| 00:00.13 | anthm | are all those other contexts present ? |
| 00:00.27 | anthm | did you try just "show dialplan entrada" |
| 00:00.33 | Paavum | no... |
| 00:00.39 | Paavum | gonnado show dialplanentrada |
| 00:00.54 | Paavum | paavum*CLI> show dialplan entrada |
| 00:00.54 | Paavum | There is no existence of 'entrada' context |
| 00:01.06 | anthm | how many contexts do you see |
| 00:01.16 | anthm | when you do a normal show dialplan |
| 00:01.26 | Paavum | 19 |
| 00:01.43 | anthm | what is the last one |
| 00:02.01 | Paavum | ext-rf |
| 00:03.13 | anthm | pastebin the output of show dialplans too |
| 00:03.20 | anthm | you must have a stray or missing ( somewhere |
| 00:03.42 | Paavum | ok |
| 00:04.36 | *** join/#asterisk Kerry_G (n=Kerry_G@ip70-187-129-227.oc.oc.cox.net) |
| 00:05.07 | anthm | try commenting out the line 126 include=>extensiones |
| 00:05.15 | anthm | as well |
| 00:05.23 | anthm | there is no context called that |
| 00:06.15 | *** join/#asterisk |dennis| (n=dennis@200.32.215.82) |
| 00:06.21 | Paavum | ok just did that |
| 00:11.27 | Ciber311 | http://pastebin.ca/117529 anyone have any idea what i screwed up? |
| 00:12.29 | anthm | allow=all in the general of sip.conf or in the peer |
| 00:14.13 | Ciber311 | to add it or remove it? :P |
| 00:15.43 | anthm | add |
| 00:18.41 | *** join/#asterisk ESCulapio__ (n=ESCulapi@183stb68.codetel.net.do) |
| 00:27.18 | Paavum | anthm... |
| 00:27.26 | Paavum | you there? |
| 00:30.37 | Paavum | I just moved the entradascontext up |
| 00:30.51 | Paavum | but Ikeep getting the same context |
| 00:31.17 | Paavum | ack |
| 00:31.21 | Paavum | the sameproblem |
| 00:33.05 | ManxPower | *sigh* The BOFH is my hero. *sigh* |
| 00:33.39 | Dorphalsig | 0 |
| 00:38.26 | *** join/#asterisk tempest1 (n=asf@adsl-144-57-36.chs.bellsouth.net) |
| 00:38.46 | *** join/#asterisk test34 (n=test34@unaffiliated/test34) |
| 00:41.45 | Paavum | Hello? I cant recieve any calls on my * box... I get this message on thecli: Extension '5922999' in context 'ENTRANTES' from '15922999' does not exist. Rejecting call on channel 0/31, span 1 |
| 00:42.14 | Paavum | I do show dialplan entrantes and I can see it! |
| 00:42.42 | Paavum | '_592299[79]' => -2. goto(IVR-CAYRE|s|1) [pbx_config] |
| 00:44.24 | Paavum | Anybody?? |
| 00:44.30 | *** join/#asterisk trbldwine (i=troubled@71.194.161.170) |
| 00:44.38 | *** join/#asterisk jart (n=user@ool-4356f82f.dyn.optonline.net) |
| 00:44.42 | Paavum | hello? |
| 00:44.44 | jart | hello there |
| 00:44.52 | Paavum | hey jart |
| 00:45.02 | jart | just got back from clue con, reppin' * |
| 00:45.28 | Paavum | and how wasit? |
| 00:45.47 | jart | it was fun, met a lot of cool people |
| 00:46.03 | jart | got some companies interested in my consulting services |
| 00:47.01 | Paavum | I think I am interested in your consulting services |
| 00:47.07 | Paavum | :P |
| 00:47.22 | Paavum | I have the funkiest issue withmy* box |
| 00:47.46 | Paavum | It wont take anycalls |
| 00:48.08 | Paavum | its been 4 hours since I started trying to see what was up |
| 00:48.22 | Paavum | and I havent been able to dig the thing |
| 00:48.26 | anthm | your syntax is wrong |
| 00:48.27 | Paavum | can ya givemea hand¿ |
| 00:48.31 | Qwell | Paavum: You've never said what errors you're getting, did you? |
| 00:48.35 | anthm | you just fixed your dialplan |
| 00:48.43 | anthm | now you need to change the exten |
| 00:49.16 | Kerry_G | no, you have spent 4 hours begging people to fix the problems you made |
