (last 5 speed manual front wheel drive they made) |
| 18:19.14 | jaytee | needs a new water pump :-( |
| 18:19.54 | drmessano | [gnubie]: Then thats easy.. put a second NIC in your asterisk box, make a cat cable with a plug on one end, and at the other end, short 1 to 3, 2 to 6. |
| 18:20.03 | drmessano | You now have a crossover for your non=pbx |
| 18:20.56 | [gnubie] | i see.. |
| 18:21.10 | drmessano | Oh |
| 18:21.24 | drmessano | and if you want to tie in another legacy IP based PBX |
| 18:21.42 | drmessano | make a SIP peer to 127.0.0.1 and a matching peer to 127.0.0.1 |
| 18:21.50 | drmessano | viola! |
| 18:23.15 | jaytee | my legacy pbx didn't do sip without an exhorbitantly expensive SIP ITG gateway card (Nortel) so I had to put Asterisk in between it and my telco on my T1 PRI spans. |
| 18:23.24 | zchaos | drmessano so you think i need express for a little in home setup??? |
| 18:23.50 | drmessano | zchaos: The offerings you showed me.. |
| 18:23.56 | zchaos | yeah |
| 18:23.59 | drmessano | zchaos: the one below express was 1/256 |
| 18:24.09 | zchaos | ya |
| 18:24.37 | zchaos | thats to slow for a small basic voip setup? |
| 18:24.37 | drmessano | I know for a fact you'll have problems with more than one call if theres ANY internet traffic.. I went through this not too long ago with a customer |
| 18:24.49 | zchaos | ok |
| 18:25.18 | *** join/#asterisk Marquel (n=Marquel@port-300.pppoe.wtnet.de) |
| 18:25.22 | Marquel | morning |
| 18:25.27 | zchaos | is that because the d/l speeds or u/l speeds drmessano |
| 18:25.32 | drmessano | u/l speed |
| 18:25.45 | zchaos | express should be fine then eh |
| 18:26.15 | drmessano | Getting above 256 is a huge hump |
| 18:26.31 | zchaos | hmm |
| 18:27.07 | drmessano | Youre the one with the home based busines, right? |
| 18:27.49 | Marquel | is there a possibility to play a sound to both legs of a call, after the called party picked up, before bridging them? |
| 18:28.03 | drmessano | So you've got your business calls and occasional home call, correct? |
| 18:28.32 | drmessano | Lets do some math here |
| 18:28.45 | drmessano | What are the prices of the 1MB and 7MB, the two we are stuck on |
| 18:29.06 | [gnubie] | waves to all.. gtg now.. thanks.. |
| 18:29.25 | drmessano | and answer my first question.. Are you gonna use the VOIP for the home business and POTS for the home line, or going all SIP, or what? |
| 18:29.34 | drmessano | I dont remember all the details |
| 18:31.44 | drmessano | zchaos? |
| 18:36.41 | *** join/#asterisk seanmh (n=seanmh@c-68-35-21-64.hsd1.nm.comcast.net) |
| 18:39.55 | *** join/#asterisk sucituanbo (n=free@c-24-21-121-148.hsd1.wa.comcast.net) |
| 18:42.44 | [TK]D-Fender | ~iwmwb |
| 18:42.44 | jbot | I WANT MY WEEKEND BACK! |
| 18:45.03 | drmessano | I was going to give him a sensible plan |
| 18:45.37 | drmessano | Like buying G729 licenses, and saving some bacon if he's not gonna use that connection for much |
| 18:45.46 | drmessano | 1MB/256k would be enough |
| 18:46.41 | zchaos | sorry drmessano |
| 18:46.46 | zchaos | had to dos omething int he kitchen |
| 18:47.09 | zchaos | sorry drmessano please continue to help me haha |
| 18:47.12 | drmessano | So what is the plan for the comms |
| 18:47.14 | zchaos | yeah im' the guy with the home business |
| 18:47.17 | drmessano | SIP, POTS? |
| 18:47.35 | zchaos | so i will get the odd calls for the business and home |
| 18:47.48 | zchaos | probably only 2 calls at once, once in a blue mooon |
| 18:47.49 | drmessano | Both over SIP? |
| 18:47.54 | zchaos | sorry what is sip again |
| 18:48.04 | drmessano | Are you going IP or getting phone lines? |
| 18:48.17 | zchaos | phone lines |
| 18:48.22 | zchaos | i'm going ot order 2 phone lines |
| 18:48.26 | zchaos | and use ata boxes |
| 18:48.28 | ManxPower | zchaos: You should read the Asterisk Book before you do anything else. You will learn a lot. |
| 18:48.33 | drmessano | So bandwidth is irrelevant |
| 18:48.48 | zchaos | so i dont need express? |
| 18:48.49 | drmessano | Hang on |
| 18:48.56 | zchaos | drmessano |
| 18:49.00 | zchaos | i have a diagram one sec |
| 18:49.04 | drmessano | Two COPPER LINES? or two DIDs from a provider? |
| 18:49.06 | ManxPower | what is an express? |
| 18:49.17 | drmessano | Service level from Rogers cabel |
| 18:49.23 | drmessano | cable |
| 18:49.29 | ManxPower | Ah. The poor sod. |
| 18:49.35 | drmessano | Hes got 4 options.. |
| 18:49.52 | drmessano | The two I am teetering on are 1MB/256k and 7MB/512K |
| 18:50.02 | drmessano | 256k is weak |
| 18:50.05 | ManxPower | I'll bet they don't do CPC |
| 18:50.06 | zchaos | http://imagebin.org/28979 |
| 18:50.17 | zchaos | the only difff thing is |
| 18:50.20 | zchaos | tehre are going to be 2 ata boxes |
| 18:50.23 | zchaos | one for each phone line... |
| 18:50.30 | zchaos | thats the only thing missing form the diagram |
