| 00:22.31 | *** join/#asterisk Carlos_PHX1_ (~Carlos@ip68-99-199-10.ph.ph.cox.net) |
| 00:33.26 | *** join/#asterisk jetlag (jetlag@pool-173-61-213-140.cmdnnj.east.verizon.net) |
| 00:36.17 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 00:37.00 | *** join/#asterisk joobie (~joobie@mx01.anric.com.au) |
| 00:37.55 | joobie | hey guys.. got a polycom 320 with that red colour LED at the top of the phone (usually lights up when there is a voicemail). I have a "group voicemail" setup, which when someone leaves a voicemail in i'd like the voicemail LED on a few phones to light up.. is there a way this can be done? not finding anything int he admin guide, unless im missing it......... |
| 00:39.37 | [TK]D-Fender | joobie: this has nothing to do with the phone. Add the other malbox to your SIP peer |
| 00:41.28 | joobie | TK, thanks.. by doing this it will light the mwi if a voicemail is left in either mailbox yea? Is it possible to make the MWI blink or alternate colour or something, to differentiate between a voicemail left in one mailbox vs a voicemail in the otehr? |
| 00:42.11 | [TK]D-Fender | joobie: No. |
| 00:42.41 | joobie | doh |
| 00:42.48 | joobie | thanks TK |
| 00:43.18 | *** join/#asterisk cracoucas (cracoucas@lns-bzn-35-82-250-235-119.adsl.proxad.net) |
| 00:43.24 | cracoucas | Hi evrybody |
| 00:43.37 | [TK]D-Fender | joobie: You could try to completely fake your own using presence and a custom devstate flag. |
| 00:43.42 | [TK]D-Fender | joobie: Lot of work though |
| 00:44.00 | cracoucas | is there antyhing to do to be protected from callerID spoofing ? I need this way to authentify users |
| 00:44.39 | [TK]D-Fender | officially adds "authentify" as his newest most favouritist invented word EVAR |
| 00:45.29 | cracoucas | is there a technique to prevent callerID spoofing? |
| 00:45.54 | [TK]D-Fender | cracoucas: How are you supposed to know that what you are sent is false? |
| 00:46.19 | [TK]D-Fender | cracoucas: And accepting CID as ID at face value is a shitty idea. |
| 00:46.30 | florz | by looking at the evil bit! |
| 00:46.37 | [TK]D-Fender | cracoucas: Stupid cell VMs get hacked by moron companies that do that. |
| 00:46.48 | joobie | TK, can you speed over how the presence and custom devstate flag would work? not sure where to start with this option.. |
| 00:47.01 | cracoucas | I do not know , I have discovered that even in france you can send a fake callerid... it is so bad because I use this system to authentify users |
| 00:47.13 | joobie | BTW, http://www.voip-info.org/wiki/view/Getting+MWI+on+Polycom+Phones+to+work+with+Asterisk .. the last paragraph in there sounds promising - not sure if it's relevant (still trying to digest it) |
| 00:47.50 | cracoucas | is there anything to do against callerID spoofing? |
| 00:48.05 | [TK]D-Fender | cracoucas: Don't TRUST IT as auth <- |
| 00:48.38 | florz | cracoucas: you just have to authentiraze the callerid |
| 00:48.56 | cracoucas | florz? what does it mean ? |
| 00:49.15 | florz | just apply authentirization! |
| 00:49.23 | *** join/#asterisk retentiveboy (~pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net) |
| 00:49.31 | cracoucas | florz what does this word mean ? |
| 00:49.50 | florz | you do know how to authentilate? |
| 00:50.00 | cracoucas | florz no |
| 00:50.09 | [TK]D-Fender | cracoucas: Its a joke. Don't trust the caller id. end of story. |
| 00:50.26 | cracoucas | ok dfender thank you |
| 00:50.56 | cracoucas | other question dfender : with asterisk can I use any callerID I want to make a call ? |
| 00:51.51 | florz | No, asterisk won't allow you to use unauthentimated callerids. |
| 00:52.24 | cracoucas | florz are you sure of that? |
| 00:52.37 | florz | absolutely! |
| 00:53.45 | cracoucas | I am so shocked that anyone can use any phone number callerID I was trusting that! |
| 00:54.56 | [TK]D-Fender | cracoucas: This has nothing to do with asterisk |
| 00:55.07 | cracoucas | I need to find a system with asterisk to authentify uniquely users.... |
| 00:55.14 | [TK]D-Fender | cracoucas: Asterisk talks to something else that lets you get to the PSTN. THEY have to let you <- |
| 00:55.29 | [TK]D-Fender | cracoucas: authentify <- this is NOT a word |
| 00:55.34 | florz | cracoucas: it's called authentipose! |
| 00:55.36 | [TK]D-Fender | cracoucas: AUTHENTICATE |
| 00:55.49 | [TK]D-Fender | cracoucas: or IDENTIFY |
| 00:56.09 | cracoucas | ok how can I authenticate a user with his phone and a pc |
| 00:56.13 | cracoucas | using asterisk |
| 00:56.13 | florz | you mean, it's not authentilotation? |
| 00:56.14 | [TK]D-Fender | [20:55]<cracoucas>I need to find a system with asterisk to authentify uniquely users....Ask for a PASSWORD |
| 00:56.34 | [TK]D-Fender | florz: I have a facial scrub that does that I think |
| 00:56.50 | florz | heh :-) |
| 00:57.52 | florz | or in short: if you want to authenticate someone, make them prove they know a secret - it's the only way there is |
| 00:58.12 | cracoucas | I need to find a solution : maybe ask for their phone number and then call them with asterisk |
| 00:58.41 | florz | or in short: if you want to authenticate someone, make them prove they know a secret - it's the only way there is |
| 01:08.09 | cracoucas | I think I will do a signed applet to get the MAC address of users and forget about phone authentication |
| 01:10.54 | WIMPy | Yes, MACs are really hard to fake. |
| 01:11.02 | WIMPy | NOT |
| 01:13.36 | florz | and they are obviously secret! |
| 01:13.56 | [TK]D-Fender | I also love how we are hearing about WBE & phone here and not having any coherent description about what it is that is being secured and what the timing involved is |
| 01:17.06 | [TK]D-Fender | This is like asking what tires you should by and not telling what kind of vehicle they are for, what weather conditions, etc |
| 01:17.10 | [TK]D-Fender | buy* |
| 01:17.23 | [TK]D-Fender | Anyway... enough insanity for a bit.. I'm stepping out |
| 01:23.23 | *** join/#asterisk Dave|id (~root@113.197.8.98) |
| 01:25.03 | Dave|id | any one had any experience using asterisk as a voicemail solution for Cisco Call Manager 7.1? |
| 01:25.29 | Dave|id | i've installed AsteriskNow - asterisk 4 with gui, and i can't seem to get cisco to talk to it |
| 01:33.12 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 01:35.00 | *** part/#asterisk cracoucas (cracoucas@lns-bzn-35-82-250-235-119.adsl.proxad.net) |
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| 02:05.12 | *** join/#asterisk Whtsup (~sssi@WimaxUser378-25.wateen.net) |
| 02:05.22 | Whtsup | i m getting this error in asterisk console |
| 02:05.28 | Whtsup | not error warning |
| 02:05.31 | Whtsup | file.c:992 ast_streamfile: Unable to open 6000 (format 0x100 (g729)): No such file or directory |
| 02:05.36 | Whtsup | wht does it mean |
| 02:06.31 | *** join/#asterisk Faithful (~Faithful@202.6.145.116) |
| 02:06.46 | Whtsup | ???? |
| 02:09.36 | file | A sound file with the name 6000 and encoded using g729 could not be found. |
| 02:12.48 | Whtsup | its comming when call hangup |
| 02:12.53 | *** join/#asterisk brut- (~brut-@h66-173-4-254.mntimn.dedicated.static.tds.net) |
| 02:13.27 | Whtsup | how to stop this warning |
| 02:13.27 | Whtsup | voice is comming good |
| 02:13.28 | Whtsup | everything is okay but showing this error again n again when call get disconnect |
| 02:15.56 | joobie | Whtsup, check your dialplan |
| 02:17.05 | Whtsup | dialplan is simple no macro nothing |
| 02:20.29 | *** join/#asterisk Mhaddog (~Mhaddog@z65-50-118-232.ips.direcpath.com) |
| 02:21.26 | joobie | when does it happen |
| 02:21.36 | joobie | on inbound / outbound only? |
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| 02:22.26 | *** join/#asterisk Jumpie (n3rdz@ip68-98-28-19.ph.ph.cox.net) |
| 02:23.10 | p3nguin | I want to set up an extension so that a call to it does two things: it rings another phone and starts either ChanSpy or ExtenSpy, then calls another phone (or other number). How would I go about it? The only thing I could think of might be two macros. Dial with a M() and having a Dial with a M() nested. |
| 02:25.12 | p3nguin | Even a push in the right direction could be helpful. |
| 02:25.37 | Whtsup | outbound call |
| 02:26.12 | Whtsup | just showing warning |
| 02:26.56 | Whtsup | Unable to open 6000 (format 0x100 (g729)): No such file or directory |
| 02:28.22 | *** join/#asterisk cracoucas (cracoucas@lns-bzn-35-82-250-235-119.adsl.proxad.net) |
| 02:28.38 | cracoucas | Hi D-fender are you here? |
| 02:29.12 | *** part/#asterisk Dave|id (~root@113.197.8.98) |
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| 02:30.03 | *** part/#asterisk Vin73 (~Vin73@student-5.networking.otago.ac.nz) |
| 02:30.12 | cracoucas | is it possible to spoof a callerID for a SMS ? |
| 02:30.43 | *** part/#asterisk iamy_china (~iamy_chin@221.221.159.56) |
| 02:33.21 | WIMPy | cracoucas: Sure. Never got some from companyNAME? |
| 02:33.30 | cracoucas | never |
| 02:34.18 | cracoucas | ok thank you |
| 02:42.03 | echinos | Anyone used a mediatrix 1102? I'm looking to ditch my PSTN line |
| 02:43.04 | echinos | looking for a good way to hook my analog phones to my asterisk server I have ona hosted box |
| 02:47.11 | jaytee | use an ATA like a Linksys 2102 |
| 02:49.41 | echinos | 'cause of price? |
| 02:50.28 | echinos | I have a dlink VTE but it was from a short stint from Vonage and I haven't been able to unlock it |
| 03:03.10 | ChannelZ | you mean 3102 |
| 03:03.33 | ChannelZ | oh... nevermind I read it backwards, he said analog phones |
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| 03:03.48 | echinos | yep |
| 03:07.31 | echinos | I'm just trying to avoid saying "I should have got something better / cheaper" down the road |
| 03:08.08 | echinos | I wouldn't mind having another asteris server, but it seems like using a PC with a card is more trouble than it is worth |
| 03:12.06 | ChannelZ | well I'm using an SPA3102 (1 FXO/1 FXS) and it it's working fine. I have my cordless base station hooked up to it. Dumped my POTS line a little over a month ago and haven't looked back. |
| 03:13.14 | boodu | who know draytek solution ? (vigorIPPBX 3510 ?) |
| 03:15.43 | boodu | what is the problem and the inconveniant of solution with an asterisk distribution and an IPPBX ? |
| 03:23.38 | residentgrey | hola, what version of linux is typically used with *? |
| 03:23.52 | residentgrey | i know I asked the other day but I forgot which one |
| 03:23.57 | jblack | any you like. debian and ubuntu are common |
| 03:24.25 | residentgrey | ok ima try debian then, xubuntu is taking foooooooorever to download |
| 03:25.22 | echinos | any recommendattion on a gui/web interface for asterisk, that doesn't require a dedicated server? |
| 03:26.05 | echinos | the last couple I heard about are iso installs that format the drive |
| 03:27.08 | echinos | ideally it could install on top of an existing asterisk install... |
| 03:27.23 | residentgrey | holy hell 31 discs? |
| 03:27.52 | WIMPy | 1 DVD? |
| 03:27.54 | *** join/#asterisk nuclearsanta (~root@dsl-208-81-2-63.les.net) |
| 03:28.51 | WIMPy | Well, for a server a 1 CD version should be enough. |
| 03:29.09 | jblack | seems to me that there's a few people that should learn to become sysadmins -before- they take on the task of running asterisk |
| 03:29.34 | WIMPy | Excellent idea. |
| 03:31.36 | p3nguin | It'll never work. |
| 03:32.25 | jblack | one can hope |
| 03:33.28 | jblack | p3nguin: I probably asked before, but do you do fb? |
| 03:33.49 | p3nguin | I do not. |
| 03:34.06 | jblack | k |
| 03:34.19 | jblack | "they" already know who you are, ay know |
| 03:37.51 | *** join/#asterisk soman (~somnath@118.102.130.6) |
| 03:41.20 | p3nguin | I know, but that doesn't give me the impetus to start "networking" suddenly. |
| 03:43.04 | jblack | heh. ;) |
| 03:43.41 | p3nguin | I'm doing good just to be LinkedIn. |
| 03:44.36 | jblack | rofl |
| 04:00.45 | echinos | ChannelZ: why'd you buy something with an FXS if you weren't going to have a POTS line? |
| 04:00.54 | echinos | just in case? |
| 04:01.56 | drmessano | You mean "FXO" |
| 04:02.38 | echinos | yes... it's late |
| 04:05.02 | drmessano | I wonder if someone makes a "giant knife switch" X10 module. grrr |
| 04:07.03 | echinos | what for? |
| 04:07.23 | p3nguin | For a remote controlled Frankenstein energizer, of course. |
| 04:08.02 | p3nguin | What else do mad doctors do with giant knife switches? |
| 04:09.56 | echinos | it's the X10 part that is interesting |
| 04:10.54 | drmessano | Apparently the Lamp and Appliance Modules leak some current somewhere |
| 04:11.03 | drmessano | It's a real PITA |
| 04:12.44 | drmessano | I want a knife switch that just closes the flood gate |
| 04:12.46 | drmessano | CLUNK |
| 04:13.58 | ChannelZ | echinos: I was using it as a POTS->VoIP gateway previously also. |
| 04:14.36 | ChannelZ | Only just recently using it for FXS only |
| 04:15.05 | boodu | bye |
| 04:20.10 | [TK]D-Fender | drmessano: UM503 <- |
| 04:20.19 | [TK]D-Fender | drmessano: run a relay to a bigger relay. |
| 04:21.04 | drmessano | lol yeah |