| 00:49.39 | Paavum | Qwell --> Extension '5922999' in context 'ENTRANTES' from '15922999' does not exist. Rejecting call on channel 0/5, span 1 |
| 00:50.09 | anthm | you need an extension that will match in that context the one you have wont |
| 00:50.10 | Qwell | and does the extension 5922999 exist in the context ENTRANTES? |
| 00:50.16 | Paavum | yeah |
| 00:50.34 | Qwell | pastebin your dialplan |
| 00:50.40 | anthm | no you are making assumptions about the syntax of the exten |
| 00:51.00 | ManxPower | I'll bet the extensions exists in the contest "entrantes", but not in context "ENTRANTES" |
| 00:51.12 | Qwell | ManxPower: s'what I'm betting on too, actually |
| 00:52.08 | Paavum | exten => _592299[79],s,goto(IVR-CAYRE,s,1) |
| 00:52.33 | Qwell | Paavum: in what context? |
| 00:52.39 | Paavum | [ Context 'ENTRANTES' created by 'pbx_config' ] |
| 00:52.45 | Paavum | <PROTECTED> |
| 00:53.01 | Qwell | why are you using priority s? |
| 00:53.03 | anthm | do you not see you are at a new level in your problem before you didnt even see that exten in your dialplan ? |
| 00:53.28 | Paavum | anthm... yes I do realize that,I said so before |
| 00:53.37 | Qwell | Paavum: change the s to a 1 |
| 00:53.41 | anthm | yes make ,s, ,1, |
| 00:54.28 | Paavum | Qwell.. that did it thnx |
| 00:55.26 | jart | if you can give me any info real quick |
| 00:55.26 | jart | in this chat, i can help you for free now |
| 00:55.26 | jart | if you need me on site: http://www.lobstertech.com/solutions.php |
| 00:57.20 | anthm | he left =D |
| 00:57.41 | Qwell | Do you seriously expect that picture to attract clients? |
| 01:03.19 | Dorphalsig | yes he use to do that! |
| 01:03.40 | *** join/#asterisk tempest1 (n=asf@c-68-58-178-46.hsd1.sc.comcast.net) |
| 01:07.45 | *** join/#asterisk Trakkasure (n=Sgemtum@69-163-145-203.atlsfl.adelphia.net) |
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| 01:12.46 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 01:12.55 | Dorphalsig | ++ |
| 01:16.10 | *** part/#asterisk trbldwine (i=troubled@71.194.161.170) |
| 01:18.21 | *** join/#asterisk SwK_ (n=Silik0nJ@12-218-74-89.client.mchsi.com) |
| 01:27.30 | *** join/#asterisk brut- (n=brut@66.173.4.254) |
| 01:27.44 | brut- | lo everyone |
| 01:29.39 | Dorphalsig | qwell . . .when you said " Do you seriously expect that picture to attract clients?" . . . what about were you talking? |
| 01:30.10 | hads|home | http://www.lobstertech.com/solutions.php |
| 01:30.42 | brut- | can anyone tell me if asterisk's call accounting system supports hobic? |
| 01:35.45 | *** join/#asterisk ESCulapio__ (n=ESCulapi@231stb68.codetel.net.do) |
| 01:35.50 | ESCulapio__ | j #codigolibre |
| 01:40.03 | *** part/#asterisk n838901_2 (n=name@24-117-16-198.cpe.cableone.net) |
| 01:43.08 | Dorphalsig | How can I assign some extensions to its specific PBX (some PBXs in the same *Box) without sharing every extension . . . every PBX only using the extensions that belongs! |
| 01:45.35 | ManxPower | Dorphalsig, We have NO idea what you just said. |
| 01:48.27 | niter3 | hey guys, what's a good softphone for linux? |
| 01:57.40 | ManxPower | All softphones suck!(tm) |
| 01:57.50 | niter3 | what the fuck is this |
| 01:58.00 | niter3 | -- x=0, open writing: /var/spool/asterisk/voicemail/default/777/unavail format: wav49, 0x86ada00 -- x=1, open writing: /var/spool/asterisk/voicemail/default/777/unavail format: gsm, 0x86adc00 -- x=2, open writing: /var/spool/asterisk/voicemail/default/777/unavail format: wav, 0x86ac000 |