| 18:50.44 | ManxPower | zchaos: those are NOT lines, they are DIDs |
| 18:50.51 | drmessano | Wait |
| 18:50.55 | drmessano | The ATAs go to the PBX |
| 18:51.07 | ManxPower | in any case until you read The Book you're screwed and it is pointless to try to help. |
| 18:51.55 | drmessano | If you're getting 2 DIDs from a VoIP provider, you are using SIP |
| 18:51.59 | drmessano | and that will go to the PBX |
| 18:52.01 | zchaos | manxpower |
| 18:52.03 | zchaos | i have someone setting it up |
| 18:52.09 | zchaos | i was just trying ot confirm what connection i need |
| 18:52.10 | zchaos | thats all |
| 18:52.15 | drmessano | Is that the diagram they gave you? |
| 18:52.24 | zchaos | nah i did it with some assistance |
| 18:52.32 | drmessano | Because the flow is all wrong |
| 18:52.38 | [TK]D-Fender | completely |
| 18:52.44 | *** join/#asterisk beek (n=klinebl@65.211.106.242) |
| 18:52.46 | zchaos | ..... lol |
| 18:52.48 | zchaos | whats great |
| 18:52.49 | zchaos | sigh |
| 18:53.09 | zchaos | change the ata boxes to the pbx? |
| 18:53.11 | [TK]D-Fender | zchaos: If you have someone setting all this up, let them do their job. |
| 18:53.22 | zchaos | i'm doing the hardware |
| 18:53.25 | zchaos | he will do the software |
| 18:55.05 | ManxPower | This channel is too Monday'ish right now. See y'all later. |
| 18:55.08 | *** part/#asterisk ManxPower (n=manxpowe@208.sub-75-200-201.myvzw.com) |
| 18:55.38 | [TK]D-Fender | zchaos: Actually your picture is fine. |
| 18:55.42 | drmessano | zchaos: What service level do you have now? |
| 18:55.46 | zchaos | http://imagebin.org/28980 |
| 18:55.49 | zchaos | i changed the diagram |
| 18:55.51 | zchaos | fender you sure? |
| 18:55.56 | zchaos | the guy helping me was sure my diagram was right too |
| 18:55.59 | zchaos | he said it wuold work fine |
| 18:56.04 | drmessano | Get a 2 port ATA |
| 18:56.07 | drmessano | Dont get 2 ATAs |
| 18:56.12 | drmessano | Thats a huge waste |
| 18:56.19 | drmessano | 1 Port cost more than the 2 |
| 18:56.28 | drmessano | SPA-2102 |
| 18:56.28 | zchaos | i was told i need 2 |
| 18:56.30 | drmessano | No |
| 18:56.32 | drmessano | You dont |
| 18:56.32 | zchaos | so the pbx konws how to handle |
| 18:56.34 | zchaos | the calls |
| 18:56.37 | drmessano | No |
| 18:56.41 | drmessano | You were told wrong |
| 18:57.16 | zchaos | uhmmmm |
| 18:57.26 | drmessano | They make 8 port ATA's and 24 port channel banks (big ATA) that don't confuse anything |
| 18:57.35 | drmessano | So thats incorrect |
| 18:57.46 | [TK]D-Fender | zchaos: a single 2-port ATA = $50. And screw the part of the diagram showing that calls will forward to cell, etc from that side of things. Call flow is handled by the PBX, it is not an extension of your phone end-points on the ATA |
| 18:58.48 | zchaos | so was hte picture correct hte first time? |
| 18:59.36 | drmessano | no |
| 18:59.47 | drmessano | ATA was going to the router |
| 18:59.51 | drmessano | That is NOT CORRECT |
| 18:59.58 | [TK]D-Fender | drmessano: It can be for sure |
| 18:59.59 | drmessano | ATA to the PBX |
| 19:00.02 | *** join/#asterisk moy (n=moy@189.169.85.251) |
| 19:00.16 | zchaos | argh |
| 19:00.17 | drmessano | Not for call flow |
| 19:00.18 | [TK]D-Fender | drmessano: it is pgoing to be physically wired to the SWITCH on th router |
| 19:00.34 | drmessano | ok |
| 19:00.50 | zchaos | sooooo |
| 19:00.52 | zchaos | is it fine? lol |
| 19:01.10 | [TK]D-Fender | zchaos: http://imagebin.org/28979 <- this is fine for WIRING, and drop that "cell" bit off the right side |
| 19:02.08 | [TK]D-Fender | zchaos: And don't call one port "land-line business", and the other "home phone". these sound like 2 different animals. you plug a PHONE onto an ATA port. The fact you want 1 to ring for calls concerning BUSINESS is irrelevent |
| 19:02.30 | zchaos | http://imagebin.org/28981 |
| 19:02.32 | zchaos | there..... |
| 19:02.50 | zchaos | fender i know |
| 19:02.52 | zchaos | i just did it for me |
| 19:02.53 | zchaos | thats all |
| 19:03.02 | [TK]D-Fender | zchaos: What is this magical term "land line" doing on a phone connected to an ATA? |
| 19:03.34 | zchaos | a stationary phone thats all i mean by it |
| 19:03.45 | [TK]D-Fender | zchaos: they are BOTH stationary phones |
| 19:04.01 | zchaos | its just wording everyone konws what i mean |
| 19:04.21 | [TK]D-Fender | zchaos: When nothing you say adds up, no, we don't |
| 19:04.33 | zchaos | http://imagebin.org/28982 |
| 19:04.35 | zchaos | there.... |
| 19:04.46 | [TK]D-Fender | zchaos: Who can trust someones understanding when they terms are passed through a blender? |
| 19:05.03 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.35.36) |
| 19:05.34 | zchaos | there |
| 19:05.57 | [TK]D-Fender | zchaos: Better. No you can have * ring whatever phones you want for any given call. You can also choose what outbound resources to use for each port on that ATA. |