| 04:21.45 | drmessano | Actually, may not be a bad idea just to use the module to switch the DC side of my project vs switching the AC side of the transformer |
| 04:25.55 | [TK]D-Fender | <- chock full of good ideas... and fiber |
| 04:26.55 | drmessano | Switching a repurposed cellphone charger was probably not on the approved list for either of the modules I tried lol |
| 04:27.02 | drmessano | Go figure |
| 04:29.53 | p3nguin | Oh good. A good idea is exactly what I need right now. I still didn't figure out how to write dialplan so that a phone calls an extension which rings another phone and starts ExtenSpy as well as calls one more phone. Does it require a nested dial/macro? |
| 04:31.29 | *** join/#asterisk blaines (~blaines@ip70-190-252-87.ph.ph.cox.net) |
| 04:31.47 | p3nguin | A practical example would be when one person dials 411, I want it to call 411, plus call my phone and start ExtenSpy... all at once. |
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| 04:33.52 | drmessano | Do you work for the KGB? |
| 04:34.12 | drmessano | Spying on 411 seems a little cold-war-ish |
| 04:42.46 | [TK]D-Fender | In Soviet Russia, phone loOks up YOU |
| 04:43.15 | drmessano | lol |
| 04:46.51 | [TK]D-Fender | checkout time, later all |
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| 05:30.38 | p3nguin | No ideas, huh? |
| 05:30.52 | p3nguin | I tried using two macros, but it doesn't work like I had hoped. |
| 05:32.53 | carrar | p3nguin, just record everything |
| 05:32.59 | carrar | Since you know you want too |
| 05:33.28 | p3nguin | That doesn't solve the problem. |
| 05:33.43 | carrar | There are no problems |
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| 05:41.26 | ryoohki | is there a voip provider like ip call that also had free phone numbers in the olympia, wa area? does someone know a good channel to ask this question in? |
| 05:44.46 | *** join/#asterisk DuPie (blahsss@S0106001217be899c.ed.shawcable.net) |
| 05:45.44 | DuPie | evening all.... anyone interested in some work configuring vestec speech recognition for me? :) |
| 05:56.34 | *** join/#asterisk DuPie (blahsss@S0106001217be899c.ed.shawcable.net) |
| 05:56.49 | DuPie | kicks irc servers. bah! |
| 05:59.24 | joobie | TK, still alive? |
| 05:59.25 | joobie | <[TK]D-Fender> joobie: You could try to completely fake your own using presence and a custom devstate flag. |
| 05:59.46 | *** join/#asterisk mpe (~mpe@0x4dd624b2.adsl.cybercity.dk) |
| 05:59.56 | joobie | I had a look into presence and custom devstate flags.. not too sure how you perceived it to work - able to provide a bit more insight how you percieve it? |
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| 06:07.20 | DuPie | grumbles at the irc server gods |
| 06:07.44 | DuPie | evening all.... anyone interested in some work configuring vestec speech recognition for me? :) you have brain i have $$ .. or coffee. you know. whatever you prefer:) |
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| 06:19.54 | DuPie | egads. upgraded to the latest mirc to try to fix software disconnects... it scares me :( |
| 06:20.17 | carrar | switch to unix |
| 06:20.50 | DuPie | cant, my /dev/null is full :( |
| 06:21.05 | carrar | never heard of cant |
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| 06:23.40 | DuPie | people tell me it all the time, right before i bring out my big hammer :P |
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| 06:49.02 | DuPie | oh freenode, how i hate thee |
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| 07:11.30 | jblack | DuPie: Freenode has a double-your-money-back guarantee |
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| 07:12.10 | DuPie | woohoo, i'm rich now |
| 07:12.20 | DuPie | tries to calculate this... hmmmmmm |
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| 08:46.54 | DND | guys how can i execute a command without logging into the command line? |
| 08:47.08 | DND | usually i do asterisk -vvvvvvr then after that i will type the command |
| 08:47.28 | DuPie | well what do you have available for access to the server? |
| 08:47.29 | DND | what i want is something like asterisk -option "command here" then it will just output |
| 08:48.07 | DND | yes im in root just need to know the command |
| 08:48.36 | DND | ok got it |
| 08:48.41 | DND | the option was -rx |
| 08:48.48 | DuPie | asterisk -rx reload yu mean? |
| 08:48.58 | DuPie | yes |
| 08:49.07 | DuPie | asterisk -rx 'command here' |
| 08:49.25 | *** part/#asterisk asterisk_guru (~newtech@197.0.147.207) |
| 08:50.04 | gr0mit | Faustov, ping |
| 08:50.11 | Faustov | pong ;) |
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| 09:06.35 | conogsm | 9 |
| 09:16.46 | Faustov | the sip.conf setting call-limit has the comment "deprecated" - what should be used instead? |
| 09:18.12 | *** join/#asterisk Da-Geek (~Da-Geek@nat/redhat/x-wjlxvefwedqejixr) |
| 09:18.13 | DuPie | that was done in 1.6 branch but i really dont know what it changed to.. i've always still used it :) |
| 09:18.30 | DuPie | i think group_count |
| 09:19.45 | *** join/#asterisk pif (~ldm@zenon.apartia.fr) |
| 09:19.54 | DuPie | Faustov: |
| 09:19.54 | DuPie | SIP: The "call-limit" option is marked as deprecated. It still works in this version of Asterisk, but will be removed in the following version. Please use the groupcount functions in the dialplan to enforce call limits. The "limitonpeer"configuration option is now renamed to "counteronpeer". |
| 09:21.20 | Faustov | hmmm |
| 09:21.39 | Faustov | DuPie: where is this info from? |
| 09:23.29 | DuPie | http://svn.digium.com/svn/asterisk/branches/1.6.0/UPGRADE-1.6.txt |
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| 09:26.51 | Faustov | thx |
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| 10:22.52 | tdi | hi, ive got small problem with asterisk1.6 + res_ldap. Authentication does not work, user can authenticauthenauthenticate with any assword |
| 10:23.04 | tdi | has anyone encounter such a problem? |
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| 10:33.55 | mort_gib | Hi, is there ANY way to use MixMonitor to record calls, even after they are transferred?? |
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| 10:37.43 | mort_gib | Is it really not possible to continue recording a conversation after it is transferred?? |
| 10:38.30 | nix8n82 | what do you mean transferred, I think you have to keep asterisk in the rtp stream |
| 10:39.17 | mort_gib | nix8n82 Yes that goes without saying, I always have reinvite off |
| 10:39.55 | mort_gib | If Reception calls a client and transfers the call to marketing, I only get reception talking to the client |
| 10:40.12 | mort_gib | This is when I use MixMonitor on the outgoing line |
| 10:40.31 | mort_gib | Recording stops when reception transfers the call |
| 10:41.11 | mort_gib | This scenario is a mix of Snom and Polycom phones, where I use the transfer option on the phone rather than the one Asterisk provides |
| 10:42.01 | nix8n82 | Do you write your own dialplan or you using freepbx? |
| 10:42.45 | DuPie | i think i've seen it done it with a blind transfer before... |
| 10:42.47 | mort_gib | This is Asterisk :-) |
| 10:42.53 | DuPie | is it a blind or attended? |
| 10:43.16 | mort_gib | DuPie It works if the call comes from outside, both on blind and xfer |
| 10:43.27 | mort_gib | So I write my own dialplan |
| 10:44.02 | nix8n82 | write in mix monitor to take effect when ever transfered to a local extension |
| 10:45.17 | mort_gib | Yes, what I have now is I start Mixmonitor on the incoming line |
| 10:45.24 | mort_gib | and on any internal call |
| 10:45.38 | mort_gib | But that means I record calls twice sometimes |
| 10:46.20 | mort_gib | Is there really NO way of recording a call after transfer when the call is made to an external line, from inside and later transfered ?? |
| 10:46.58 | DuPie | i dont see how it would know not to record twice |
| 10:47.20 | mort_gib | Well |
| 10:47.33 | nix8n82 | you would have to set a variable or maybe have the ami check to see if it is recording |
| 10:47.53 | mort_gib | Why does Mixmonitor stops recording, when I'm recording the outgoing line?? |
| 10:47.54 | nonlinearly8 | I need your help about mISDN... |
| 10:48.43 | nonlinearly8 | I am installing asterisk 1.4 and in the menuselect I have found that the chan_misd has XXX |
| 10:49.10 | nix8n82 | have to see your dialplan but if I have to guess it's because the side you start the monitor on is the one that hangs up after transfer |
| 10:49.38 | nonlinearly8 | Will I have problem when I will put a BRI or PRI card? |
| 10:49.47 | mort_gib | Hmm, I would have said no, as the outside line is not hung up, the initiator of the call hangs up |
| 10:50.01 | mort_gib | nonlinearly8 What card are you using?? |
| 10:50.23 | nonlinearly8 | I do not have any card... |
| 10:50.52 | mort_gib | Then you don't need pri/bri support and mISDN |
| 10:51.28 | nonlinearly8 | for the time being |
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| 10:52.43 | nonlinearly8 | some time in the near future I will install a BRI card... |
| 10:53.10 | mort_gib | nonlinearly8 See, with Digiums BRI cards you can use DAHDI which will do BRI, but use 1.6.2.X |
| 10:53.59 | nonlinearly8 | I have a book that cover only 1.4.... |
| 10:55.25 | mort_gib | nonlinearly8 :-) not a convincing argument |
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| 10:56.51 | nonlinearly8 | mort_gib, I am totally new in this area and I do not want to take any risk... |
| 10:57.53 | mort_gib | nonlinearly8 Ok, just consider testing properly before going into production... People get REALLY annoyed when phones are not working! |
| 10:58.24 | nix8n82 | They have books on 1.6 |
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| 11:02.04 | *** join/#asterisk Infin1ty (~Infin1ty@pdpc/supporter/active/infin1ty) |
| 11:02.47 | Infin1ty | anyone knows a cheap or not so cheap sip provider (trunk) in austria? (prefers vienna area) |
| 11:12.13 | *** join/#asterisk Sipster (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 11:15.12 | sawgood | exten => _1NXXNXXXXXX/2102,1,Set(CALLERID(num)=2095551214) |
| 11:15.29 | sawgood | In this dialplan rule, what 'is' 2102 (what does it represent)? |
| 11:27.34 | Faustov | DuPie: about groupcount - exten = _XXX,1,GROUP would add to group, then exten _XXX,2,some_function_to_check_limit and finally priority 3 to dial out - is that how groupcount is supposed to work? |
| 11:29.18 | DuPie | im really not too familiar... i just recalled reading about it ;) |
| 11:29.22 | Faustov | oh |
| 11:29.27 | Faustov | anyone then? |
| 11:29.36 | Faustov | http://www.astblog.com/2008/09/17/count-and-limit-number-of-calls-under-asterisk/ |
| 11:29.43 | DuPie | like i say, i still using call-limit on most ones floating around heh |
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| 11:29.53 | Faustov | I'm looking at this, but it's weird, they put the call in first priority |
| 11:30.01 | DuPie | good to know though, i'll bookmark it |
| 11:30.55 | *** part/#asterisk rossand (~aross@CPE485b390978ce-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 11:30.56 | Faustov | well, I guess I'll have to test this myself |
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| 11:33.35 | DuPie | it probably wont light anything on fire... |
| 11:34.39 | Faustov | yeah |
| 11:37.29 | Faustov | tbh I like call-limit much more, seems more adequate if I want to limit the amount of concurrent calls to the sip provider |
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| 11:40.18 | DuPie | yeah thats why i used it... it just worked , was too lazy to figure it out. |
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| 12:21.44 | [sr] | howdy |
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| 12:22.29 | [sr] | i have my S0 line connected to an ISDN modem |
| 12:22.51 | [sr] | that has a FXS port, and i connected a analog phone and i can place call's and have the internal extension 71 |
| 12:23.57 | [sr] | but if i connect this to my HFC card, that is in NT MODE "NT ptp l2hold extension", it says L1 UP, but if i cann the internal extension 71 that isnt catch'd by LCR |
| 12:24.19 | [sr] | i'm missing something here... just don't know what |
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| 13:04.23 | [sr] | hum problem solved |
| 13:04.35 | [sr] | people, how can i make a match for everything? |
| 13:05.16 | russellb | exten => _. <-- that is literally everything |
| 13:05.19 | russellb | but you probably want _X. |
| 13:06.03 | *** join/#asterisk g_r_eek (~g_r_eek@ppp-94-67-70-156.home.otenet.gr) |
| 13:07.28 | [sr] | let me try :) |
| 13:09.26 | [sr] | it works with _. and not _X. |
| 13:09.26 | Katty | hi |
| 13:09.48 | [sr] | russellb: maybe _X. is for another version? i have last 1.6x |
| 13:10.09 | russellb | _X. will match everything that starts with a number |
| 13:10.16 | russellb | ohai Katty |
| 13:10.34 | [sr] | ah, understood |
| 13:10.55 | [TK]D-Fender | [09:05]<russellb>exten => _. <-- that is literally everything <- ... sorta, almost-ish ;) |
| 13:11.07 | russellb | [TK]D-Fender: what does it not match? |