| 02:02.41 | jeebusroxors | anyone remember if the old phones used 6 or 12v? |
| 02:04.01 | Qwell | 42vdc |
| 02:04.14 | Supaplex | 42 sounds right. |
| 02:04.16 | jeebusroxors | 42v :( thats alot |
| 02:04.23 | Qwell | What old phones? |
| 02:04.49 | *** join/#asterisk }btorch{ (n=btorch@c-66-176-87-59.hsd1.fl.comcast.net) |
| 02:04.51 | jeebusroxors | i just hacked one of those 4 wire phones to work with my ata and a 13.8v power sup |
| 02:05.19 | jeebusroxors | but it wasnt ringing....so im guessing that is because its missing ~30v |
| 02:05.29 | Supaplex | then we beg to ask how old is old for you :) |
| 02:05.44 | jeebusroxors | heh |
| 02:05.54 | }btorch{ | hey guys is it really necessary to remove the old libpri, asterisk and zaptel files before compiling the new version? |
| 02:06.04 | Qwell | }btorch{: yes |
| 02:06.22 | jeebusroxors | old was a relitive term... |
| 02:07.05 | }btorch{ | i can't just install over it ? |
| 02:07.14 | Qwell | You *CAN*, sure |
| 02:07.17 | Qwell | but it's stupid |
| 02:07.19 | jeebusroxors | fack where am i supposed to find 42v :( thats alot of car batteries |
| 02:07.24 | Qwell | and it won't...work |
| 02:07.28 | Qwell | jeebusmobile: i was joking :p |
| 02:07.29 | }btorch{ | it's not like they are installed under /usr or /usr/local |
| 02:07.35 | Qwell | ring voltage is like 48vdc though, iirc |
| 02:07.40 | Qwell | on new phones |
| 02:07.44 | jeebusroxors | ya |
| 02:07.55 | jeebusroxors | not fun to get hit with heh |
| 02:08.02 | Qwell | So, any phone should kinda just work |
| 02:08.11 | Qwell | You wired it wrong ;) |
| 02:08.34 | jeebusroxors | no, older phones had 4 wires per line 2 for voltage and 2 for the signal |
| 02:09.12 | *** join/#asterisk |dennis| (n=dennis@200.32.215.82) |
| 02:13.17 | niter3 | anyone know why i'm having troube writing? |
| 02:14.47 | hads|home | No pen? |
| 02:15.03 | Qwell | no paper is my guess |
| 02:15.13 | niter3 | -- x=0, open writing: /var/spool/asterisk/voicemail/default/777/unavail format: wav49, 0x86ada00 -- x=1, open writing: /var/spool/asterisk/voicemail/default/777/unavail format: gsm, 0x86adc00 -- x=2, open writing: /var/spool/asterisk/voicemail/default/777/unavail format: wav, 0x86ac000 |
| 02:15.21 | niter3 | getting this crap |
| 02:15.27 | Qwell | niter3: and? |
| 02:15.29 | niter3 | when trying to record a message when i'm gone |
| 02:15.45 | Qwell | That is completely normal |
| 02:18.18 | }btorch{ | hey so i got remove all zaptel modules and files manually before making and installing the new zaptel ? |
| 02:18.36 | Qwell | }btorch{: no, all of a sudden the answer suddenly changed |
| 02:18.54 | }btorch{ | :-) |
| 02:18.59 | Qwell | between now and 10 minutes ago when you asked the same exact question |
| 02:19.14 | }btorch{ | no before was refering to asterisk |
| 02:19.30 | Qwell | You specifically mentioned libpri, zaptel, AND asterisk |
| 02:19.37 | }btorch{ | opps sorry |
| 02:19.43 | }btorch{ | my bad u r right |
| 02:20.39 | }btorch{ | its just that the makefile for zaptel has rm statements in case it finds the old file |
| 02:20.40 | niter3 | Qwell: really? |
| 02:21.07 | }btorch{ | acccording to the makefile it removes the old ones before installing the new ones |
| 02:25.49 | *** part/#asterisk Kerry_G (n=Kerry_G@ip70-187-129-227.oc.oc.cox.net) |
| 02:25.53 | Qwell | }btorch{: You're reading it wrong |
| 02:26.30 | Qwell | }btorch{: just try it. if it doesn't work, I'll save an "I told you so" for you |