| 19:06.00 | [TK]D-Fender | Now* |
| 19:06.47 | loca|host | i have a big latency between the moment asterisk receive the call and the moment my sip phones (ring_all group) ring, some 5 seconds and that's a huge delay ... on the the flash panel, i can see the call is ringing on the trunk but transfered to the group after some time ... how to fix it ? same thing when the caller hangup, my sip phones continue ringing after that for some 5 seconds |
| 19:07.07 | [TK]D-Fender | zchaos: ATA's do not have any association to DID's, lines, or any other resources. All they are are independant SIP devices who's calls are processed by the PBX in whatever way you tell it to. |
| 19:07.18 | cvnet | anyone here used asterisk2bill ? |
| 19:07.31 | zchaos | ok |
| 19:08.46 | zchaos | ok so its all good now |
| 19:08.56 | zchaos | i think the other guy was tyring to tell me |
| 19:09.02 | zchaos | the reason he wanted 2 atas |
| 19:09.14 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:09.25 | zchaos | was because * wouldn't knbow how ot distinguish which line is ringing and how to handle it without 2 atas |
| 19:10.39 | maxxim | $AGI->exec('UserEvent','_SIP_Auth|User-Name:\ '.$UserName."|".'DNID:\ '.$dnid."|".'Channel:\ '.$input{'channel'}); |
| 19:11.02 | maxxim | unfortunately, UserEvent is not splitting the variables in Manager. why? |
| 19:11.49 | zchaos | fender 2 more questions |
| 19:11.57 | zchaos | 1) is this the one i should ordre - http://shop.gloria.ca/voip/adapters/voip-2-port-fxs-analog-adapter/prod_5352.html |
| 19:12.23 | zchaos | 2) back to my original question... so what type of connection speed do i need for this? http://pastebin.com/m488ea235 |
| 19:12.32 | maxxim | it goes like a strigth line: UserEv UserEvent: _SIP_Auth|User-Name: uknorth|DNID: 91239123|Channel: OOH323/h323gw-eec7 |
| 19:12.59 | [TK]D-Fender | zchaos: Each port is 100% separate from the other and call control is always handled by *. If you want 500 phones to ring after * accepts the inbound call, so be it |
| 19:13.17 | zchaos | gotcha |
| 19:13.22 | [TK]D-Fender | czhAnd if your other guy doesn't know this then you might want to find someone else who does |
| 19:14.06 | [TK]D-Fender | zchaos: Yes that ATA is fine |
| 19:14.44 | [TK]D-Fender | zchaos: Express. |
| 19:15.27 | zchaos | would be nice if i could find a damn provider in canada who carries the SPA2102 |
| 19:15.28 | zchaos | sheesh |
| 19:17.51 | zchaos | know of any good voip hardware providers? |
| 19:17.57 | zchaos | google is coming up dry |
| 19:18.02 | zchaos | for canada of course |
| 19:18.22 | [TK]D-Fender | http://www.canadianvoipstore.com/home.php |
| 19:18.42 | [TK]D-Fender | http://www.voipdepot.ca/ |
| 19:18.55 | [TK]D-Fender | http://www.voipware.ca/ |
| 19:19.27 | maxxim | what is the right version of UserEvent application for * 6 ? |
| 19:19.28 | zchaos | canadavoip store wants $25 ins hipping lol |
| 19:19.30 | zchaos | i saw all those places |
| 19:19.45 | maxxim | what is the right version of UserEvent application for * 1.6.0 ? |
| 19:20.02 | maxxim | what is the right usage (can you give me an example) of UserEvent application for * 1.6.0 ? |
| 19:20.30 | zchaos | thanks fora ll yoru help fender |
| 19:20.54 | [TK]D-Fender | maxxim: Is your AGI wrapper eve 1.6 compatible? |
| 19:21.31 | zchaos | fender i'm confused with these 2 products.... |
| 19:21.32 | zchaos | 1) http://www.voipdepot.ca/index.php?main_page=product_info&cPath=1&products_id=90 |
| 19:21.40 | zchaos | 2) http://www.voipdepot.ca/index.php?main_page=product_info&cPath=1&products_id=79 |
| 19:21.46 | zchaos | i dont need one wtih a router |
| 19:21.52 | zchaos | is #1 the same as #2 without hte router? |
| 19:22.04 | *** join/#asterisk maxxim (n=maxxxim@93-96-96-106.zone4.bethere.co.uk) |
| 19:22.10 | maxxim | sorry, got a disconnect |
| 19:22.24 | maxxim | i've tried directoly in extension file. got the example from wiki |
| 19:22.29 | maxxim | UserEvent(ASTDB|Family: dnd|State: on) |
| 19:22.42 | *** join/#asterisk tkbeat (n=tk@p54B962F4.dip.t-dialin.net) |
| 19:22.55 | maxxim | the problem is that in Manager it shows variables in one row |
| 19:23.19 | maxxim | but in the definiton of userEvent, it says that | should split variables |
| 19:23.24 | [TK]D-Fender | zchaos: 2102 includes a router, T.38 support, bigger CPU, etc. Worth a few extra $ |
| 19:23.47 | zchaos | point noted |
| 19:23.48 | zchaos | thanks |
| 19:23.51 | [TK]D-Fender | maxxim: Again, are you even sure your AGI wrapper is 1.6 compatible? |
| 19:24.45 | maxxim | [TK]D-Fender> exten => s,n,UserEvent(ASTDB|Family: dnd|State: on) |
| 19:24.55 | maxxim | [TK]D-Fender> i'm using directly, withou AGi |
| 19:26.19 | *** join/#asterisk rdgr (n=rich@82.32.1.139) |
| 19:26.27 | [TK]D-Fender | maxxim: Ok, looks fine. pastebin some usable backup. |