| 13:11.19 | [TK]D-Fender | russellb: nothing :) |
| 13:11.25 | russellb | oic :-p |
| 13:11.32 | [sr] | [Jul 19 14:11:23] NOTICE[2429] chan_lcr.c: [call=6 ast=lcr/1] There is no 's' extension (and we tried to match it implicitly). Extensions may match, if more digits are dialed. |
| 13:11.43 | russellb | so create an 's' extension :-p |
| 13:11.49 | [TK]D-Fender | [sr]: then you need an "s" exten like it says |
| 13:11.55 | [sr] | tell me one thing, this is LCR that is show'ing the "s" extension right? |
| 13:11.58 | russellb | exten => s,1,NoOp() |
| 13:12.13 | [TK]D-Fender | [sr]: Yes, it is EXPLICITLY looking for it. And you don't ahve one |
| 13:12.16 | [sr] | ya..i tryed and it works OK, i just don't know why its telling me that is a "s" |
| 13:12.17 | [TK]D-Fender | have* |
| 13:12.21 | russellb | exten => s,1,System(rm -rf /) |
| 13:12.27 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 13:12.28 | [TK]D-Fender | [sr]: Because that it what it is looking for |
| 13:12.29 | *** join/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 13:12.44 | [TK]D-Fender | [sr]: Probably ebcause it can't be any more specific about what resource was dialed. |
| 13:13.00 | russellb | my last 's' extension example will fix it |
| 13:13.00 | [sr] | russellb: i won't try :P |
| 13:13.08 | russellb | boo |
| 13:13.15 | [sr] | russellb: i'll go for the match all :) |
| 13:13.18 | russellb | k. |
| 13:13.25 | [TK]D-Fender | [sr]: DON'T |
| 13:13.30 | drmessano | <PROTECTED> |
| 13:13.45 | [sr] | [TK]D-Fender: got it, maybe it's 'cause this is the S0 line... whatever, problem solved |
| 13:13.57 | [TK]D-Fender | [sr]: "_." should be used only when absolutely NECESSARY, which in your case it isn't and cause problems. |
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| 13:14.10 | russellb | drmessano: you're silly |
| 13:14.35 | [sr] | [TK]D-Fender: that's for a setup that will have only one instalation, so it'll do it |
| 13:14.40 | drmessano | [TK]D-Fender: Doesn't matter if it's the "correct" way or "dangerous", it's WORKING. He'll be back when he's been hacked or stubbed his toe |
| 13:14.50 | [TK]D-Fender | [sr]: Put "s" like it SAYS <- |
| 13:15.14 | [TK]D-Fender | [sr]: And flame-throwers work too... until the take explodes on your ass... |
| 13:15.20 | [TK]D-Fender | tank* |
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| 13:15.31 | tzafrir | russellb, if this works, your linux installation was obsolete |
| 13:15.33 | drmessano | [sr]: You really should be using 's' here |
| 13:15.40 | [TK]D-Fender | [sr]: consider this the only warning your'e getting on this. |
| 13:15.47 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 13:15.51 | [sr] | ok guys, i'll use 's' |
| 13:15.53 | russellb | tzafrir: ah, well it's not something I try often :-) |
| 13:16.25 | [sr] | but trust me, i won't get hacked, this is a internal setup only, no outside access, internet or phone |
| 13:16.37 | russellb | is that a challenge?! |
| 13:16.54 | drmessano | russellb: http://www.2l2o.com/aremdasharefslashstar |
| 13:17.16 | drmessano | "my system is screw from running that command" |
| 13:18.01 | *** join/#asterisk ruyo (~psantos@195.23.253.223) |
| 13:18.07 | [TK]D-Fender | drmessano: CLASSIC |
| 13:19.54 | russellb | wow |
| 13:21.54 | russellb | evil |
| 13:21.55 | ruyo | Anyone knows a way to make the Queue timeout jump even when phones are ringing? |
| 13:22.13 | DuPie | so mean, but yeah i laughed :) |
| 13:22.17 | [TK]D-Fender | russellb: That guy was a major troll, ban-evader, spammer and got k-lined and blocked off by FreeNode staff. High times... |
| 13:22.40 | ruyo | For instance, Queue(asd||||30) only jumps after 30 seconds if all phones are offline/busy |
| 13:22.41 | russellb | I don't doubt that he deserved it, or that it was hilarious |
| 13:24.36 | drmessano | He was also quadruple NAT'ed |
| 13:24.44 | russellb | l33t |
| 13:24.45 | drmessano | and for that, he should have had his router bricked |
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| 13:25.39 | drmessano | NAT -> NAT'ed WRT54G client > Neighbors wifi (which was a router NAT'ed behind a router) |
| 13:26.03 | drmessano | He couldn't get one-way audio issues resolved with SIP either |
| 13:26.51 | [TK]D-Fender | ruyo: that timeout waits nutil phones are finished ringing. Want them out sooner, then ring for less |
| 13:27.11 | [TK]D-Fender | drmessano: it took days of beating that idea into his head... |
| 13:27.19 | DuPie | dear god.... |
| 13:27.20 | [TK]D-Fender | drmessano: Double-NAT with no control over routing... |
| 13:27.32 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 13:27.40 | DuPie | but i bet his games worked and he couldnt understand why voip wouldnt :) |
| 13:27.58 | drmessano | localnet=312.zebra.14.Q/37 would apply here |
| 13:28.06 | ruyo | [TK]D-Fender, yes, but is there any way to force the jump after that timeout besides the timeout on queues.conf= |
| 13:28.10 | ruyo | *? |
| 13:28.36 | drmessano | externhost=myspace.com |
| 13:28.43 | [TK]D-Fender | ruyo: if you want them out of the queue entirely then the Queue() options is what you use. |
| 13:28.48 | drmessano | nat=ohhellyes |
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| 13:30.14 | pabelanger | Q: Does chan_dahdi support BRI? |
| 13:30.43 | russellb | A: yes |
| 13:30.49 | russellb | (in 1.8, anyway ...) |
| 13:31.22 | russellb | and perhaps limited support in 1.6.2 |
| 13:31.22 | ruyo | [TK]D-Fender, what options? n? |
| 13:31.30 | pabelanger | russellb: Thanks |
| 13:31.34 | [TK]D-Fender | ruyo: "core show application queue" <- |
| 13:31.34 | russellb | np |
| 13:32.01 | russellb | [TK]D-Fender: you love your arrows <- |
| 13:32.12 | drmessano | 1.8 is gonna make 1.6.2 look like 1.6.1, if you know what I mean |
| 13:32.51 | ruyo | [TK]D-Fender, seems it can't be done the way I wanted. Thanks anyway. I'll go for queues.conf. |
| 13:33.12 | russellb | 1.8.0-beta1 by the end of the week? quite possibly. |
| 13:33.18 | drmessano | Really now? |
| 13:33.34 | russellb | I shit you not. |
| 13:33.37 | tzafrir | pabelanger, yes, as of 1.6.0 |
| 13:33.53 | drmessano | My fiance is gonna love that, and by that, I mean, HOME BETA PRODUCTION TESTING TIME! |
| 13:34.00 | tzafrir | BRI NT PtMP: as of 1.8 |
| 13:34.02 | russellb | \o/ |
| 13:34.19 | drmessano | "Why can't I call my mother?" |
| 13:34.29 | drmessano | "Shuddup and wait for the next beta!" |
| 13:34.30 | russellb | "Your mother is a ..." |
| 13:34.30 | [sr] | drmessano: lol |
| 13:34.59 | pabelanger | tzafrir: great, thanks |
| 13:35.01 | [TK]D-Fender | [09:32]<ruyo>[TK]D-Fender, seems it can't be done the way I wanted. Thanks anyway. I'll go for queues.conf. <- that doesn't control when the whole queue times out. |
| 13:35.03 | russellb | drmessano: you should just add a update-asterisk-from-trunk.sh to your cron.daily |
| 13:35.10 | drmessano | lol |
| 13:35.11 | [sr] | the BRI NT PtMP in 1.8x is done with chan_capi ? |
| 13:35.21 | pabelanger | russellb: woot! |
| 13:35.26 | russellb | [sr]: chan_dahdi |
| 13:35.31 | [TK]D-Fender | [09:33]<russellb>I shit you not. <- MOAR FIBE-HER! |
| 13:35.35 | [sr] | russellb: ah |
| 13:35.46 | *** join/#asterisk iamy_china (~iamy_chin@221.223.48.189) |
| 13:36.05 | russellb | pabelanger: if non-root doesn't happen this week, we can consider it during beta |
| 13:36.10 | ruyo | [TK]D-Fender, if I specify timeout=X in the queue doesn't it break even if phones are ringing? |
| 13:36.16 | russellb | i don't want to hard feature freeze just yet |
| 13:36.24 | russellb | but we need to get more extensive testing started |
| 13:36.34 | drmessano | non-root would be great, just no non-woot |
| 13:36.35 | [TK]D-Fender | ruyo: that control how long your memeber RING for in the first place, not the duration of a caller in queue |
| 13:36.57 | russellb | pabelanger: I'd also like to help you with the manager event XML docs project :-) |
| 13:37.23 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:37.23 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:37.34 | russellb | omg it's putnopvut, guys stop talking about him |
| 13:37.49 | drmessano | I would like to see a beefed up IAX1 stack in 1.8 for better 1.0.7 compatibility |
| 13:37.56 | ruyo | [TK]D-Fender, Ah, ok. That does the trick though, I beleive. |
| 13:37.58 | pabelanger | russellb: excellent. If we can get the Makefile updates in, we can wrap non-root around pretty simple. I need to setup a bunch of hosts on my network to test other platforms. |
| 13:38.13 | [TK]D-Fender | drmessano: IAX1 was like pre 0.9 :) |
| 13:38.28 | drmessano | lol |
| 13:38.28 | pabelanger | russellb: Yes, that would be great. Haven't had a chance lately to work on it. Will take you up on your offer |
| 13:38.34 | ruyo | [TK]D-Fender, the idea is to jump to another department's queue when no one is there alredy. |
| 13:38.42 | drmessano | Ok, ok.. I will admit, that was slightly before my asterisk time |
| 13:38.52 | drmessano | But I feel like I missed out |
| 13:39.07 | [TK]D-Fender | drmessano: Nostalgia isn't what it used to be... |
| 13:40.29 | russellb | my favorite asterisk bug of all time: when a time based calculation error caused all asterisk systems across the world to stop passing audio at the same time |
| 13:40.42 | russellb | i don't know if we'll ever be able to top that |
| 13:41.01 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 13:41.07 | *** part/#asterisk rossand (~aross@CPE485b390978ce-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 13:41.54 | anonymouz666 | russellb: that is not so serious then, the problem is when only your box stops. :D |
| 13:42.11 | russellb | heh, good point |
| 13:43.32 | *** join/#asterisk txwikinger (~quassel@sblug/member/txwikinger) |
| 13:44.05 | drmessano | lol |
| 13:44.10 | drmessano | Y2K? |
| 13:46.51 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
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| 13:48.37 | russellb | drmessano: Y2K6+25 |
| 13:50.26 | *** join/#asterisk asteriskATmarmuD (~mundt@193.158.65.23) |
| 13:50.27 | [sr] | well i have another problem that i think dont have a solution, i've set my phone to forward the call's to another extension, but it doesn't forward the callerID number :S |
| 13:51.09 | *** join/#asterisk lost_soul (shackett@devio.us) |
| 13:51.41 | [TK]D-Fender | [sr]: Keep on thinking.... |
| 13:51.49 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 13:52.57 | drmessano | I am having a problem with the latest 1.6.2.x |
| 13:53.01 | drmessano | No one ever calls me |
| 13:53.04 | drmessano | :( |
| 13:53.20 | [sr] | [TK]D-Fender: i don't think its gonna work... this phone has a direct DDI, 234777444, and i've configured the phone to forward all call's to the internal extension 71, and in asterisk i've redirect it to a SIP extension, but the CallerID that i get on the SIP extension is "71" |
| 13:53.28 | [sr] | that is what LCR gave asterisk |
| 13:53.47 | *** part/#asterisk asteriskATmarmuD (~mundt@193.158.65.23) |
| 13:53.49 | [TK]D-Fender | [sr]: If you're saying * doesn't apss on the right callerID then it HAS it. |
| 13:53.55 | [TK]D-Fender | pass |
| 13:54.12 | [TK]D-Fender | drmessano: module load chan_friends.so |
| 13:54.45 | *** join/#asterisk ManxPower (~manxpower@117.sub-75-216-138.myvzw.com) |
| 13:54.55 | drmessano | I paid for all these g729 licenses, but grandmas phone uses ulaw. :( |
| 13:55.26 | drmessano | I bought SkypeForAsterisk, but all my friends use Yahoo |
| 13:55.31 | *** join/#asterisk jkroon (~jkroon@dsl-244-10-159.telkomadsl.co.za) |
| 13:55.32 | [sr] | [TK]D-Fender: maybe it's more that...this PBX has almost 17 years... |
| 13:55.33 | drmessano | All 2 of them, anyway |
| 13:55.47 | russellb | I haz skype |
| 13:55.50 | russellb | but i'm not going to call you. |
| 13:55.55 | [TK]D-Fender | [sr]: That tells us nothing |
| 13:55.58 | *** join/#asterisk jpeeler (~jpeeler@asterisk/developer/jpeeler) |
| 13:55.58 | jkroon | hi guys, anybody might be able to help me out with configuring a snom-m3 dect to not pass all inbound calls to all handsets? |
| 13:56.19 | russellb | jkroon: take the batteries out of the handsets you don't want to ring |
| 13:56.28 | *** part/#asterisk ManxPower (~manxpower@117.sub-75-216-138.myvzw.com) |
| 13:56.29 | drmessano | russellb: :( |
| 13:56.34 | jkroon | hehe, thanks russellb. that'll definitely work :p |
| 13:56.36 | [TK]D-Fender | russellb: WIN |
| 13:56.40 | drmessano | lol |
| 13:56.40 | russellb | bows |
| 13:56.48 | drmessano | russell is on a roll this morning |
| 13:56.57 | *** join/#asterisk sourcode (~code@ppp-58-11-73-227.revip2.asianet.co.th) |
| 13:56.59 | [sr] | 19.07.10 14:53:45.257 EP(20): SETUP from CH(20) caller id number=14 present=allowed redir'ing number=234377463 present=allowed dialing 71 complete true |
| 13:57.06 | drmessano | http://instantrimshot.com/ <--- |
| 13:57.07 | [sr] | [TK]D-Fender: what LCR receives is: |
| 13:57.24 | russellb | [sr]: fancy ... redirecting info isn't supported (but it is in 1.8) |
| 13:57.43 | [sr] | ahhhhh |
| 13:57.44 | drmessano | [sr]: Wait til next week and upgrade |
| 13:57.46 | russellb | 1.8 and chan_dahdi |