| 02:30.09 | }btorch{ | ok ... one question though for anyone who has experience with zaptel ... |
| 02:30.39 | *** join/#asterisk psymon101 (i=psymon10@24-236-222-55.dhcp.aldl.mi.charter.com) |
| 02:31.05 | }btorch{ | i currently have this asterisk (which I'm upgrading now) and it uses a digium card to talk to the PSTN |
| 02:31.11 | linlin | ok, I am prepred to pay someone (not all that much) to remote into my box and try to diagnose the issue with my IAX connections. Going on 5 days of messing with this and its wracking my head. I need to get this working. Any takers? |
| 02:32.01 | }btorch{ | for some reason dtmf tones coming from some phone lines are not detected correctly by asterisk |
| 02:32.59 | }btorch{ | for example from my crappy vonage line I could never enter an extension number while in a menu i created on my dialplan |
| 02:33.23 | Rowter | set_actual_txgain: Failed to read gains: Invalid argument anyone has seen this? |
| 02:33.24 | }btorch{ | i had to try it like several times .. from my cell phone was no problem |
| 02:33.51 | }btorch{ | if i use my smart phone though i have the same issue |
| 02:35.54 | psymon101 | Btorch, I am real new to Asterisk so dont take this ad bible, your DTMF is set to the RFC ? |
| 02:36.26 | }btorch{ | what you mean by my DTMF set to RFC ? |
| 02:36.42 | }btorch{ | in some zaptel or asterisk config file you mean ? |
| 02:37.14 | psymon101 | actually in sip.conf |
| 02:37.47 | psymon101 | dtmfmode = rfc28 |
| 02:38.00 | psymon101 | again, I am new so its just a hunch |
| 02:38.01 | psymon101 | :) |
| 02:38.48 | }btorch{ | ok i use iax2 but the issue is not when a call is originated from a iax client is when a call comes from an outside caller like a cell phone , or a lan line and ends up on a menu i created on my dialplan |
| 02:40.55 | *** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk) |
| 02:41.06 | psymon101 | you can put it in your iax.conf also |
| 02:41.57 | psymon101 | thats what a quick google showed |
| 02:42.19 | psymon101 | People said it worked, nothing official from Asterisk though. |
| 02:44.55 | psymon101 | also seeing people saying codecs mungling tones |
| 02:45.16 | psymon101 | brb |
| 02:48.00 | brut- | does anyone know if asterisk's cdr application supports hobic? I can't find any reference to it in the docs, but I'm hoping :S |
| 02:57.14 | Qwell | okay, wtf is Strom |
| 02:57.44 | }btorch{ | Qwell, have you compiled the latest zaptel ? |
| 02:58.19 | }btorch{ | i get a bunch of errors on some vpm450m.c file during make |
| 03:00.26 | *** join/#asterisk s0lid (n=jlq@210.213.198.167) |
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| 03:03.01 | *** join/#asterisk steven_Laptop (n=steven@c-24-118-44-83.hsd1.mn.comcast.net) |
| 03:03.19 | steven_Laptop | hello all |
| 03:04.46 | steven_Laptop | ? I own a small Co. and want to try asterisk out but need to get some hardware ? aswered first.. |
| 03:06.27 | psymon101 | If you download "Asterisk The future of Telephony" it outlines hardware specs |
| 03:06.41 | steven_Laptop | 1 I have one phone line coming into my home/office witch is Phone/Fax for now but I see 4 lines coming soon What is the best card for me? |
| 03:07.15 | steven_Laptop | IC |
| 03:08.06 | steven_Laptop | from here? http://www.asterisk.org/ |
| 03:09.38 | psymon101 | here ya go http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 03:09.50 | russellb | the TDM400P supports up to 4 phones/phone lines ... or the TDM2400P supports up to 24 phones/phone lines |