| 19:27.39 | maxxim | [TK]D-Fender> sorry, what to put in pastebin? |
| 19:27.59 | [TK]D-Fender | maxxim: AMI dumps of message attempts, CLI output from when you issue it, etc |
| 19:29.57 | maxxim | [TK]D-Fender> http://rafb.net/p/TgNfH846.html |
| 19:34.26 | maxxim | [TK]D-Fender> any other output? just tell me how |
| 19:34.39 | [TK]D-Fender | maxxim: Not sure... reading up on it now. |
| 19:37.58 | maxxim | manager_event(EVENT_FLAG_USER, "UserEvent", "UserEvent: %s\r\n%s", args.eventname, buf); |
| 19:38.20 | maxxim | who should split | , manager_event function, or in app_userevent.c ? |
| 19:39.08 | [TK]D-Fender | maxxim: Not sure, and I can't see any obvious error... |
| 19:42.20 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 19:42.35 | maxxim | [TK]D-Fender> are you still, looking? what shoul I do? |
| 19:43.43 | [TK]D-Fender | maxxim: no idea |
| 19:44.28 | maxxim | [TK]D-Fender> have you ever used this UserEvent 'application'? |
| 19:44.51 | [TK]D-Fender | maxxim: not personally |
| 19:58.23 | *** join/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 19:58.24 | *** mode/#asterisk [+o mog] by ChanServ |
| 20:03.54 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 20:09.42 | lesouvage | I have to check an Asterisk server that has trouble at random. Now and then the person that has been called doesn't hear the caller. With the next try/call it is working ok. Server is in operation for a long time without problems. Any suggestions about what can cause this kind of problems. I doubt it has to do with the Asterisk server itself. |
| 20:09.50 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:16.03 | *** join/#asterisk fatherx (i=RCP@abraham.sh.cvut.cz) |
| 20:17.00 | lesouvage | I noticed I didn't ask a question. Does any of you has suggestions about what can cause this kind of problems? |
| 20:27.31 | lesouvage | . |
| 20:29.05 | lesouvage | has the world finally come to an end to make space for a interstallair bypass ? |
| 20:32.06 | *** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net) |
| 20:32.18 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 20:33.14 | *** join/#asterisk Dr-Linux|home (n=Nothing@221.132.117.17) |
| 20:35.35 | *** join/#asterisk jmb287 (n=noc@67.16.85.15) |
| 20:35.46 | sdaniels | what is the network topology like? Iv seen this sort of thing wiith bad DSPs in a media gateway. |
| 20:35.58 | jmb287 | hi, i'm looking for some help with Ast 1.6.1 and problems with MWI |
| 20:37.02 | Dr-Linux|home | I'd like to ask same question again: |
| 20:37.03 | Dr-Linux|home | actually i want when a caller is waiting in the queue, he/she listen to a MOH but as that call is assigned to an agent/ext then caller should listen to a ring |
| 20:38.13 | jmb287 | i've got a linksys spa 942 and Asterisk 1.6.. Message Waiting seems broke, any thoughts on how to troubleshoot |
| 20:43.10 | lesouvage | sdaniels: I just have access to the asterisk server. Personally I suspect the SIP provider. |
| 20:44.01 | lesouvage | jmb287: what do you mean with "Message Waiting seems broke"? |
| 20:44.59 | jmb287 | lesouvage: I leave a vm and the phone light doesn't come on. I power cycle the phone and the light comes on. I delete msg and the light does not go off, power cycle phone and light goes off |
| 20:45.36 | sdaniels | lesouvage: well your problem is that one of the RTP streams is jacked up for some reason, there isnt really anything I can tell you without knowing the full path for each call leg.. |
| 20:45.59 | lesouvage | jmb287: sorry, I'm not a blinking light expert. |
| 20:46.36 | sdaniels | lesouvage: I suggest that if the asterisk box is on a managed witch that you set up a port monitor and wireshark it to see whats going on. id also wireshard the port the phone is on. |
| 20:46.50 | sdaniels | this keyboard sucks |
| 20:47.17 | jmb287 | lesouvage: you might also want to make sure that there isn't a firewall jacking with the RTP stream |
| 20:49.12 | lesouvage | jmb287: I suspect them to have change some network components or settings of the network components like the firewall without having their voip solution and sip trunk into consideration. |
| 20:51.06 | jmb287 | yup, time to wireshark the link, or run tcpdump on the * box |
| 20:52.42 | SQLDarkly | Has anyone noticed when using realtime sip they do not show in the CLI when using "sip show peers/users" but still register? Is this an asterisk bug or am I misssing something |
| 20:53.54 | lesouvage | sdaniels and jmb287: thanks for the input, you have been really helpfull |
| 20:54.34 | jmb287 | No worries. back to bashing my head on this stupid MWI issue, grrrr |
| 20:58.45 | maxxim | UserEvent application in 1.6 doesn't split body parameters like in 1.4 * .Could anyone help me find out why? |
| 21:04.35 | hesco | . |