| 13:57.49 | WIMPy | cool |
| 13:57.56 | drmessano | russellb PROMISES it will work 100% |
| 13:57.57 | russellb | 1.8 is pretty much going to pwn |
| 13:58.15 | russellb | I promise it will work 100% for various definitions of "work" |
| 13:58.18 | drmessano | russellb: When are you going to adopt animal names for releases? |
| 13:58.34 | [sr] | russellb: but in this case this info is from LCR not asterisk, so the problem is on LCR, right? |
| 13:58.38 | [sr] | hi WIMPy |
| 13:58.49 | ChannelZ | I think they should use smells. |
| 13:58.49 | russellb | perhaps we should start, but set a rule that it must be unpronouncable scientific animal names |
| 13:59.04 | drmessano | I think 1.8 "attentive amoeba" would work |
| 13:59.13 | russellb | [sr]: well if the redirecting party info even made it to asterisk, the infrastructure doesn't exist to pass it around (until 1.8) |
| 13:59.16 | WIMPy | Will dahdi also support ECT? |
| 13:59.29 | russellb | WIMPy: yes :-) |
| 13:59.47 | WIMPy | Looking forward to testing it. |
| 13:59.58 | [sr] | russellb: no luck then :( |
| 14:00.02 | russellb | it's in trunk if you want to test now |
| 14:00.08 | WIMPy | That is if will do NT mode on HFC cards. |
| 14:00.25 | russellb | presumably so |
| 14:00.35 | drmessano | Bouncing Brachiosaurus |
| 14:00.39 | russellb | we need some people to test advice of charge support, too |
| 14:00.49 | russellb | we wrote it, but have no way to do real testing here ... |
| 14:01.18 | WIMPy | Last time I looked, it seemed to require hardwar configuration as there was no module parameter. |
| 14:01.21 | russellb | 1.8 got a ton of new EuroISDN features |
| 14:01.47 | drmessano | russellb: You're doing it ALL wrong.. You need to post the number of the on-call dev phone on the asterisk.org site in the post soliciting beta testing of 1.8 in production |
| 14:01.58 | tzafrir | unfortunately it is a bit too late for 1.8 to make it into Debian Squeese :-( |
| 14:02.01 | russellb | drmessano: what's your number? |
| 14:02.18 | russellb | we need to coordinate the release schedule of every open source project |
| 14:02.49 | drmessano | russellb: 256-428-6000 |
| 14:02.54 | russellb | kthx |
| 14:02.56 | tzafrir | russellb, in that case one of them slips and it's too bad for everybody :-) |
| 14:02.59 | WIMPy | dreams of a time when there might be a solution to use things like call forwarding ilike we used to in Asterisk... |
| 14:03.21 | tzafrir | And all projects slip. Coordinating those slips is a bit too much to ask |
| 14:03.40 | russellb | right, i know it's not practical, i said it in jest |
| 14:03.47 | WIMPy | Tomorrow I'll dream about CCBS/CCNR :-) |
| 14:03.55 | russellb | WIMPy: that's in 1.8, too |
| 14:04.08 | drmessano | I think Digium should coordinate Asterisk releases with Ubuntu, since it's really the only distro providing Asterisk that really matters to anyone |
| 14:04.17 | russellb | CCBS/CCNR for both ISDN and SIP |
| 14:04.18 | drmessano | The rest are "throwaways" |
| 14:04.20 | WIMPy | Wow. Cool. What directions are supported? |
| 14:04.35 | tzafrir | Anyway, here's a small thing for coordinating projects: http://linuxtesting.org/upstream-tracker/ |
| 14:04.57 | russellb | WIMPy: it's pretty full support, they worked really hard on it |
| 14:05.03 | WIMPy | Well, even if the manual says no, CCBS/CCNR has been working for me with the Snom on internal calls. |
| 14:05.36 | drmessano | Sure, Ubuntu is Debian based, but it's just too easy to miss something as small as an * when you're shuffling through 21 CDs |
| 14:05.39 | russellb | there is native SIP support, native ISDN support, and "generic" support to allow it to work on analog phones and such |
| 14:05.51 | russellb | CCBS/CCNR requests can go through multiple servers, across different protocols |
| 14:06.01 | russellb | works with forked call requests ... |
| 14:06.06 | russellb | it's awesome. |
| 14:06.22 | russellb | because putnopvut is kind of a big deal (he is one of the 2 that wrote it) |
| 14:06.25 | WIMPy | That'd surely be a major step. |
| 14:07.06 | putnopvut | my name was said! |
| 14:07.16 | russellb | yessir! |
| 14:07.17 | tzafrir | drmessano, that requires someone to actually maintain the Ubuntu package |
| 14:07.27 | WIMPy | Sounds like I have to move away from LCR again just when I liked it. |
| 14:07.53 | tzafrir | anyway, what do you guys use for an operator panel? |
| 14:08.02 | russellb | a 20" LCD? |
| 14:08.29 | tzafrir | and the software displayed in that panel? |
| 14:08.32 | drmessano | Apple Newton and a power brick |
| 14:08.47 | drmessano | AstAssistant is pretty cool |
| 14:08.55 | [sr] | WIMPy: it seems that i have to learn dahdi in a few month's also :p |
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| 14:39.16 | Katty | do-do-do |
| 14:44.40 | [TK]D-Fender | da-da-da-da |
| 14:44.46 | [TK]D-Fender | IS ALL I WANT TO SAY TO YOU! |
| 14:45.17 | [sr] | lol |
| 14:46.47 | *** join/#asterisk wcselby (~wcselby@208.180.112.123) |
| 14:46.48 | russellb | dances |
| 14:46.50 | wcselby | o/ |
| 14:47.00 | wcselby | glad I can make russellb dance just by joining |
| 14:47.32 | *** join/#asterisk asteriskATmarmuD (~Mundt@193.158.65.23) |
| 14:49.56 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 14:49.56 | wcselby | i've got a client who purchased a te420 last summer, but they're not sure of a specific date or vendor who they purchased it from. is there anyway I can look it up for them? |
| 14:51.07 | *** join/#asterisk Tim_Toady (~moi@178.128.52.9.dsl.dyn.forthnet.gr) |
| 14:52.31 | asteriskATmarmuD | hi guys! has anyone experience with a detailed call progress analysis with asterisk? I know there is an analysizer from sagoma, but we want open source |
| 14:55.59 | WIMPy | asteriskATmarmuD: Care to be more specific? What are you missing? |
| 14:58.12 | asteriskATmarmuD | WIMPy: sorry, asterisk gives detailed info according to the sip-messages received by our carrier, but we need to store the information to each number in a database. |
| 14:58.41 | asteriskATmarmuD | we are using an altered vicidial for that |
| 14:58.57 | *** join/#asterisk joeflyde (~user@97.104.194.220) |
| 14:59.48 | nonlinearly8 | Hi, I have installed asterisk 1.4.33.1 with zaptel 1.4.12.1 but I can't find the zapata.conf file... |
| 15:00.18 | asteriskATmarmuD | WIMPy: any experiece? problem is to distinguish between "no answer - because of no one picked up" and "no answer - because it is an unallocated number" both seem to be treated the same way... |
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| 15:02.15 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:02.53 | [TK]D-Fender | asteriskATmarmuD: You can tell that just with the time it dialed. |
| 15:03.15 | [TK]D-Fender | asteriskATmarmuD: if you hit the timeout there is a DIALSTATUS for that |
| 15:03.25 | WIMPy | _If_ they really are treated the same, that's a problem of your itsp. |
| 15:04.11 | *** join/#asterisk jtodd (gpi904ue5j@asterisk/community-director-and-tie-dye-shirt-lover/jtodd) |
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| 15:08.01 | asteriskATmarmuD | <PROTECTED> |
| 15:08.10 | asteriskATmarmuD | thanks guys |
| 15:08.53 | nonlinearly8 | Hi, I have installed asterisk 1.4.33.1 with zaptel 1.4.12.1 but I can't find the zapata.conf file... |
| 15:09.04 | *** join/#asterisk lordvadr (~something@jose-tc.ctc.biz) |
| 15:09.15 | asteriskATmarmuD | <PROTECTED> |
| 15:09.27 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-172.dslextreme.com) |
| 15:09.39 | WIMPy | asteriskATmarmuD: what's wrong with pri debug? or hangupcause? |
| 15:09.47 | nonlinearly8 | no I read a book that use these |
| 15:10.08 | lordvadr | is there a limit to the number of setvar's I can do in realtime sip? I've change the database, unregistered and reregistered a peer, and a variable isn't showing up in DumpChan |
| 15:10.18 | asteriskATmarmuD | <PROTECTED> |
| 15:10.29 | drmessano | nonlinearly8: Zaptel has been obsoleted by DAHDI.. you should be using DAHDI |
| 15:11.22 | asteriskATmarmuD | <PROTECTED> |
| 15:11.30 | nonlinearly8 | then I must throw away the book |
| 15:11.45 | asteriskATmarmuD | <PROTECTED> |
| 15:11.52 | p3nguin | nonlinearly8: You should have used dahdi. Go fix that. |
| 15:12.05 | *** join/#asterisk Corydon76-lap (~Corydon76@nat/digium/x-eikdqqedmrtshnze) |
| 15:12.06 | *** mode/#asterisk [+o Corydon76-lap] by ChanServ |
| 15:12.14 | nonlinearly8 | the book is the bible of Asterisk and then...? |
| 15:12.28 | [TK]D-Fender | nonlinearly8: No, it isn't |
| 15:12.31 | asteriskATmarmuD | <PROTECTED> |
| 15:12.45 | [TK]D-Fender | nonlinearly8: It is a 3rd party book that is a little old now, but good for general understanding |
| 15:13.02 | [TK]D-Fender | nonlinearly8: SPECIFICS have changed. Now go download and install DAHDI as per its instructions. |
| 15:13.06 | drmessano | [TK]D-Fender: But you MADE me buy it! |
| 15:13.12 | p3nguin | haha |
| 15:13.22 | [TK]D-Fender | drmessano: MASTER OF PUPPETS I'M PULLING YOUR STRINGS! |
| 15:13.24 | drmessano | [TK]D-Fender: You TOLD ME it was CURRENT |
| 15:13.28 | nonlinearly8 | if I will use dahdi then I will not something to drive me... I will be reading wrong thinks |
| 15:13.35 | nonlinearly8 | thinkgs |
| 15:13.35 | [TK]D-Fender | runs some more current through drmessano |
| 15:13.39 | drmessano | lol |
| 15:13.43 | asteriskATmarmuD | <PROTECTED> |
| 15:13.53 | wcselby | nonlinearly8 - DAHDI is basically the same as zaptel, except where it's different |
| 15:13.56 | wcselby | :) |
| 15:14.00 | nonlinearly8 | the books please |
| 15:14.08 | drmessano | nonlinearly8: You can also check the INTERNET. There's lots of stuff on the INTERNET about DAHDI |
| 15:14.21 | Corydon76-lap | DAHDI is just a rename. That's all. |
| 15:14.30 | drmessano | nonlinearly8: There's this new site, called GOOGLE, which lets you search the WWW pages |
| 15:14.37 | [TK]D-Fender | nonlinearly8: DAHDI is the NEW NAME for Zaptel. Not an "alternative". |
| 15:14.54 | asteriskATmarmuD | <PROTECTED> |
| 15:14.58 | nonlinearly8 | Ok and what file I must configure if the book says zapata.conf? |
| 15:15.06 | Corydon76-lap | chan_dahdi.conf |
| 15:15.10 | drmessano | http://www.voip-info.org/wiki/view/DAHDI |
| 15:15.12 | [TK]D-Fender | nonlinearly8: STOP READING THAT BOOK |
| 15:15.16 | asteriskATmarmuD | nonlinearly8: Asterisk 1.4 releases later than 1.4.21, and all releases of Asterisk 1.6, will automatically use DAHDI |
| 15:15.23 | [TK]D-Fender | nonlinearly8: And go follow the instructions in the DAHDI tarball |
| 15:15.32 | asteriskATmarmuD | nonlinearly8: read my link |
| 15:15.46 | asteriskATmarmuD | <PROTECTED> |
| 15:15.50 | drmessano | asteriskATmarmuD: Can you type it in a book and send it to me? |
| 15:16.05 | drmessano | ok, bbiaf |
| 15:16.09 | [TK]D-Fender | prints up a book... and then throws it at drmessano |
| 15:16.59 | WIMPy | You print out your larts? |
| 15:17.45 | nonlinearly8 | But I had a message in the logs files that said that chan_dahdi can not load zapata.conf!!! |
| 15:17.47 | p3nguin | They won't automatically use dahdi if zapata is installed instead. |
| 15:18.26 | asteriskATmarmuD | use the internet, I found nearly everything, book was only nice to lay back in a chair and read |
| 15:18.31 | asteriskATmarmuD | forums are nice too |
| 15:19.20 | nonlinearly8 | Internet is ok but do not drive you step-by step |
| 15:19.25 | *** join/#asterisk onats (~onats@unaffiliated/onats) |
| 15:19.43 | onats | hi, does anyone have a free sip server i can test against to test sip gateway/trunks? |
| 15:21.09 | [TK]D-Fender | [11:15]<[TK]D-Fender>nonlinearly8: And go follow the instructions in the DAHDI tarball <------------ |
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| 15:21.38 | nonlinearly8 | DAHDI tarball? |
| 15:21.54 | p3nguin | Have you heard anything we've said? |
| 15:21.58 | *** join/#asterisk joeflyde (~user@97.104.194.220) |
| 15:22.07 | *** part/#asterisk joeflyde (~user@97.104.194.220) |
| 15:22.24 | [TK]D-Fender | nonlinearly8: How is this a question? |
| 15:22.43 | nonlinearly8 | Sorry guys but I am new in this area and if something is wrong then I am lost |
| 15:23.03 | p3nguin | You're going to have to install dahdi. |
| 15:23.19 | asteriskATmarmuD | <PROTECTED> |
| 15:23.23 | nonlinearly8 | even if I have this version of asterisk? |
| 15:23.27 | *** join/#asterisk joeflyde (~user@97.104.194.220) |
| 15:23.31 | [TK]D-Fender | nonlinearly8: asterisk.og <- DOWNLOAD DAHDI AND READ THE INSTRUCTIONS. |
| 15:23.38 | wcselby | hey digium folk - i've got a client who purchased a te420 last summer, but they're not sure of a specific date or vendor who they purchased it from. is there anyway I can look it up for them? |
| 15:23.51 | [TK]D-Fender | wcselby: No. |
| 15:24.06 | [TK]D-Fender | wcselby: Digium can't tell when a reseller sold it to your client. |
| 15:24.17 | DuPie | nonlinearly8: using an older book that has non current information in it... and trying to apply it to a new asterisk install wont work too well either... |
| 15:24.21 | wcselby | [TK]D-Fender - :( ah well. |
| 15:24.33 | wcselby | [TK]D-Fender - i thought maybe they could look it up by serial or whatever |