| 03:10.20 | psymon101 | Russell beat me to the model :) |
| 03:10.48 | steven_Laptop | I think 4 Will be good for me for the next 8 yrs :) |
| 03:11.13 | russellb | specifically, for 4 phone lines, you will want the TDM04B (i think) |
| 03:11.37 | steven_Laptop | cool thanks |
| 03:11.52 | psymon101 | http://www.telephonyware.com/telephonyware/tw00031.html |
| 03:12.49 | steven_Laptop | cool im ordering now! |
| 03:12.56 | psymon101 | shop around:) |
| 03:13.02 | steven_Laptop | yes |
| 03:13.05 | steven_Laptop | :) |
| 03:13.19 | steven_Laptop | anythink else I may need? |
| 03:13.36 | steven_Laptop | or be aware of |
| 03:13.41 | russellb | depends on what you want to do, of course |
| 03:13.46 | wunderkin | that is assuming you will be using all ip phones |
| 03:14.14 | russellb | if you want to use analog phones, the TDM2400P would probably be a better choice |
| 03:15.55 | steven_Laptop | well hehe I will need to find out what I have here IP/analog for starters |
| 03:16.32 | russellb | yep, that would be helpful :) |
| 03:16.38 | psymon101 | heh |
| 03:17.52 | *** join/#asterisk bugz (n=bugz@cpe-70-123-122-41.houston.res.rr.com) |
| 03:20.18 | steven_Laptop | well one last ? here I run a ubuntu sever here /file/samba/web/mail and want to add asterisk to it, any problems with doing this? |
| 03:21.25 | russellb | nope |
| 03:21.56 | russellb | if it's going to be a pretty small install, anyway ... |
| 03:22.13 | russellb | most people run it on a dedicated box, though |
| 03:22.39 | russellb | since unlike those other server applications, asterisk is a real-time system |
| 03:23.18 | steven_Laptop | im short on boxes :) and its a P4 3.0 with 4G of ram and 600G harddrive |
| 03:23.27 | steven_Laptop | ic |
| 03:24.02 | steven_Laptop | its going to be used as a Voice mail/fax server |
| 03:24.06 | }btorch{ | i run asterisk on the same P4 i use for my wildfire server ... it works ok |
| 03:25.33 | steven_Laptop | }btorch{, thanks |
| 03:25.39 | *** join/#asterisk technoid_ (n=technoid@host-69-95-125-54.cwon.choiceone.net) |
| 03:29.03 | *** join/#asterisk jart (n=user@ool-4356f82f.dyn.optonline.net) |
| 03:29.32 | steven_Laptop | I must have ip being its a reg phone line :) |
| 03:30.43 | Un1x | i have a dedicated box for mine :) |
| 03:30.50 | Un1x | better that way ASterisk will get all of the CPU :) |
| 03:31.38 | jart | anyone going to the vampyre club 2night? |
| 03:31.55 | steven_Laptop | Un1x, will a p2 400mhz work with 128ram? |
| 03:32.14 | Un1x | really depends on how much lines youre going to use, and what kinda of codec etc.. |
| 03:32.36 | Un1x | i mean you could, but if you wanna use a codec other then Ulaw, then u cant but if youre ok with Ulaw then go ahead ;) |
| 03:32.41 | Un1x | Ulaw = 70Kb |
| 03:32.47 | Un1x | so better have about 90Kb upload |
| 03:32.56 | Un1x | gotta count Overhead, and system usage :) |
| 03:33.40 | jart | ulaw has no cpu overhead |
| 03:33.59 | steven_Laptop | yaaa well it will go on my reg sever then thats a p4 3.0 with 4g of ram |
| 03:34.05 | jart | the high memory asterisk implementation simply does table lookups to transcode between ulaw and slin |
| 03:34.35 | steven_Laptop | Im not even using 10% of the system under load now |
| 03:35.02 | russellb | high memort asterisk implementation? |
| 03:35.19 | jart | yea, as in not turning on that low memory define |
| 03:35.20 | russellb | memory* |
| 03:35.23 | russellb | ok. |