| 21:04.42 | lesouvage | sdaniels: isn't a media gateway with DSPs a typical SIP telco device? |
| 21:05.27 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 21:05.40 | hesco | I'm getting lots of cdr results looking like this: duration: 88, disposition: NO ANSWER, lastapp: Hangup |
| 21:05.45 | sdaniels | lesouvage: Its a Cisco router that has voice DSPs in it to convert TDM to G711 or whatever codec you want. |
| 21:06.26 | hesco | how does that work? Hangup follows the playing of a 48 second message. |
| 21:07.04 | sdaniels | lesouvage: usually you would get something like a 3825 and put a bunch of PRI wics in it, this would be your gateway to the PSTN from your netowrk |
| 21:07.15 | lesouvage | sdaniels: The kind of device that is used by SIP providers to internconnect to the pstn network? |
| 21:08.15 | *** join/#asterisk LiNeTuX (n=LiNeTuX@253.238.95.24.cfl.res.rr.com) |
| 21:09.30 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 21:12.12 | *** join/#asterisk sbc383 (n=stuff@S0106000fea3b527f.cg.shawcable.net) |
| 21:12.33 | sdaniels | lesouvage: well yeah probably that too.. but you could call ATT or whoever and tell em you want a couple of t1s to the PSTN then plug em into the 3825, then plug your core router into the 3825 via ehternet and hang your asterisk box and phones off of the core router. |
| 21:12.58 | sdaniels | pstn---3825----yournetwork |
| 21:13.41 | sbc383 | I'm quite new to asterisk, and have a question that I can't find the answer to anywhere. In the situation where an outside caller reaches the Asterisk PBX and hits an extension to dial an internal phone, is it possible to first play a message to the inside caller, then bridge the channels? |
| 21:13.48 | lesouvage | sdaniels: yes I know, it is actually what is the "normal" scenario in Europe with 1 or more E1 . |
| 21:14.35 | lesouvage | sbc383: exten => s,n,Playback(beep) ; or any other message |
| 21:15.26 | hesco | how does that work? Hangup follows the playing of a 48 second message. Where does that disposition string come from and why does it interpret an answered call that runs to completion as having not been answered? |
| 21:15.37 | Dr-Linux|home | actually i want when a caller is waiting in the queue, he/she listen to a MOH but as that call is assigned to an agent/ext then caller should listen to a ring |
| 21:15.45 | Dr-Linux|home | is that possible? |
| 21:16.01 | lesouvage | sbc383: exten => s,n,Answer() before the Playback |
| 21:16.05 | maxxim | UserEvent application in 1.6 doesn't split body parameters like in 1.4 * .Could anyone help me find out why? Where can i report this bug? |
| 21:16.06 | [TK]D-Fender | sbc383: the A() option for Dial. |
| 21:17.00 | sbc383 | [TK]D-Fender: hehe, right in the man pages, doh! thanks dude |
| 21:18.12 | *** join/#asterisk ManxPower (n=manxpowe@208.sub-75-200-201.myvzw.com) |
| 21:21.52 | [TK]D-Fender | sbc383: when in doubt : "core show application dial" |
| 21:23.47 | jmb287 | anyone around that can help with a MWI (message waiting indication) prbm ? |
| 21:24.58 | maxxim | [TK]D-Fender> where shoul i report this issue with userevent? |
| 21:25.13 | [TK]D-Fender | maxxim: Mantis |
| 21:25.47 | maxxim | [TK]D-Fender> can you give me pls more details, where is URL for mantis? |
| 21:28.26 | ManxPower | bugs.digium.com |
| 21:28.37 | jmb287 | bugs.digium.com is not working :( |
| 21:29.08 | jmb287 | i get a tcp connection reset when trying to reach the bugs site |
| 21:29.50 | ManxPower | jmb287: then you'll have to wait. |
| 21:42.54 | Dr-Linux|home | ManxPower: moving around for a long time but my questoin is still there :P |
| 21:43.04 | *** join/#asterisk BBHoss (n=bbhoss@c-68-62-170-33.hsd1.al.comcast.net) |
| 21:43.31 | Dr-Linux|home | actually i want when a caller is waiting in the queue, he/she listen to a MOH but as that call is assigned to an agent/ext then caller should listen to a ring |
| 21:46.40 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 21:54.09 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 21:54.25 | lesouvage | . |
| 21:56.20 | jeev | sup foo |
| 21:56.25 | jeev | got any $ left over from SNL ? |
| 21:58.51 | jmb287 | looking for help with a Message Waiting issue MWI ??? |
| 22:09.14 | *** join/#asterisk mvanbaak (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 22:14.39 | *** join/#asterisk errr (n=errr@fedora/errr) |
| 22:21.48 | Qwell | drmessano: wake up |
| 22:23.31 | Dr-Linux|home | actually i want when a caller is waiting in the queue, he/she listen to a MOH but as that call is assigned to an agent/ext then caller should listen to a ring |
| 22:24.38 | mvanbaak | Qwell: let him enjoy weekend |
| 22:24.41 | Qwell | NO! |
| 22:24.45 | Qwell | :D |
| 22:24.51 | mvanbaak | meh |
| 22:25.02 | Qwell | mvanbaak: how's it going? |
| 22:25.07 | Dr-Linux|home | so any advice for that, if that is possible or not? |