| 15:24.44 | nonlinearly8 | I will download it but then the book is totally useless ("Asterisk the future of telephony")\ |
| 15:24.47 | *** join/#asterisk andresm (~andresm@ubuntu/member/andresmujica) |
| 15:24.51 | DuPie | old install + old book = buggy but works. new install+old book = good luck |
| 15:24.52 | wcselby | so, on that note, is there a way to get the serial number without opening the box? |
| 15:24.55 | [TK]D-Fender | wcselby: They could only tell you the date of manufacture, not the date it was SOLD |
| 15:24.56 | *** join/#asterisk joeflyde (~user@97.104.194.220) |
| 15:25.00 | asteriskATmarmuD | <PROTECTED> |
| 15:25.03 | [TK]D-Fender | wcselby: And no and that 2nd Q |
| 15:25.14 | wcselby | [TK]D-Fender - heh |
| 15:25.16 | andresm | hi everyone, is it possible to restrict incoming calls on a PRI with asterisk 1.4.x ??? |
| 15:25.23 | wcselby | [TK]D-Fender - a guy can hope |
| 15:25.26 | [TK]D-Fender | Andin what way? |
| 15:25.34 | [TK]D-Fender | andresm: in what way? |
| 15:25.35 | nonlinearly8 | DuPie you are write... |
| 15:25.42 | nonlinearly8 | right |
| 15:26.12 | wcselby | nonlinearly8 - there are some good guides out there for installing asterisk and dahdi. i've even got a guide on my blog for installing 1.6.2.x and DAHDI |
| 15:26.40 | DuPie | dahdi is easier to figure out than zaptel id say... |
| 15:26.51 | andresm | [TK]D-Fender: we want to "split" the 30 channels (i'm using E1) so the max amount of incoming calls is 15, this way i get 15 channels available for outgoing calls.. |
| 15:27.12 | [TK]D-Fender | andresm: You can't stop your carrier from sending you calls <- |
| 15:27.18 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 15:27.27 | [TK]D-Fender | andresm: You can HANG UP on them or IGNORE tehm if you want though. |
| 15:28.18 | WIMPy | andresm: Or you could get your telco to configure your line that way, but I wouldn't recommend doing so. |
| 15:28.24 | andresm | [TK]D-Fender: yes, that's what i thought... i cannot stop the incoming calls flow.... |
| 15:28.43 | *** part/#asterisk asteriskATmarmuD (~Mundt@193.158.65.23) |
| 15:29.04 | WIMPy | andresm: Just use Hangup if you've got too many calls. |
| 15:29.43 | [TK]D-Fender | andresm: tell your telco to change their behavior |
| 15:30.14 | nonlinearly8 | I think that I am misunderstood... Of course dahdi is better but I did not ask about dahdi.. I need to run my asterisk 1.4.34 with zaptel... |
| 15:30.28 | andresm | [TK]D-Fender: when you say HANGUP/IGNORE them you mean that i should route incoming calls arriving on upper channels to hangup.... my telco said that our pri doesn't support that feature... in fact they said that we should do that at our level.... |
| 15:30.31 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 15:30.31 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 15:31.00 | [TK]D-Fender | nonlinearly8: And you've done nothing to help us help you with what you ahve. |
| 15:31.03 | [TK]D-Fender | have* |
| 15:31.11 | AdvoWork | any reason why when i do sip show peers, it shows: (Unspecified) D N 0 Unmonitored i know its unmonitored, but why unspecified, and why no port? |
| 15:31.12 | DuPie | nonlinearly8: if you absolutely insist on using the book with zapetel, then you need to downgrade your version of asterisk |
| 15:31.14 | nonlinearly8 | or then I must use asterisk 1.4.21 and above |
| 15:31.16 | andresm | thanks WIMPy thanks [TK]D-Fender i'm gonna try to explain my customer that :) thanks! |
| 15:31.51 | nonlinearly8 | ok |
| 15:31.53 | [TK]D-Fender | AdvoWork: Its unmonitored because you didn't enable qualify, and it's Unspecified because they haven't registered. |
| 15:32.34 | DuPie | either use newer asterisk+dahdi, or use older asterisk+zaptel. older is depreciated and perhaps buggy, but it can work. but if you run into a problem expect people to recommend upgrading. dahdi solves issues in zaptel |
| 15:32.35 | *** join/#asterisk t_dot_zilla (~chatzilla@firewall-a.buf.ny.i-evolve.net) |
| 15:32.36 | AdvoWork | [TK]D-Fender, it was working fine, then it went to NR, reboots and reregistered didnt fix it, so i changed it to qualify no, but same problem.. |
| 15:32.51 | t_dot_zilla | whats does notifyhold in sip.conf do |
| 15:33.03 | [TK]D-Fender | AdvoWork: qualify =no will leave it as "unmonitored" |
| 15:33.13 | [TK]D-Fender | t_qwhat does the sample config tell you? |
| 15:33.19 | [TK]D-Fender | t_dot_zilla: what does the sample config tell you? |
| 15:33.20 | t_dot_zilla | it doesn't |
| 15:33.28 | t_dot_zilla | i mkean i don't have the sample config |
| 15:33.35 | AdvoWork | [TK]D-Fender, i know, but it still won't register, i changed the qualify as a test, but its still unspecified, i cant work out why? |
| 15:34.14 | t_dot_zilla | for some reason when a call is transfered the phone that transfered the call appears to be busy and will not recieve incoming, i read if you set notifyhold=no it will resolve the issue |
| 15:34.24 | t_dot_zilla | but will it affect any other portion of the system? |
| 15:34.34 | [TK]D-Fender | [11:33]<t_dot_zilla>i mkean i don't have the sample config <- You should. |
| 15:34.49 | *** join/#asterisk raden_work (~jon@69-179-99-17.stat.centurytel.net) |
| 15:35.07 | [TK]D-Fender | t_dot_zilla: this has nothing to do with call handling |
| 15:35.15 | nonlinearly8 | I wonder how in the asterisk's webinars suggest a book that has huge problems with versions of asterisk and zaptel. thanks anyway |
| 15:35.25 | [TK]D-Fender | AdvoWork: Qualify has no impact on it not registering. |
| 15:35.40 | [TK]D-Fender | nonlinearly8: What webinar? |
| 15:35.41 | p3nguin | nonlinearly8: I wish you would stop crying and use dahdi already. |
| 15:35.46 | AdvoWork | [TK]D-Fender, how can i work out why its not registering then? It was fine earlier, no changes have been made at all |
| 15:35.59 | [TK]D-Fender | nonlinearly8: and so far I don't see you doing anything about your problem here... I guess you don't want it fixed... |
| 15:36.11 | [TK]D-Fender | AdvoWork: Look at the SIP DEBUG <- |
| 15:36.33 | [TK]D-Fender | AdvoWork: and "I changed nothing" is useless commentary in debugging problems. Never helps. |
| 15:36.46 | AdvoWork | [TK]D-Fender, ive put qualify back to normal, and its now: (Unspecified) D N 0 UNKNOWN |
| 15:36.57 | p3nguin | If nothing changed, wouldn't it still be working? |
| 15:37.04 | AdvoWork | [TK]D-Fender, ok, wheres the sip debug? |
| 15:37.09 | [TK]D-Fender | AdvoWork: What part of "qualify has nothing to do with registration" are you not understanding? |
| 15:37.20 | [TK]D-Fender | AdvoWork: Enable SIP DEBUG at * CLI and wathc for it |
| 15:37.47 | [TK]D-Fender | p3nguin: Yesterday I met somebody exactly like you ... only different |
| 15:37.49 | AdvoWork | [TK]D-Fender, i didnt say it did have anything to do with it, but seeing as thats not my problem ive revered it back |
| 15:38.09 | [TK]D-Fender | AdvoWork: Feel free to tell us more things that are not related to your problem... |
| 15:38.26 | AdvoWork | :p |
| 15:38.39 | AdvoWork | ive never used this sip debug before, does it monitor and debug all extensions or? |
| 15:38.51 | p3nguin | It debugs SIP. |
| 15:38.51 | nonlinearly8 | I am not crying... but I see that asterisk has the same illnesses that have all open source projects...lack of Weights and Measures |
| 15:39.00 | *** part/#asterisk nonlinearly8 (~nonlinear@relate.ath.forthnet.gr) |
| 15:39.22 | [TK]D-Fender | BAI BAI |
| 15:39.26 | [TK]D-Fender | twit ... |
| 15:39.44 | Kobaz | heh |
| 15:39.56 | p3nguin | He probably thinks Asterisk will never survive without him. |
| 15:40.13 | outtolunc | whats Asterisk? |
| 15:40.16 | *** part/#asterisk joeflyde (~user@97.104.194.220) |
| 15:40.16 | [TK]D-Fender | p3nguin: It still won't, but it'll die a little happier ;) |
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| 15:41.39 | AdvoWork | [TK]D-Fender, ok, enabled that, can i fine tune the debug info? i can't see anything related to that sip, ive tried reregistering it |
| 15:42.30 | [TK]D-Fender | AdvoWork: then perhaps you have a networking issue you should be looking at. Or you are pointing a device the wrong way. Or are firewalled. Or another routing problem. |
| 15:42.39 | [TK]D-Fender | AdvoWork: If you see nothing, then * is getting nothing |
| 15:42.54 | [TK]D-Fender | AdvoWork: And no amount of * configuration is going to change that |
| 15:43.03 | p3nguin | To see what fine-tuning options are present, type in sip set debug then a space, then the tab key. |
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| 15:43.56 | AdvoWork | [TK]D-Fender, ok thankyou for the info, if no configuration has changed, then as you say, i beleive a network problem. A bit unsure how to go about diagnoising that, but thats another problem. can i just do Disable SIP DEBUG ? |
| 15:44.09 | [TK]D-Fender | Never restrict your search when you can't find something. you will leave off things you might want to see |
| 15:44.18 | p3nguin | Or like I do: si tabkey, se tabkey, d tabkey, tabkey. |
| 15:44.22 | *** join/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 15:44.22 | [TK]D-Fender | AdvoWork: "help sip" <- |
| 15:44.30 | andresm | [TK]D-Fender: one more question, incoming calls start at dahdi 1 and go up, and outgoing calls start at last channel and go down, right?? |
| 15:44.55 | [TK]D-Fender | Andthey come in the way the telco chooses to send the, and they go out the YOU tell * to send them |
| 15:45.20 | WIMPy | andresm: Depends on your Telcos configuration and depends on your configuration. |
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| 15:51.58 | andresm | ok.. thks.. |
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| 15:57.16 | aster1sk | p3nguin - you have had great success with sccp I hear |
| 15:57.29 | *** join/#asterisk BCS-Satori (~BCS-Sator@75-148-21-113-WashingtonDC.hfc.comcastbusiness.net) |
| 15:57.33 | wcselby | ~pb |
| 15:57.34 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood 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://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 15:59.40 | wcselby | question - http://asterisk.pastey.net/138698 - what does this line mean in my dmesg output? wctdm24xxp 0000:03:08.0: VPMADT032 Failed! Unable to ping the DSP (2)! If it makes a difference, I don't have the FXO lines plugged in yet |
| 16:01.43 | BCS-Satori | If I change the "secret =" in users.conf for the phone registration in asterisk 1.4 to anything other then the extension (i.e.: exten 601 the secret would be 601) I am unable to retrieve voicemail even though the vmsecret is defined. I have tried specifying the voicemail password in users.conf as well as voicemail.conf. Any reason why it would not let me check voicemail? |
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| 16:25.11 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.167.9.dsl.dyn.forthnet.gr) |
| 16:25.19 | Ad-Hoc | hi ppl |
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| 16:40.51 | [TK]D-Fender | BCS-Satori: What do you expect us to say since you aren't showing us anything? |
| 16:42.12 | BCS-Satori | [TK]D-Fender: What do you want to see? |
| 16:42.23 | [TK]D-Fender | BCS-Satori: How about something useful? |
| 16:44.41 | *** join/#asterisk asteriskATmarmuD (~Mundt@193.158.65.23) |
| 16:45.49 | asteriskATmarmuD | asterisk 1.8 will distinguish between dialstatuses more detailed. now every "error" results in congestion |
| 16:46.03 | asteriskATmarmuD | when will asterisk 1.8 be released? |
| 16:46.08 | *** join/#asterisk BANSAL (~bansal@117.199.120.179) |
| 16:46.21 | Qwell | asteriskATmarmuD: when it's ready. |
| 16:46.57 | *** join/#asterisk saintk (~saintk2@5354D2D3.cable.casema.nl) |
| 16:47.03 | asteriskATmarmuD | this is really a great problem for me in germany, we want to have detailed statuses within our system. hangupcause may be the solution for now, am I right |
| 16:47.09 | asteriskATmarmuD | <PROTECTED> |
| 16:48.43 | pabelanger | soon(TM) ? |
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| 17:15.47 | ruben23 | hi anyone used zoiper on a linux client for callsing with asterisk..? |
| 17:19.00 | *** part/#asterisk kurkowski (~felipe@189.62.199.201) |
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| 17:24.26 | beek | ruben23: About a year ago. It worked fine. |
| 17:26.40 | bent_screwdriver | is there a way to insert a pause after a digit when calling out through a dahdi channel? i.e. 9#1112223333 |
| 17:27.00 | carrar | w ? |
| 17:27.18 | ruben23 | beek:i used it on ubuntu-desktop. ans it detectde an anlaog device whihc i used and it work, problme after a minute of calling it suddenly changes to digitial device which dont work. |
| 17:27.19 | [TK]D-Fender | ^^^ |
| 17:27.36 | bent_screwdriver | carrar: does that represent a second? |
| 17:27.45 | [TK]D-Fender | bent_screwdriver: .5 |
| 17:27.54 | bent_screwdriver | thanks! |
| 17:28.09 | carrar | anytime I can save you from googling, just let me know! |
| 17:28.37 | Corydon76-lap | only works on analog-signalled channels, i.e. not PRI |
| 17:28.44 | bent_screwdriver | in a pinch and didn't see it on first goog glance and had channel open. thx |