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| 03:42.07 | *** join/#asterisk Ciber311 (n=Ciber311@user-1087e94.cable.mindspring.com) |
| 03:42.18 | Ciber311 | hey guys |
| 03:42.40 | Ciber311 | having a weird issue, getting no incoming sound on one of my sip DID's |
| 03:42.48 | Ciber311 | yet i get sound on another |
| 03:42.51 | Ciber311 | different providers |
| 03:42.59 | Ciber311 | both have nat=yes |
| 03:51.54 | *** join/#asterisk LakeSolon (n=blake@12-227-169-99.client.mchsi.com) |
| 03:53.16 | Un1x | nat=no :p |
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| 04:32.54 | steve_Laptop | ? is this the same as asterisk - http://www.intuitivevoice.com/evolutionps.htm |
| 04:35.42 | Ciber311 | "Evolution has a powerful arscenol of features ready to be unleashed under its hood." |
| 04:35.51 | Ciber311 | arscenol? lol |
| 04:35.52 | coppice | well, they use digium cards |
| 04:36.17 | coppice | Ciber311: its a derivative of arsenic |
| 04:36.40 | Ciber311 | for a minute there i thought they really fucked up arsenal :P |
| 04:37.52 | steve_Laptop | so is it good and a quick way to get your phone system up? WTF $500.00 for it to |
| 04:38.21 | Ciber311 | coppice: you sure about that? i can't find that word ANYWHERE |
| 04:38.31 | Ciber311 | not in my dictionaries and 0 results on google |
| 04:39.03 | coppice | gee, is a :-) needed for every joke on the internet? |
| 04:39.14 | Ciber311 | oh lol |
| 04:40.18 | coppice | if it was a derivative of arsenic, it would be a serious poison, and I guess unleashing that would relate it to arsenal |
| 04:40.59 | steve_Laptop | im using faxtalk now and what a pc of shit.... |
| 04:41.36 | steve_Laptop | if you care http://www.faxtalk.com/products/ftmp7/index.htm?source=Google&product=FTMP&gclid=CNWzga2fyoYCFUAjJAod5SJZ7Q |
| 04:41.38 | steve_Laptop | :) |
| 05:15.28 | niter3 | hey guys where is teh default directory that mp3s are to be stored into? |
| 05:17.13 | docelmo | ... /var/spool/mail |
| 05:17.51 | }btorch{ | niter3, for moh ? |
| 05:18.54 | niter3 | what |
| 05:18.59 | niter3 | where asterisk looks for the mp3s |
| 05:19.06 | niter3 | i want to play music when my income trunk picks up |
| 05:20.12 | }btorch{ | well i know that moh is at /var/lib/asterisk/mohmp3 |
| 05:20.19 | }btorch{ | but you can change that |
| 05:20.40 | niter3 | hrm.... |
| 05:20.45 | niter3 | i stored my mp3 inj there |
| 05:20.57 | niter3 | and it trys to play the mp3 but i get a timed out errored |
| 05:21.06 | niter3 | i have a feeling it's looking in the wrong place |
| 05:21.11 | niter3 | this isn't for music on hold to let you know |
| 05:22.23 | niter3 | ok that worked |
| 05:22.31 | niter3 | i gave a full path to the mp3 |
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| 05:33.21 | niter3 | hey just wondering, how can I make an overall extension that willll apply to everyone to bring them to the beginning of the phone system? As if somebody was dialing in? |
| 05:33.27 | niter3 | bring them to the default context. |
| 05:43.00 | *** join/#asterisk nailbags|laptop (n=neil@203-206-217-36.perm.iinet.net.au) |
| 05:46.50 | Corydon76-home | niter3: Goto |
| 05:47.37 | *** join/#asterisk constfilin (n=cf@204.147.182.37) |
| 06:00.26 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 06:01.26 | *** join/#asterisk topping (n=topping@207.47.6.192.static.nextweb.net) |
| 06:02.52 | *** join/#asterisk oomph (n=jimmy@ip68-98-166-99.dc.dc.cox.net) |
| 06:03.40 | oomph | what are some good cheap phones to use? |
| 06:03.56 | russellb | walmart analog phones |