| 22:25.24 | mvanbaak | good :) almost ready for a week of painting, redoing bathroom etc |
| 22:25.37 | Qwell | woot |
| 22:25.49 | Qwell | already moved? |
| 22:25.53 | mvanbaak | next weekend we'll move into our new house |
| 22:25.56 | Qwell | ahh |
| 22:26.10 | mvanbaak | nah, painting ceiling and walls is easier when there's no stuff in the new house |
| 22:26.14 | jmb287 | anyone here that can help with a MWI issue ? |
| 22:26.50 | mvanbaak | one more week and we're moved \o/ |
| 22:27.24 | mvanbaak | Qwell: Nov 29th party at my place |
| 22:27.35 | Qwell | I'll be there! |
| 22:27.38 | Qwell | well, I'll be here |
| 22:27.41 | Qwell | but I'll be there in spirit |
| 22:27.47 | mvanbaak | :) |
| 22:28.24 | mvanbaak | I'll dcc you a beer from time to time |
| 22:29.00 | mvanbaak | my new house rox, specially the kitchen |
| 22:29.32 | mvanbaak | Qwell: please reboot bugs.digium.com |
| 22:29.45 | Qwell | mvanbaak: I think only russellb can |
| 22:29.48 | mvanbaak | here's a pic of my new kitchen: |
| 22:29.51 | mvanbaak | http://picasaweb.google.com/mvanbaak/NewHouseDenHaag#5257486581735917090 |
| 22:30.51 | *** join/#asterisk Defraz (n=T0tal@63.228.246.250) |
| 22:32.52 | mvanbaak | Qwell: then call him to do it ! |
| 22:32.53 | mvanbaak | ;) |
| 22:33.07 | Qwell | just sent him an sms actually |
| 22:34.06 | jaytee | that is an awesome kitchen |
| 22:34.48 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 22:34.48 | *** mode/#asterisk [+o russellb] by ChanServ |
| 22:35.07 | *** join/#asterisk moy (n=moy@189.169.70.164) |
| 22:41.18 | mvanbaak | hey russellb ! |
| 22:41.25 | russellb | hi |
| 22:41.33 | mvanbaak | bugs. is down |
| 22:41.36 | russellb | i know |
| 22:41.39 | russellb | that's why i'm on IRC |
| 22:41.41 | russellb | trying to work on it |
| 22:41.51 | mvanbaak | poor you. even in the weekend we need you |
| 22:41.58 | russellb | :) |
| 22:42.51 | *** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net) |
| 22:44.26 | *** join/#asterisk mvanbaak_ (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 22:48.16 | drmessano | bugs is down? |
| 22:48.17 | drmessano | Hmm |
| 22:48.30 | drmessano | Did someone submit a bug report about that? |
| 22:49.13 | russellb | Qwell did |
| 22:49.17 | russellb | he submitted the bug report via sms |
| 22:49.18 | russellb | :-p |
| 22:49.22 | Qwell | to russellb |
| 22:49.26 | Qwell | <3 |
| 22:54.02 | *** join/#asterisk kerx (n=prepro@adsl-69-104-18-9.dsl.irvnca.pacbell.net) |
| 22:54.07 | kerx | howzde |
| 22:56.33 | kerx | hi, can somebody please help me? |
| 22:56.45 | kerx | having an issue w/ a sip call |
| 22:57.06 | kerx | do you know if i am using NAT, I have to do any port-forwarding setup's on the router for SIP to establish properly? |
| 22:58.07 | mvanbaak_ | ~sipnat |
| 22:58.07 | 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 |
| 22:58.34 | kerx | Thanks |
| 22:58.40 | mvanbaak | THIS IS NOT THE TIME TO GO TO SLEEP ! |
| 22:58.41 | mvanbaak | ;) |
| 22:58.52 | kerx | What ? |
| 22:59.06 | kerx | Confused me :) |
| 22:59.06 | mvanbaak | kerx: nothing. ignore that ;) |
| 22:59.09 | kerx | ok |
| 22:59.10 | kerx | :) |
| 22:59.26 | mvanbaak | kerx: it's because of the quit message of russell |
| 22:59.37 | kerx | do we have pastebin here? |
| 22:59.43 | kerx | oh i see ! :!!!! |
| 22:59.46 | kerx | slaps himself |
| 22:59.47 | mvanbaak | ~pastebin |
| 22:59.47 | jbot | [~pastebin] 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 |
| 22:59.55 | kerx | oo, nice list |
| 23:00.15 | kerx | http://pastebin.ca/1230531 |
| 23:00.17 | kerx | can u please check that out |
| 23:00.20 | kerx | something weird is going on |
| 23:01.32 | kerx | Oh wow |
| 23:01.36 | kerx | Oct 18 19:09:39 DEBUG[4968] chan_sip.c: Registration successful |
| 23:01.41 | kerx | first time I saw that in the log file :) |
| 23:01.53 | kerx | 1 sip peers [1 online , 0 offline] |
| 23:01.54 | kerx | nice |
| 23:03.48 | mvanbaak | hhmm, that pastebin is not enough info to find the problem |
| 23:03.59 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 23:03.59 | *** mode/#asterisk [+o russellb] by ChanServ |
| 23:04.08 | kerx | wb russellb who goes to sleep |
| 23:04.10 | mvanbaak | wb poor russellb |
| 23:04.20 | kerx | mvanbaak, what else do u need, i can try to give u everything :) |
| 23:04.30 | mvanbaak | <--- feels sad that russellb has to work in the weekend |
| 23:04.50 | mvanbaak | kerx: try to make the call with 'sip set debug on' |
| 23:05.07 | kerx | No such command 'sip set' (type 'help' for help) |
| 23:05.45 | mvanbaak | kerx: ok, try: sip debug |
| 23:05.56 | kerx | ok, i did |
| 23:06.07 | kerx | vici*CLI> sip debug peer gafachi |
| 23:06.07 | kerx | SIP Debugging Enabled for IP: 64.192.112.13:5060 |
| 23:07.00 | kerx | ok, got it |
| 23:07.35 | kerx | http://pastebin.ca/1230532 |