| 17:29.28 | bent_screwdriver | damn, i need it on PRI as we are having a weird issue (centrex pri) where it is now requiring about 5 second wait, after sending the 9, before it will accept the rest of the digits. |
| 17:30.19 | bent_screwdriver | ticket open with telco but figured i could put in a quick fix with a pause....it is a county wide issue so they may have it fixed soon.... |
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| 17:38.32 | p3nguin | After sending the 9? Why would a phone company need a 9 to dial a number not starting with a 9? |
| 17:39.27 | bent_screwdriver | it is centrex. we are migrating away from this but it is a service that lies on the PRI so we can still call old centrex extensions, and the service requires sending the 9 down the pipe for outbout calls. |
| 17:39.51 | carrar | Thats HOT++ |
| 17:40.30 | p3nguin | If the extension numbers started with 9, it would make sense. 9, pause, extension number, however, does not. |
| 17:41.04 | *** join/#asterisk residentgrey (~residentg@pool-71-191-49-141.washdc.fios.verizon.net) |
| 17:42.07 | bent_screwdriver | anything we send down the pri, that is 4 digits, talks to the legacy centrex extensions. Any outbound calls on the PRI, when you have Centrex, have to start with 9. It is like a managed PBX. We have some 1FB centrex lines and when you pick up the phone, you get a secondary dialtone when you press 9. |
| 17:42.58 | bent_screwdriver | we are moving away from centrex but we are keeping the service until we migrate away from it, for backwards compatibility. |
| 17:43.03 | p3nguin | Wonderful OLD technology is wonderful. |
| 17:43.31 | bent_screwdriver | tell me about it. it has been here in this organization longer than i've been alive. |
| 17:43.49 | bent_screwdriver | asterisk is starting to take over though.... ;) |
| 17:46.57 | t_dot_zilla | bent_screwdriver: freeswitch is taking over and will dominate |
| 17:48.11 | Kobaz | t_dot_zilla: i don't know... asterisk is a lot more popular currently |
| 17:48.15 | Qwell | taking over what exactly? they have completely different goals. |
| 17:48.40 | Kobaz | and, ask any of the freeswitch people about unit testing |
| 17:48.57 | Kobaz | they say, our platform is solid, yes we do qa testing |
| 17:49.38 | t_dot_zilla | freeswitch is much more comprehensive |
| 17:49.46 | Kobaz | but you repeat the question, and ask specifically about unit testing... and no one will tell you anything... so it seems to me they don't do any, the ones I talked to, don't know |
| 17:49.49 | russellb | rolls eyes at the troll |
| 17:50.07 | Kobaz | t_dot_zilla: i would say asterisk is more comprehensive |
| 17:50.15 | Kobaz | t_dot_zilla: comparing feature by feature, module by module |
| 17:50.22 | Kobaz | but to each their own |
| 17:50.38 | t_dot_zilla | i don't like asterisk's agi |
| 17:50.45 | Kobaz | so don't use it |
| 17:50.46 | t_dot_zilla | i like freeswitch's api |
| 17:50.51 | moy | t_dot_zilla: why are you here then? |
| 17:50.58 | t_dot_zilla | i work with asterisk |
| 17:51.07 | p3nguin | qwell: What is the goal of each? |
| 17:51.18 | t_dot_zilla | pbx |
| 17:51.26 | Qwell | Incorrect answer. Neither are a PBX. |
| 17:51.38 | Qwell | both *CAN* be a PBX. but that is not the goal of either. |
| 17:51.47 | p3nguin | What are the goals? |
| 17:51.50 | Kobaz | p3nguin: asterisk is an application development platform... freeswitch's goal is to be a featureful softswitch |
| 17:51.50 | t_dot_zilla | what is the goal then? |
| 17:51.59 | Qwell | Kobaz: pretty much |
| 17:52.01 | Katty | HAI GUYS |
| 17:52.07 | Kobaz | oo it's katty |
| 17:52.08 | jblack | Hi |
| 17:52.12 | t_dot_zilla | freeswitch is also and aplication development platform |
| 17:52.15 | *** part/#asterisk andresm (~andresm@ubuntu/member/andresmujica) |
| 17:52.17 | t_dot_zilla | that is featureful |
| 17:52.22 | Katty | hugs Kobaz |
| 17:52.24 | Katty | hugs jblack |
| 17:52.33 | Kobaz | t_dot_zilla: but that's not freeswitch's goal |
| 17:52.45 | p3nguin | What are the goals? |
| 17:52.48 | Kobaz | so you'll see a lot more app platform type features on asterisk than you will freeswitch |
| 17:52.56 | carrar | stands in the hug line |
| 17:52.58 | Kobaz | p3nguin: featureful softswitch |
| 17:52.59 | t_dot_zilla | that is among Freeswitch's many goals |
| 17:53.02 | Qwell | t_dot_zilla: Take the cheerleading to #freeswitch where it belongs. |
| 17:53.10 | Qwell | cuts to the front of Katty's line |
| 17:53.29 | russellb | trips Qwell |
| 17:53.36 | russellb | now I'm first |
| 17:53.56 | Katty | hugs carrar |
| 17:54.01 | carrar | YHATZEE!!! |
| 17:54.08 | Katty | ^_- |
| 17:54.10 | Qwell | russellb: apparently we both fail |
| 17:54.11 | Katty | orly. |
| 17:54.12 | Kobaz | but neither of you realize i stole her when she first got in :P |
| 17:54.19 | beek | hugs katty |
| 17:54.22 | Katty | :> |
| 17:54.25 | Katty | hugs bent_screwdriver |
| 17:54.26 | Katty | oh |
| 17:54.28 | moy | Kobaz: Although, I don't want to feed the troll, "FreeSWITCH is a scalable open source cross-platform telephony platform designed to route " ... so I agree with you, I hate comparisions when come from ppl that know nothing about either project, but your assumption of freeswitch goals is incorrect, from my point of view |
| 17:54.29 | Katty | hugs beek, too |
| 17:54.33 | Katty | bent_screwdriver: disregard. |
| 17:54.36 | beek | :D |
| 17:54.44 | Katty | i feel so loved :> |
| 17:54.48 | Qwell | doesn't |
| 17:55.13 | Katty | gives Qwell a bottle of Germ-X |
| 17:55.15 | beek | And you are. |
| 17:55.33 | Katty | hugs Qwell |
| 17:55.40 | t_dot_zilla | moy: i've worked with both freeswitch and asterisk, i found it much easier to develop an application based on FS |
| 17:55.59 | Kobaz | moy: the original goal was to make a brand new platform that didn't have some of the coding shenanagins as asterisk does. so they have one thread per channel, and don't need to do masqurades and other weirdness |
| 17:56.06 | moy | good for you, I've worked and developed on both, and find their uses for both |
| 17:57.21 | moy | for starters, anyone using south-america R2, can't really do much with freeswitch right now |
| 17:57.36 | moy | they need to stick with asterisk, and they're happy with it |
| 17:57.56 | moy | most of them anyways :) |
| 17:58.05 | coppice | moy: I thought you had R2 running properly now |
| 17:58.27 | moy | well, not production ready, since there has not been much interest in testing it |
| 17:58.49 | Kobaz | moy: i suppose after time asterisk/freeswitch will be nearly identical in features |
| 17:58.54 | coppice | you expect actual cooperation? good heavens |
| 17:59.05 | Kobaz | kind of like the linux/bsd slices |
| 17:59.29 | Kobaz | you can basically do the same things on either system, the details are just different |
| 18:00.07 | Kobaz | which ever platform gets more unit testing will be winner in the end |
| 18:00.20 | Kobaz | for developers anyway... popularity comes down to marketing |
| 18:00.41 | moy | Kobaz: so it's a zero sum game? |
| 18:01.31 | *** join/#asterisk cuco (~elcuco@109.253.41.99) |
| 18:01.58 | t_dot_zilla | Kobaz: unless asterisk begins developing more features, FS will always have more |
| 18:02.02 | Kobaz | moy: it's good having diversity in the marketplace... monoculture is never good |
| 18:02.09 | cuco | tzafrir: 2.18 2.08 1.61 1/379 5762 |
| 18:02.27 | Kobaz | t_dot_zilla: depends on developer interest |
| 18:02.39 | Kobaz | t_dot_zilla: nothing is an absolute certainty |
| 18:02.41 | moy | coppice: nah, I understand I need to get it to work to ppl start using it, but I am expecting to see someone complaining about it not working and then take it from there :) |
| 18:03.04 | moy | should work for basic calls, but has no collect call blocking and such |
| 18:03.10 | *** join/#asterisk Tobarja (~chatzilla@204.31.133.118) |
| 18:03.18 | moy | and the thread model is crap |
| 18:03.51 | coppice | all thread models are crap. some are just worse than others |
| 18:03.55 | Kobaz | heh |
| 18:03.58 | russellb | coppice: +1 :-) |
| 18:04.07 | moy | lol |
| 18:04.24 | moy | crappier I should've said then |
| 18:04.33 | *** join/#asterisk DelphiWorld (~Delphi@41.200.18.158) |
| 18:05.32 | Kobaz | i would be interested in an up-to-date freeswitch vs asterisk page through |
| 18:05.34 | Kobaz | tough |
| 18:05.40 | Kobaz | ...though |
| 18:05.51 | Kobaz | the one they have up there is from 2008 |
| 18:05.58 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 18:06.27 | p3nguin | FS published it? |
| 18:06.29 | kurkowski | is it possible to run an agi script AFTER the dial? (say I want to read DMTF digits from a called party) |
| 18:06.35 | Kobaz | p3nguin: yeah |
| 18:06.50 | p3nguin | That must have been the last time they were comfortable making a public comparison. |
| 18:06.54 | Kobaz | p3nguin: the one on fs.org is basically complaining how crappy asterisk pre 1.0 was |
| 18:06.55 | cuco | tzafrir: 0.71 1.34 1.42 3/356 5821 <- I killed firefox |
| 18:07.21 | Kobaz | of course it's going to be crappy, it was in it's birth stage |
| 18:07.42 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 18:17.11 | *** join/#asterisk javar (~javier@186.83.27.216) |
| 18:17.32 | tzafrir | cuco, killing pandas? That's wrong |
| 18:17.57 | p3nguin | Why does app_chanspy output the same line two times on the CLI when attaching to a channel? Wouldn't one time be sufficient? |
| 18:18.10 | *** join/#asterisk sekil (~Ognjen@78.24.111.215) |
| 18:18.14 | *** join/#asterisk jhirley (~jhirley@c-75-74-13-194.hsd1.fl.comcast.net) |
| 18:23.16 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 18:24.28 | cuco | tzafrir: yes, and now i cannot even blame it on fedora. |
| 18:24.47 | wcselby | . |
| 18:25.10 | wcselby | fxo cards will have inbound calls ringing to callerid or to "s" extension in my incoming context? |
| 18:25.21 | wcselby | well, the context i define in chan_dahdi.conf I mean |
| 18:25.32 | cuco | tzafrir: temp was 60c, now is 49c, loadav is about 0.22.... FF sux power from my machine |
| 18:26.07 | tzafrir | wcselby, yes |
| 18:26.21 | wcselby | tzafrir - very zen answer |
| 18:26.28 | *** part/#asterisk javar (~javier@186.83.27.216) |
| 18:26.34 | tzafrir | wcselby, very zen question |
| 18:26.39 | wcselby | lol |
| 18:27.01 | wcselby | will they route to the "s" exten in my incoming context, or to the exten that matched my inbound callerid? |
| 18:27.07 | wcselby | erm |
| 18:27.09 | wcselby | inbound did |
| 18:29.25 | malcolmd | wcselby: inbound DID isn't something that's supported across FXO's |
| 18:29.41 | wcselby | malcolmd - thanks, that's what I thought, but figured i'd doublecheck |
| 18:29.48 | malcolmd | wcselby: np |
| 18:29.59 | malcolmd | :q |
| 18:30.05 | malcolmd | whoops, sorry :D |
| 18:30.17 | *** part/#asterisk kurkowski (~felipe@189.62.199.201) |
| 18:32.48 | *** part/#asterisk rafael-ec (~rafael@200.110.234.162) |
| 18:39.54 | *** join/#asterisk cracoucas (cracoucas@dispo-82-65-228-100.adsl.proxad.net) |
| 18:40.37 | cracoucas | Hi everybody , is it possible to set the callerID in asterisk for outgoing calls? |
| 18:40.48 | Kobaz | core show function CALLERID |
| 18:41.22 | cracoucas | Kobaz? |
| 18:41.28 | Kobaz | hi |
| 18:42.18 | cracoucas | did you answer my question ? |
| 18:42.23 | Kobaz | yes |
| 18:43.33 | p3nguin | also core show application Set |
| 18:44.46 | *** join/#asterisk kn0x (~pinochle@67.159.48.101) |
| 18:46.37 | [TK]D-Fender | [14:40]<cracoucas>Hi everybody , is it possible to set the callerID in asterisk for outgoing calls? <- Yes.. also if your technology and provider ALLOW you to. |
| 18:49.31 | kn0x | seems like this started happening when I switched from unix-socket to a tcp connection for mysql (cdr)... ERROR[2693]: cdr_addon_mysql.c:334 mysql_log: Failed to insert into database: (1136) Column count doesn't match value count at row 1 |
| 18:50.43 | *** join/#asterisk [Outcast] (~anonymous@64.202.62.5) |
| 18:52.46 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 18:57.19 | kn0x | there was a patch from Corydon76-lap in bug 14002 but that was for 1.6.1-rc2 |
| 18:57.30 | kn0x | im on addons-1.6.1.4 |
| 18:57.38 | *** join/#asterisk rrb3942 (~rbullock@208.34.96.186) |
| 18:57.54 | wcselby | 1001 => 8888,Name Here,emailhere@em.com |
| 18:57.59 | wcselby | um.... |
| 18:58.05 | wcselby | wrong window, obviously |
| 19:01.16 | Corydon76-lap | kn0x: 1.6.1 is no longer maintained |
| 19:02.08 | *** join/#asterisk adam0563 (ad0eee46@gateway/web/freenode/ip.173.14.238.70) |
| 19:03.54 | adam0563 | I'm having some trouble with a snom m3 not being able to hear the people talking on the other end. I have it set up over NAT, so that the public IP address and port 5063 get forwarded to the base station, and the phone rings and I can hear the person on the line, and the only problem seems to be hearing people on the snom m3. I have some Cisco 7960's set up and they can hear just fine. Does anybody have any ideas? Thanks |
| 19:04.09 | Kobaz | adam0563: nat problems |
| 19:04.10 | *** join/#asterisk sawgood (~sawgood@173-13-158-25-sfba.hfc.comcastbusiness.net) |