| 06:04.01 | oomph | i currently use the polycoms but their webinterface kinda sucks |
| 06:04.12 | oomph | i configure them via ftp |
| 06:04.24 | oomph | but im looking into other alternatives |
| 06:04.30 | Corydon76-home | Walgreen's analog phones are good, too |
| 06:04.50 | russellb | target has some just a small step up |
| 06:04.54 | russellb | you know, for things like callerid |
| 06:05.12 | oomph | Corydon76-home: preferably, SIP phones for business use |
| 06:05.29 | Corydon76-home | but you have to purchase them on sale, less than an hour before you're supposed to be done setting up your booth on the convention floor |
| 06:06.29 | file | dejavu |
| 06:06.40 | Corydon76-home | all over again |
| 06:06.53 | *** join/#asterisk SheriF_WorK (n=sherif@212.103.170.135) |
| 06:08.41 | *** join/#asterisk tempest1 (n=asf@adsl-144-57-36.chs.bellsouth.net) |
| 06:09.13 | *** join/#asterisk svenna (n=svenna@p548D2EB2.dip0.t-ipconnect.de) |
| 06:09.36 | Ciber311 | grrrrr |
| 06:09.41 | Ciber311 | still no incoming sound |
| 06:13.02 | *** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135) |
| 06:13.56 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:15.14 | *** part/#asterisk ManxPower (n=ewieling@71-8-11-111.dhcp.leds.al.charter.com) |
| 06:26.26 | Ciber311 | anyone here using voxbone? |
| 06:30.58 | oomph | nope, whats that? |
| 06:41.48 | Ciber311 | DID provider |
| 06:43.57 | *** join/#asterisk Bobcat_1966 (n=chatzill@cpe-069-132-139-254.carolina.res.rr.com) |
| 06:44.54 | oomph | Ciber311: they got decent rates? |
| 06:45.11 | Ciber311 | think 8 bucks a month for unlimited incoming? |
| 06:45.36 | Ciber311 | depends where the number you want is |
| 06:46.19 | oomph | thats not too bad |
| 06:46.28 | oomph | i used junction networks before |
| 06:46.35 | oomph | they were good |
| 06:46.47 | oomph | per minute plans mostly |
| 06:47.07 | Ciber311 | screw per minute :P |
| 06:49.35 | *** join/#asterisk budmang (n=budman@12.206.134.162) |
| 06:49.36 | budmang | anyone good with queues? When ever I add more then 3 static agents we get all circuts are busy now. |
| 06:49.38 | oomph | yeah i agree with ya there |
| 06:49.41 | oomph | but i used it for testing |
| 06:49.50 | oomph | so a few cents here and there |
| 06:49.51 | oomph | no biggie |
| 06:51.10 | *** join/#asterisk bmg505 (n=leon@c1-226-14.rndf.isadsl.co.za) |
| 06:51.58 | Ciber311 | oomph: know how to setup an iax trunk? |
| 06:53.11 | oomph | i've done it before, but dont have an asterisk install in front of me to refer to |
| 06:53.30 | oomph | how many trunks you trying o configure? |
| 06:53.35 | Ciber311 | 1 |
| 06:53.36 | Ciber311 | hehe |
| 06:53.44 | Ciber311 | others are working as sip just fine |
| 06:53.49 | Ciber311 | except for the voxbone one |
| 06:54.01 | Ciber311 | for some damn reason i keep getting one way audio |
| 06:54.08 | Ciber311 | no incoming sound |
| 06:54.11 | oomph | ah ya |
| 06:54.16 | oomph | iax will fix that |
| 06:54.19 | Ciber311 | making no sense |
| 06:54.32 | Ciber311 | other trunks from different providers are setup the same |
| 06:54.37 | oomph | what other providers you use for trunks? |
| 06:54.42 | bmg505 | Ciber311, check the firewall settings, if there is one in between |
| 06:54.59 | Ciber311 | bmg505: they all use the same ports :P |
| 06:55.16 | Ciber311 | i'd get no sound from the others if it was the firewall |
| 06:55.30 | Ciber311 | and it worked fine in a previous install |