| 23:08.54 | mvanbaak | can you pastebin your sip.conf ? |
| 23:10.08 | kerx | yep |
| 23:10.13 | kerx | http://pastebin.ca/1230533 |
| 23:10.47 | kerx | ouchie, there he goes again :P |
| 23:11.04 | mvanbaak | and your box is behind nat ? |
| 23:11.30 | kerx | yep, my ip is 192.168.1.2 (asterisk machine), and 192.168.1.1 is the linksys nat |
| 23:12.05 | mvanbaak | kerx: then you need the 'externip' and 'localnet' settings in sip.conf |
| 23:12.11 | kerx | oh |
| 23:13.09 | kerx | in general or gefachi? |
| 23:13.32 | mvanbaak | general |
| 23:14.00 | kerx | ok, i added those, and i did a 'sip reload' |
| 23:14.38 | kerx | let me try now |
| 23:14.52 | kerx | same |
| 23:14.53 | kerx | Oct 18 19:23:02 NOTICE[9343]: pbx_spool.c:269 attempt_thread: Call failed to go through, reason 8 |
| 23:15.25 | [TK]D-Fender | kerx: Read up : |
| 23:15.27 | [TK]D-Fender | ~sipnat |
| 23:15.27 | 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 |
| 23:15.30 | [TK]D-Fender | ^^^^^^ |
| 23:15.40 | mvanbaak | kerx: try this: http://michiel.vanbaak.info/temp/sip_nat_settings_6.txt |
| 23:15.42 | kerx | u sure it's a nat issue at this point? |
| 23:16.04 | [TK]D-Fender | kerx: you aren't set up properly for it at all |
| 23:16.11 | kerx | why? |
| 23:17.25 | kerx | mvanbaak, i used the script, and plugged in those settings |
| 23:17.27 | kerx | externip = 69.104.18.9 |
| 23:17.27 | kerx | localnet = 192.168.1.255/255.255.255.0 |
| 23:17.29 | kerx | sip reload |
| 23:17.32 | kerx | and it did same thing |
| 23:17.33 | [TK]D-Fender | kerx: Contact: <sip:a7281st6JLPt1tEj@192.168.1.2> <-- because you did not tell * what local subnets you have and what your WAN IP is. This contact header shows that you haven't done it right. Now go read the guide |
| 23:18.35 | kerx | http://pastebin.ca/1230538 |
| 23:18.38 | kerx | can you please check this one |
| 23:18.42 | kerx | i've done the correct settings now |
| 23:18.44 | kerx | and it still fails |
| 23:19.56 | mvanbaak | kerx: SIP/2.0 404 Not Found |
| 23:20.38 | kerx | hrmm.... why would that happen? |
| 23:21.40 | mvanbaak | show us the dialplan |
| 23:22.44 | *** join/#asterisk freakazoid0223 (n=mattc@68.162.74.19) |
| 23:22.45 | sdaniels | Quick question, with asterisk are the RTP streams always routed though the asterisk server? pone--asterisk--phone or does asterisk tell the phones to connect directly? |
| 23:22.47 | kerx | http://pastebin.ca/1230542 < - extensions.conf |
| 23:23.00 | kerx | http://pastebin.ca/1230543 < - callme.call (my test call file that goes into /var/spool/asterisk/outgoing) |
| 23:23.07 | mvanbaak | sdaniels: depends on the setup |
| 23:23.21 | sdaniels | mvanbaak: whats default? |
| 23:25.56 | mvanbaak | sdaniels: see sip.conf section ';----------------------------------- MEDIA HANDLING --------------------------------' |
| 23:26.55 | kerx | so any idea on my weird setup :) |
| 23:27.10 | kerx | it's most likely i have a feeling that I don't understand callback files and how they work w/ the dialplan correctly |
| 23:28.01 | sdaniels | OK Ill take a look, the reason I ask is because I'm interested in having a phone register to an * box that is behind a firewall, if you forward 5060 to the * box, then it seems that all RDP will have to go through the * for this to work. |
| 23:28.08 | *** join/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 23:28.08 | *** mode/#asterisk [+o mog] by ChanServ |
| 23:28.31 | kerx | sdaniels, I haven't done any Port Forwarding to he * box though |
| 23:28.35 | *** join/#asterisk mvanbaak_ (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 23:28.52 | kerx | wb |
| 23:29.18 | mvanbaak_ | sdaniels: you saw my lines ? |
| 23:29.42 | sdaniels | mvanbaak: ??? |
| 23:30.09 | mvanbaak_ | 01:23 < sdaniels> mvanbaak: whats default? |
| 23:30.09 | mvanbaak_ | 01:25 < mvanbaak> sdaniels: see sip.conf section ';----------------------------------- MEDIA HANDLING --------------------------------' |
| 23:30.12 | mvanbaak_ | 01:26 < mvanbaak> ; By default, Asterisk tries to re-invite the audio to an optimal path. If there's |
| 23:30.15 | mvanbaak_ | 01:26 < mvanbaak> ; no reason for Asterisk to stay in the media path, the media will be redirected. |
| 23:30.50 | mvanbaak | my connection died |
| 23:30.59 | kerx | hehe |
| 23:31.12 | kerx | stupid ghost connection's |
| 23:31.15 | kerx | i hate 'em |
| 23:31.25 | mvanbaak | yup |
| 23:31.37 | mvanbaak | my connection is shaky the last couple of days |
| 23:31.40 | sdaniels | mvanbaak: oh yes, i saw that, thx |
| 23:31.52 | mvanbaak | and I'm not going to fix it. because I'm moving to a new place next week |
| 23:32.39 | kerx | nice |
| 23:32.48 | kerx | mvanbaak, did you have a chance to check out my dialplan, did it look ok? |
| 23:32.59 | kerx | http://pastebin.ca/1230542 < - extensions.conf |