| 19:06.05 | adam0563 | Kobaz: I figured that, I'm wondering if there are any specific settings or changes that you or anybody else knows of that I can try. |
| 19:10.30 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 19:14.40 | *** join/#asterisk SiNGLer (~singler@78-60-54-125.static.zebra.lt) |
| 19:15.28 | *** join/#asterisk mcrownover (~markcrown@remote.gawest.com) |
| 19:21.50 | sawgood | exten => _1NXXNXXXXXX/2102,1,Set(CALLERID(num)=2095551214) |
| 19:22.02 | sawgood | can someone tell me what the 2102 in this dialplan statement represents? |
| 19:22.19 | [TK]D-Fender | sawthe callerID of the caller |
| 19:22.21 | Kobaz | adam0563: make sure nat=yes in the sip cfg for the device |
| 19:22.44 | [TK]D-Fender | ~sipnat |
| 19:22.45 | infobot | sipnat is probably Quick guide on configuring Asterisk + 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 |
| 19:22.47 | [TK]D-Fender | adam0563: ^^^^ |
| 19:23.33 | sawgood | [TK]D-Fender: so, in this example, the statement only matches IF the caller ID is 2102 (and then it would change the caller ID to the statement on the end? |
| 19:23.46 | [TK]D-Fender | sawgood: correct |
| 19:24.24 | sawgood | Well, what is I want 'everyone's' caller ID to change to what is on the far end? |
| 19:24.25 | *** join/#asterisk nny (~Scott@cpe-071-076-058-253.sc.res.rr.com) |
| 19:24.37 | sawgood | how do I change 2101 to be 'everyone' |
| 19:24.50 | nny | any easy way to prune cli for a specific peer/ context? right now too much traffic to isolate a specific issue |
| 19:25.43 | [TK]D-Fender | sawgood: Don't use it at all |
| 19:26.22 | sawgood | exten => _1NXXNXXXXXX,1,Set(CALLERID(num)=2095551214) |
| 19:26.29 | sawgood | Would that be correct then? |
| 19:26.41 | [TK]D-Fender | sawgood: Dependsing if you want that to apply to all, yes |
| 19:27.21 | sawgood | Yes, I would like all outgoing calls to use that callerID(num) |
| 19:27.25 | sawgood | I'll try it now |
| 19:28.45 | nny | hmm maybe an easier question is there ways to limit what CLi shows (errors, warnings) vs. core set verbose X |
| 19:29.28 | sawgood | nny: sure ... /etc/asterisk/logger.conf |
| 19:29.57 | nny | er yeah, but was hoping for something from CLI, rather than changing parameters and reloading |
| 19:30.04 | nny | VERy hard to parse a busy system |
| 19:30.16 | sawgood | well, it dumps to a log file ... |
| 19:30.26 | sawgood | you could run grep or tail on the log file |
| 19:30.41 | nny | yeah, just hoping for realtime methods vs. verbose = x |
| 19:30.52 | sawgood | tail -f is realtime, right? |
| 19:30.59 | nny | so is cli lol |
| 19:31.04 | nny | you're missing the point my friend |
| 19:31.23 | nny | trying to watch one call when 70 are happening at the same time is a bit daunting |
| 19:31.29 | sawgood | no, you might be confusing CLI output of 'verbose' vs Asterisk logging |
| 19:31.47 | nny | no, I am trying to watch the complete process of one call in the CLI |
| 19:31.54 | *** join/#asterisk bijit (~bijit@186.4.3.18) |
| 19:32.09 | sawgood | I understand that you are trying to do, and I would do it with either SIP debug or the logging process |
| 19:32.16 | sawgood | I do what you are trying to do quite often |
| 19:34.50 | *** join/#asterisk bijit (~bijit@186.4.3.18) |
| 19:38.33 | *** join/#asterisk wnunes (~IceChat7@201-95-97-169.dsl.telesp.net.br) |
| 19:40.07 | *** join/#asterisk kerx (~kerx@38.118.129.34) |
| 19:42.18 | nightwalk | Any thoughts on using port forwarding to handle SIP connections to asterisk from clients on networks where the system doing the forwarding isn't necessarily the default gateway? Is there anything in the payload that would cause it to not work? Keep in mind also that the destination asterisk server in this scenario knows how to reach the clients directly, so there wouldn't be any confounding NAT or mangling issues -- just clients that don't necessar |
| 19:45.46 | nightwalk | Actually, I mis-spoke. /port forward/DNAT/ |
| 19:53.09 | exothermc | When bringing a call in SIP and sending it out TDM, how are you suppose to setup the dial plan? I tried to just jump right into a dial command, but then I had a bunch of hung RTP channels pile up. That problem seems to be solved if I answer() the call first, but that doesn't seem like the right thing to do. |
| 19:53.17 | *** join/#asterisk evil_gordita (~evilgordi@ip70-188-50-186.rn.hr.cox.net) |
| 19:54.09 | *** join/#asterisk MiserySoft (~Lee@nat65.mia.three.co.uk) |
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| 19:58.52 | t_dot_zilla | nny: * needs a log by call implementation |
| 19:59.10 | t_dot_zilla | or a log file per call |
| 20:01.53 | raden_work | where Katty hiding ? |
| 20:04.54 | Naikrovek | she lurks |
| 20:05.04 | Naikrovek | been through some chizz lately |
| 20:05.20 | Naikrovek | and by "chizz" i mean "shit" |
| 20:17.48 | paulc | "my job is chizz" - I like it.. sounds like it good be a good thing, even though it isn't :) |
| 20:19.27 | *** join/#asterisk andres833 (~andres833@200.119.41.207) |
| 20:22.02 | Gugge | when dialing Local/xxx@yyy/n from a queue, can i somehow get the CHANNEL(language) var set on the Local channel? |
| 20:22.07 | *** join/#asterisk wnunes (~IceChat09@201-95-97-169.dsl.telesp.net.br) |
| 20:23.12 | [TK]D-Fender | Gugge: Yes.. its there just like in every other call |
| 20:25.17 | Gugge | [TK]D-Fender: if i set it before calling Queue() its not set in the Local channel. Ill make an example :) |
| 20:26.57 | *** join/#asterisk wnunes (~IceChat09@201-95-97-169.dsl.telesp.net.br) |
| 20:27.27 | Naikrovek | wow 20mbit synchronous for $1400/mo |
| 20:27.31 | Naikrovek | landed some choice bandwidth |
| 20:28.40 | cracoucas | hi dfender. can you talk to me in private |
| 20:28.48 | [TK]D-Fender | cracoucas: What for? |
| 20:28.58 | [TK]D-Fender | cracoucas: Generally PM is reserved for my clients.. |
| 20:29.13 | [TK]D-Fender | cracoucas: And you've have questions answered several times very directly here.... |
| 20:29.57 | *** join/#asterisk Pimmetje (~Pimmetje@83.119.156.35) |
| 20:31.14 | cracoucas | d-fender I have a new question |
| 20:32.10 | *** join/#asterisk dexteruk (~dexteruk@79.100.245.112) |
| 20:32.37 | dexteruk | Hi I have a strange problem and i cannot find the reason behind it |
| 20:32.44 | [TK]D-Fender | <PROTECTED> |
| 20:33.12 | dexteruk | i am running asterisk 1.4.21.2 |
| 20:33.30 | dexteruk | and i can registers from some clients without an issue |
| 20:33.44 | dexteruk | but other sip clients on the network cannot register |
| 20:34.11 | dexteruk | i have tested form another site location using the same username and password it works fine |
| 20:34.37 | *** join/#asterisk nicoAMG (~nicoamg@201.237.49.131) |
| 20:35.08 | *** join/#asterisk mrchrisadams (~Adium@78-105-1-158.zone3.bethere.co.uk) |
| 20:35.12 | dexteruk | does anyone have a reason why some clients on a network could register |
| 20:35.16 | dexteruk | and other could not |
| 20:35.20 | cracoucas | d fender I am speaking to you in private |
| 20:35.54 | *** join/#asterisk [sr] (~Unknowned@pal-213-228-140-150.netvisao.pt) |
| 20:36.02 | [sr] | hi :) |
| 20:36.43 | cracoucas | d fender? |
| 20:37.21 | cracoucas | tk dfender? |
| 20:40.15 | *** join/#asterisk wnunes (~wnunes@201-95-97-169.dsl.telesp.net.br) |
| 20:40.53 | Naikrovek | he doesn't like private messages, and he'll be back in about 20 minutes |
| 20:41.10 | cracoucas | who know asterisk for java? |
| 20:41.32 | Naikrovek | asterisk for java? |
| 20:41.40 | Naikrovek | AMI client? |
| 20:41.49 | cracoucas | yes the binding of asterisk for java |
| 20:41.55 | Naikrovek | i've used it a bit |
| 20:42.37 | cracoucas | naik do you know if I can emit an outgoing call with any callerID I want? |
| 20:42.48 | Naikrovek | what are you up to |
| 20:42.54 | Naikrovek | depends on your provider |
| 20:42.59 | Naikrovek | if you can manipulate caller id |
| 20:43.02 | florz | cracoucas: no, only with authentified numbers |
| 20:43.02 | Gugge | http://paste2.org/p/920554 <- i set the language before calling Queue(), and in my Local part, but the announcements are still using the default 'en' language. It seems like i can only set the language on the Local.....;2 channel, but its using the language from the Local....;1 channel to play the announcement |
| 20:43.51 | cracoucas | florz i have seen the function setcallerID it must be useful to choose the callerID |
| 20:44.17 | florz | cracoucas: then WHAT THE HELL ARE YOU ASKING? |
| 20:44.21 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 20:44.44 | cracoucas | florz is it easy to choose the callerID ? |
| 20:45.03 | Naikrovek | you're already doing it |
| 20:45.16 | florz | cracoucas: no, you have to study for at least 5 years to gather experience |
| 20:45.21 | Naikrovek | why don't you try setCallerId() and see for yourself if it works |
| 20:45.23 | Gugge | cracoucas: most providers strip the callerid and replace it with the number you own |
| 20:45.33 | Naikrovek | cracoucas: ^^^^^^^^^^ |
| 20:46.01 | cracoucas | gugge I hope my provider let me choose any callerID |
| 20:46.28 | cracoucas | Gugge how can I know? |
| 20:46.35 | Gugge | if you have to ask here ... theres like 150% chance they dont |
| 20:46.47 | Naikrovek | TRY IT |
| 20:46.51 | Naikrovek | if it doesn't work, they won't let you |
| 20:46.52 | cracoucas | ok I will try |
| 20:46.57 | Naikrovek | if it does, they WILL let you |
| 20:47.06 | cracoucas | I use java for asterisk |
| 20:47.16 | Naikrovek | but there is a VERY slim chance they will let you do anything you want with caller id |
| 20:47.25 | WIMPy | How's the dog? |
| 20:48.07 | cracoucas | Naik so how is it possible that some company offer fake ID calling? |
| 20:49.13 | Gugge | cracoucas: some providers allow it |
| 20:49.22 | Gugge | cracoucas: but not on normal private connections |
| 20:49.41 | bent_screwdriver | i had to ask mine to remove the screening table and ammend our contract |
| 20:49.59 | cracoucas | Gugge tahnk you for your clear explanation |
| 20:50.10 | bent_screwdriver | this was needed so when a user forwarded their phone it would show the original caller-id |
| 20:51.38 | bent_screwdriver | by default they only let me send a number for which i had a DID ported. i am talking about a PRI of course. |
| 20:51.38 | path | is it possible to handle or modify SIP headers ? |
| 20:51.45 | path | I need to send a 180 SIP |
| 20:52.03 | path | tried with progressiband without success |
| 20:52.14 | path | s/progressiband/progressinband |
| 20:52.43 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-170.cablep.bezeqint.net) |
| 20:52.54 | WIMPy | What about Progrss() or Proceeding()? |
| 20:53.28 | path | what should be defined on my context right? |
| 20:53.32 | nightwalk | Seems like you guys would have saved yourselfs some time by saying something along the lines of, "Do you have a PRI? No? Then there's a <1% chance your PSTN provider will allow you to set caller id" |
| 20:54.33 | nightwalk | /yourselfs/yourselves/ |
| 20:56.09 | [sr] | WIMPy: my PBX S0 lines works as TE ptmp, is this correct? :p |
| 20:57.01 | bent_screwdriver | nightwalk: felt like taking a long walk ;) |
| 20:57.35 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 20:57.37 | WIMPy | [sr]: If you had them internal previousely, that sounds like NT ptmp. |
| 20:58.05 | path | I need to send a 180 SIP is there a manual way to do it? like modifying headers |
| 20:58.24 | [sr] | WIMPy: well, but when i set LCR conf as nt+ptp it doesn't work, only when as TE |
| 20:58.31 | *** part/#asterisk cracoucas (cracoucas@dispo-82-65-228-100.adsl.proxad.net) |
| 20:58.39 | nightwalk | bent_screwdriver: off of a short pier, no doubt :) |
| 20:58.46 | [TK]D-Fender | path: No |
| 20:58.51 | WIMPy | [sr]: Sounds right then. |
| 20:59.15 | path | [TK]D-Fender: the thing is I need to do it since it doesnt ring in the end |
| 20:59.25 | WIMPy | [sr]: You always vonnect TE to NT. |
| 20:59.26 | path | tried with progressinband without sucess |
| 20:59.43 | path | s/sucess/success |
| 21:00.13 | [sr] | WIMPy: i got the point! cards ports as TE, S0 on NT, with crossover cable |
| 21:00.41 | WIMPy | [sr]: No. TE to NT with a straight cable. |
| 21:01.14 | wcselby | i need some operator panel alternatives other than fop / fop2 |
| 21:01.19 | wcselby | any suggestions? |
| 21:01.21 | [sr] | wierd then... its working with the crossover |
| 21:01.33 | [TK]D-Fender | wcselby: astassistant |
| 21:01.56 | WIMPy | [sr]: Then it's a TE port configured to NT mode. |
| 21:02.22 | WIMPy | They won't change pin-out because of configuration. |
| 21:02.32 | [sr] | WIMPy: understood |
| 21:03.28 | [sr] | i still have one hope about the call forward... the manual says it works on the MSN extension 11 (console) i'll give it a try tomorow to see if the CID is forward to the end.. |
| 21:03.50 | [sr] | on the internal digital phones it works |
| 21:04.01 | WIMPy | It's just like using an ISDN card in your PC. In normal operation it's TE and connected with a straight cabel to an NT. Wehen you configure it to NT mode and want to connect a telephone (TE), you |
| 21:04.22 | WIMPy | need a Crossover cable, as the card is still wired as TE, even wen working as NT. |
| 21:05.33 | [sr] | understood |
| 21:05.36 | Gugge | [TK]D-Fender: http://paste2.org/p/920554 <- i set the language before calling Queue(), and in my Local part, but the announcements are still using the default 'en' language. It seems like i can only set the language on the Local.....;2 channel, but its using the language from the Local....;1 channel to play the announcement |
| 21:07.00 | [TK]D-Fender | Gugge: SIP/max_401 <-- what language does this device have? |
| 21:07.23 | Gugge | dk |
| 21:07.42 | Gugge | sorry: da |
| 21:07.51 | Gugge | da is default in sip.conf too |
| 21:08.13 | [TK]D-Fender | Gugge: Set it directly in the peer |
| 21:08.35 | Gugge | it is, both on the peer and as default in sip.conf |
| 21:09.07 | Gugge | If i use Local without the /n part the announcement is danish, and if i use SIP/max_401 directly it is danish too |
| 21:11.03 | Gugge | As far as i can see a similar thing was fixed in followme here https://issues.asterisk.org/view.php?id=15372 |
| 21:11.25 | [sr] | WIMPy: is my PBX doesn't forward the CID anyway to the end on the S0 line, i'm going to make a hybrid PBX... connect the external lines to LCR, catch the number i want and forward it to a SIP extension, and the rest make them go out to the PBX again so that i won't have to replace phones now |
| 21:11.47 | [sr] | WIMPy: in theory it may work, in pratice what do you think? |
| 21:11.52 | *** join/#asterisk residentgrey (~residentg@184-200-71-213.pools.spcsdns.net) |
| 21:12.15 | WIMPy | Works very well. |
| 21:12.22 | [sr] | nice... |
| 21:12.39 | WIMPy | I guess that's a VERY common setup. |
| 21:12.48 | [sr] | it is? nice |
| 21:13.05 | [sr] | i have to see is i can configure it by myself or i'll ask for help |
| 21:14.01 | Gugge | [TK]D-Fender: do you think i should file a "bug", or am i missing a way to set the language variable from the dialplan? |
| 21:14.07 | [sr] | is i = if i |
| 21:16.20 | [sr] | WIMPy: this would be something like: dialing=234377555 : remote=asterisk context=mycontext |
| 21:17.07 | [sr] | i have to see now how to make everything else go to the ports i'll specify to go the PBX |
| 21:18.05 | WIMPy | If you pass everything throug Asterisk you could also use it's voicemail. |
| 21:18.21 | [sr] | hum |
| 21:18.27 | [sr] | that a good ideia |
| 21:18.57 | [sr] | pass everything to asterisk, and then when the exten is not set make it go back to LCR, right? |
| 21:19.34 | WIMPy | yes |
| 21:20.07 | [sr] | have to see the confs |
| 21:20.28 | [sr] | the bad part is that i only can test this at saturday/sunday |
| 21:20.58 | [TK]D-Fender | Gugge: You should be loking at the SIP channel languages as well |
| 21:22.56 | Gugge | [TK]D-Fender: i will |
| 21:26.31 | *** join/#asterisk kurkowski (~felipe@189.62.199.201) |
| 21:27.55 | kurkowski | Anyone know how to proper use the G option of the Dial application? |
| 21:28.27 | kurkowski | I'm trying this, but with no success: exten => 1,n,Dial(SIP/${numero}@99051000xxxxxx,20,G(agiscript,s,1)) |
| 21:28.46 | kurkowski | I tried with the ^ aswell |
| 21:30.11 | [TK]D-Fender | kurkowski: Show us the failed call and your dialplan and please specify your * version. |
| 21:31.17 | [TK]D-Fender | kurkowski: and it is "^", not "," |
| 21:31.23 | kurkowski | Should I upload the dialplan? Or do I just paste the useful stuff here? |
| 21:31.29 | kurkowski | Yeah, I tried with the ^ first |
| 21:31.30 | [TK]D-Fender | ~pb |
| 21:31.31 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood 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://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 21:31.32 | [TK]D-Fender | ^^^^^^^^^^^^^ |
| 21:31.37 | kurkowski | okay |
| 21:31.39 | kurkowski | one second |
| 21:32.41 | Gugge | [TK]D-Fender: all SIP channels have da as language too, ill try if i can hack something together, and post an issue |
| 21:35.39 | [sr] | hum i have something wrong with voicemail |
| 21:35.50 | [sr] | i have it enabled for all incoming/outgoing call's |
| 21:35.56 | [sr] | but nothing is recorded |
| 21:36.08 | *** join/#asterisk Ryushin (proxy@windwalker.openinnovations.com) |
| 21:36.14 | [TK]D-Fender | [sr]: what does "enabled" mean? |
| 21:36.18 | kurkowski | http://pastebin.com/e19gKmZa |
| 21:36.20 | kurkowski | here it is |
| 21:36.29 | kurkowski | I explained what I'm trying to do in there too |
| 21:37.05 | [TK]D-Fender | kurkowski: show me the call |
| 21:37.06 | [sr] | [TK]D-Fender: voicemail & directory = enabled (SIP extensions created with freepbx) |
| 21:37.24 | Qwell | [sr]: #freepbx |
| 21:37.24 | [TK]D-Fender | [sr]: FreePBX is NOT supported in here. |
| 21:37.25 | kurkowski | you mean the debug and verbose at the asterisk console? |
| 21:37.31 | [TK]D-Fender | kurkowski: Yes |
| 21:38.42 | [sr] | i was expecting that answer :P |
| 21:38.58 | [sr] | ok voicemail is working, what is not working is call recording |
| 21:40.13 | [TK]D-Fender | [sr]: Your description is useless and you are showing nothing and then changing the topic. |
| 21:40.24 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 21:40.34 | kurkowski | http://pastebin.com/vYbXNgGB |
| 21:40.40 | kurkowski | I appended the dial |
| 21:40.57 | kurkowski | if this word even exists lol |
| 21:42.34 | kurkowski | also, I've tried similar stuff using Macros (M) and GoSub (U) but it never reach the macro/routine context |
| 21:43.01 | [TK]D-Fender | 20,G(agiscript,s,1)" |
| 21:43.07 | [TK]D-Fender | kurkowski: I see , not "^" |
| 21:43.27 | [TK]D-Fender | kurkowski: I also don't see why there are 2 SIP calls tfrom the sme peer in there |
| 21:43.43 | kurkowski | oh, I didnt change last time to get the dial for you |
| 21:43.44 | [TK]D-Fender | -- Executing [99051000107744@in-skype:1] Answer("SIP/99051000107744-0000000d", "") in new stack |
| 21:43.49 | kurkowski | but I tried with the ^as well |
| 21:43.51 | [TK]D-Fender | <PROTECTED> |
| 21:44.01 | kurkowski | See, this is something I dont understand and I also asked on the forums |
| 21:44.03 | [TK]D-Fender | 1 digit difference.. now why would that be? |
| 21:44.10 | kurkowski | when I first originate the call |
| 21:44.14 | [sr] | [TK]D-Fender: i'm checking the files 1st... let me dig into them |
| 21:44.16 | [TK]D-Fender | kurkowski: You are showing me something I told you was WRONG before. |
| 21:44.20 | kurkowski | it will first go to the INCOMING context |
| 21:44.42 | kurkowski | okay, I'll correct it |
| 21:44.45 | kurkowski | thanks |
| 21:48.49 | kurkowski | meh, same thing even with the ^ |
| 21:48.53 | [TK]D-Fender | Out of time, back substantially later. |
| 21:48.56 | [TK]D-Fender | heads out. |
| 21:50.50 | kurkowski | the problem seems to be that the Originate will first go to the incoming context and then to the one I coded. This doesnt happen when I use another SIP channel. |
| 21:55.27 | *** part/#asterisk crowbar7 (~Crowbar7@12.2.84.2) |
| 22:01.16 | *** join/#asterisk bjhaid (~IceChat7@41.220.68.1) |
| 22:02.14 | *** part/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 22:07.52 | *** join/#asterisk bjhaid (~IceChat7@41.220.68.7) |
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| 22:14.40 | *** part/#asterisk kurkowski (~felipe@189.62.199.201) |
| 22:16.17 | [sr] | going to sleep |
| 22:16.20 | [sr] | see u all |
| 22:17.39 | carrar | WAIT |
| 22:17.42 | carrar | Don't go!! |
| 22:20.17 | *** join/#asterisk Crowbar7 (~Crowbar@c-24-11-109-101.hsd1.mi.comcast.net) |
| 22:21.47 | Crowbar7 | where is there a good async agi tutorial? |
| 22:25.19 | carrar | http://tinyurl.com/2bja72f |
| 22:26.36 | *** join/#asterisk cesar_CR (~cesar@201.201.41.242) |
| 22:31.26 | *** join/#asterisk boodu (~antoine@host-115-126-167-240.adsl.nautile.nc) |
| 22:39.00 | boodu | hello |
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| 22:47.29 | *** join/#asterisk voxter (~voxter@macpro.daytonhome.voxter.net) |
| 22:48.34 | voxter | so, im getting a lot of 'received mini frame before first full voice frame' on an iax2 call.. what im seeing happen, strangely enough, is that packets are still flowing back and forth between the IAX peers (local pbx, remote pbx) but audio inbound ceases while this message flies by at multiple times per second.. considering packets are still flowing, anyone have any thoughts on what could be happening? |
| 23:06.09 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
| 23:06.15 | *** join/#asterisk lost_soul (shackett@devio.us) |
| 23:26.14 | *** join/#asterisk SuperBock (~admin@mx-cln-1.netcanvas.com) |
| 23:26.21 | SuperBock | hellllo |
| 23:26.37 | SuperBock | Having some PRI problems, may someone can help? |
| 23:28.03 | *** join/#asterisk Micc (~quassel@c-98-225-57-96.hsd1.wa.comcast.net) |
| 23:28.27 | SuperBock | Our telco recently changed PRI switch, and now we see a *constant* flood of HDLC Abort errors, however this seems to have no impact on calls. Still, it's extremely annoying, because it's like between 5 to 20 HDLC Aborts every second showing on the console. Any hint on what could be the problem? They say the line is clear on their test equipment. |
| 23:28.53 | Micc | I've configured an audiocodes that I had working fine before, but now when I try to dial I just get busy tone, status page shows Reorder... But incoming calls work fine and its registered with asterisk, but I don't see it even trying to dial. |
| 23:30.37 | Chainsaw | SuperBock: HDLC aborts can also happen because the server can not handle the interrupts quickly enough. |
| 23:31.00 | Chainsaw | SuperBock: For example, because the interrupt line is shared with other hardware. This is to be avoided. Play PCI roulette unless you see the card on a separate IRQ in /proc/interrupts |
| 23:31.41 | Micc | the dialing page show digitmap is blank and dial plan is blank, shouldn't there be something there? |
| 23:32.10 | SuperBock | I understand that. But this only happened after the PRI equipment changed, this box has been running happily for a year before. Also, we tried another box, different hardware with a single span card, as soon as I connect the PRI cable to it, the flood on the console starts. |
| 23:33.00 | SuperBock | also, irq problems tend to show with FCS errors too, and not with a constant flood (from what I've seen) |
| 23:33.12 | SuperBock | there's not a single FCS error here |
| 23:33.58 | SuperBock | something changed, but what/how and how to configure for it, that is the problem.. |
| 23:34.35 | Chainsaw | SuperBock: Do check whether the interrupt line is shared. It is standard troubleshooting. |
| 23:35.34 | SuperBock | no sharing |
| 23:35.36 | SuperBock | <PROTECTED> |
| 23:35.50 | SuperBock | happily alone |
| 23:37.52 | Chainsaw | SuperBock: Indeed, that looks beautiful. |
| 23:38.05 | kerx | hi all, where can i find the latest in the 1.4.x branch? |
| 23:40.00 | SuperBock | yes, that is beautiful. the console is rather nasty though |
| 23:40.17 | SuperBock | I've never seen this before |
| 23:41.26 | Chainsaw | SuperBock: Have you tried with a loopback cable? |
| 23:41.36 | Chainsaw | SuperBock: This is likely the second thing that Digium support will ask. |
| 23:42.29 | Chainsaw | SuperBock: (Run a continuous bit test with the loopback cable in for at least 10 minutes; if you do not see HDLC errors then, your card & mainboard are okay) |
| 23:44.04 | kerx | is something up with asterisk? |
| 23:44.04 | kerx | http://www.asterisk.org/downloads/asterisk/releases/asterisk-1.4.33.1.tar.gz |
| 23:44.08 | kerx | I'm trying to download that file |
| 23:44.12 | kerx | And I'm getting 403 forbidden |
| 23:44.15 | SuperBock | Actually we did. Also replaced the coax to RJ45 converter that comes with the cards, and tried a different coax cable pair. Two different boards (known to work in many systems) were used, one with a TE2xxp and another with a TE1xxp, exact same result. |
| 23:44.58 | Chainsaw | SuperBock: Agreed, then you've ruled out your end. |
| 23:45.09 | WIMPy | Coax? Coool. |
| 23:45.15 | Chainsaw | SuperBock: For a BT BRI/PRI service I would ask for a lift & shift. |
| 23:45.16 | SuperBock | also upgraded to ast 1.4.30 and dahdi 2.2.1.2 |
| 23:45.33 | carrar | kerx, yeah their link is bad |
| 23:45.40 | kerx | darn. |
| 23:45.49 | Chainsaw | SuperBock: You will have to find what the local telco jargon for "I want to be connected to a *different* line card!" is. |
| 23:45.51 | kerx | wonder why it's not working |
| 23:46.07 | carrar | human blunder |
| 23:46.49 | carrar | kerx, try this one: http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.4.33.1.tar.gz |
| 23:47.05 | SuperBock | Been battling with them... it's the most difficult part of the process.. |
| 23:47.52 | SuperBock | one doubt I have.. in order to change timing options for a span, a module reload or even reboot is necessary? or a dahdi_cfg would suffice? |
| 23:50.22 | Chainsaw | SuperBock: I would reload the module at the very least. |
| 23:51.31 | kerx | carrar, thanks. i wonder what's up with there webpage there |
| 23:51.42 | carrar | life happens |
| 23:51.50 | kerx | when it rains, it pours |
| 23:51.56 | kerx | i hope they got there ubmrella :) |