| 23:33.01 | kerx | http://pastebin.ca/1230543 < - callme.call (my test call file that goes into /var/spool/asterisk/outgoing) |
| 23:35.54 | mvanbaak | kerx: it's pretty simple |
| 23:36.03 | kerx | yeah it looks like it's that 404 |
| 23:36.09 | mvanbaak | outboundmessage1 has no exten => 18183459801 |
| 23:36.11 | kerx | now i saw a 401 unauthorized in my SIP establishment |
| 23:36.46 | kerx | but it still shows the peer is online and established w/ me |
| 23:36.55 | kerx | why would i need a exten =>1818.. |
| 23:37.03 | kerx | that's my phone number, isn't that supposed to come from the callback file? |
| 23:37.19 | mvanbaak | # |
| 23:37.20 | mvanbaak | Context: outboundmessage1 |
| 23:37.20 | mvanbaak | # |
| 23:37.20 | mvanbaak | Extension: 18183459801 |
| 23:37.20 | mvanbaak | # |
| 23:37.22 | mvanbaak | Priority: 1 |
| 23:37.43 | kerx | ok? |
| 23:37.58 | mvanbaak | that's where you want to go to with the callfile |
| 23:38.07 | kerx | oh |
| 23:38.20 | kerx | i should replace those s's with the number huh? |
| 23:39.25 | mvanbaak | try it and see |
| 23:39.31 | kerx | <-- SIP read from 64.192.112.13:5060: |
| 23:39.31 | kerx | SIP/2.0 404 Not Found |
| 23:39.33 | kerx | damn :-( |
| 23:39.43 | kerx | Name/username Host Dyn Nat ACL Port Status |
| 23:39.43 | kerx | gafachi/a7281st6JLPt1tEj 64.192.112.13 N 5060 Unmonitored |
| 23:39.43 | kerx | 1 sip peers [1 online , 0 offline] |
| 23:39.46 | kerx | my peer's are online! |
| 23:39.50 | kerx | why it can't find it? |
| 23:40.11 | mvanbaak | it does find the peer |
| 23:40.31 | mvanbaak | BUT |
| 23:40.42 | mvanbaak | you call the peer, without telling the peer what number to reach |
| 23:40.46 | mvanbaak | I think that's the problem |
| 23:41.10 | mvanbaak | gafachi is a provider ? |
| 23:41.11 | kerx | oh |
| 23:41.13 | kerx | yes sir |
| 23:41.17 | kerx | that could be the problem |
| 23:41.27 | kerx | i don't know too much w/ call files and the dialplan |
| 23:41.41 | kerx | the whole time i thought extension was the phone number :-( |
| 23:41.47 | mvanbaak | so you try to setup a call to the provider without telling the provider who/what to call |
| 23:42.06 | kerx | where do u specifyc where to call? |
| 23:42.13 | mvanbaak | in the callfile |
| 23:42.34 | mvanbaak | SIP/gafachi/<your_cellphone> |
| 23:42.37 | kerx | yea,is that info supposed to be in the Channel: ? |
| 23:42.40 | mvanbaak | or something like that |
| 23:42.43 | kerx | oh |
| 23:42.48 | kerx | let me try that |
| 23:43.28 | kerx | shit |
| 23:43.29 | kerx | that was it :) |
| 23:43.33 | kerx | i hear the phone ring |
| 23:43.39 | kerx | hugs mvanbaak! |
| 23:43.47 | mvanbaak | there you go |
| 23:43.52 | kerx | it didnt play the sound though! |
| 23:44.04 | kerx | that's another thing :) |
| 23:44.10 | mvanbaak | yup |
| 23:45.28 | kerx | yeah, that text2wave is not even executing properly |
| 23:48.53 | sdaniels | Any opinions on the best sip trunk provider? |
| 23:50.08 | kerx | so far i've been hearing information that gafachi (one i've signed up with) is pretty good |
| 23:50.23 | kerx | but ive heard broadvox is the best |
| 23:50.34 | kerx | or i should say, one of the best |
| 23:50.43 | mvanbaak | ~ |
| 23:50.48 | mvanbaak | ~siptrunk |
| 23:50.49 | jbot | No such thing, my friend.. Like too much salty plum soda. |
| 23:51.05 | mvanbaak | ~trunk |
| 23:51.06 | jbot | methinks trunk is a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. |
| 23:52.40 | kerx | when u modify the dialplan in extensions.conf do u have to reload asterisk? |
| 23:52.56 | sdaniels | no you can reload the dialplan |
| 23:53.01 | kerx | ok |
| 23:53.16 | sdaniels | just dialplan reload |
| 23:53.43 | mvanbaak | or for older versions of asterisk: extensions reload |
| 23:54.42 | kerx | thx |
| 23:54.47 | kerx | i'm running the older it seems |
| 23:55.53 | kerx | yeah, definitely something is wrong with the sound coming out of my * machine |
| 23:56.17 | mvanbaak | what version is it ? |
| 23:56.46 | kerx | 1.2.27 |
| 23:57.40 | kerx | so besides the nat settings in sip.conf i don't have to setup port forwarding for an * client connecting to a sip server? |
| 23:57.49 | kerx | now i'm stuck at the audio not coming out of the machine (*) |
| 23:58.25 | mvanbaak | first of all, upgrade to 1.6 |
| 23:58.46 | [TK]D-Fender | No, first of all, fix your NAT settings |
| 23:58.58 | mvanbaak | meh |
| 23:59.17 | mvanbaak | man, I really should start upgrading my boxen |
| 23:59.17 | kerx | TK: I've done all the NAT settings finally :) |
| 23:59.25 | kerx | I installed w/ Vicidialnow |
| 23:59.29 | kerx | That's why I haven't upgraded yet |
| 23:59.42 | cvnet | i just installed vanilla fresh, edited my sip.conf and added user 100 and 101 (didnt touch the extention.conf) should i be able to connect to the box from outside network? |
| 23:59.51 | [TK]D-Fender | kerx: Go follow the guide. |
| 23:59.56 | kerx | Which guide? |