| 06:55.37 | *** join/#asterisk edorsey (n=edorsey@24-119-133-217.cpe.cableone.net) |
| 06:55.43 | Ciber311 | router hasn't changed heh |
| 06:56.25 | *** join/#asterisk X-Gen (n=X-Gen@dsl-145-212-201.telkomadsl.co.za) |
| 06:57.23 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
| 06:58.26 | oomph | http://www.asteriskguru.com/tutorials/iax_conf.html |
| 06:58.40 | oomph | you using tribox or a self installed install? |
| 06:59.08 | oomph | Ciber311: what other DID providers would you recommend? |
| 06:59.35 | Ciber311 | built from source on osx oomph :) |
| 06:59.42 | oomph | i only tried junction and voicepulse |
| 06:59.48 | Ciber311 | you'd get shot using trixbox in here :P |
| 06:59.50 | oomph | thats cool |
| 07:00.14 | oomph | hehe |
| 07:00.20 | Ciber311 | i'm pretty happy with voxbone |
| 07:00.26 | Ciber311 | they have two POP's |
| 07:00.30 | Ciber311 | and answer emails very fast |
| 07:00.34 | oomph | where are they located? |
| 07:00.37 | oomph | the pops that is |
| 07:00.50 | Ciber311 | one in belgium |
| 07:00.50 | oomph | they got any type of support for e911? |
| 07:00.54 | Ciber311 | and the other in nyc |
| 07:01.04 | Ciber311 | they don't have outgoing calls |
| 07:01.14 | Ciber311 | strictly incoming |
| 07:01.25 | oomph | interesting |
| 07:01.37 | Ciber311 | i also have an axvoice account |
| 07:01.40 | oomph | what do you use for outgoing? |
| 07:01.41 | Ciber311 | no issues with them |
| 07:02.01 | Ciber311 | they answer the phone quickly(tech support) and they use asterisk |
| 07:02.16 | Ciber311 | oomph: axvoice |
| 07:02.25 | Ciber311 | also have a vonage account lol |
| 07:02.33 | oomph | hehe |
| 07:02.38 | oomph | why so many? |
| 07:02.47 | Ciber311 | i have issues :P |
| 07:02.53 | oomph | you should use skype for outgoing :P |
| 07:03.00 | oomph | free |
| 07:03.11 | oomph | too bad they are so proprietary |
| 07:03.19 | oomph | and dont just use sip |
| 07:03.20 | Ciber311 | yeah so it's useless |
| 07:03.26 | Ciber311 | i like using my polycom 5001 |
| 07:03.29 | Ciber311 | 501 |
| 07:03.34 | oomph | i got a polycom 601 |
| 07:03.40 | Ciber311 | nice |
| 07:03.56 | Ciber311 | they need to lower the price on that sidecar though |
| 07:03.57 | Ciber311 | lol |
| 07:03.58 | oomph | i need to configure it again |
| 07:04.13 | oomph | yeah that sidecar is stee |
| 07:04.15 | oomph | p |
| 07:04.29 | oomph | you ever use Voipbuster with your polycom? |
| 07:04.36 | Ciber311 | nope |
| 07:04.39 | Ciber311 | what's that? lol |
| 07:04.49 | oomph | they got sip proxies and cheap outbound calling :) |
| 07:05.03 | Ciber311 | hmm |
| 07:05.07 | Ciber311 | how cheap? |
| 07:05.16 | oomph | real cheap |
| 07:05.24 | oomph | like free to several countries including US |
| 07:05.41 | oomph | and they actually work reasonably well |
| 07:06.06 | oomph | ive had a few call qualitie isssues with them here and there but overall not much |
| 07:07.33 | oomph | http://www.voipbuster.com/en/index.html |
| 07:07.47 | *** join/#asterisk Sponge_bob (n=None@cpe-66-27-162-13.socal.res.rr.com) |
| 07:09.02 | Ciber311 | oh |
| 07:09.07 | Ciber311 | i remember these clowns |
| 07:09.12 | Ciber311 | that you gotta buy credits |
| 07:09.14 | oomph | i spent like 12 bucks with them and talk for free overseas to friends on it lots |
| 07:09.18 | Ciber311 | and still their crap to make an account |
| 07:09.33 | oomph | you dont gotta use their software |
| 07:09.38 | Ciber311 | i know |
| 07:09.50 | Ciber311 | all of those countries are useless to me though |
| 07:09.51 | oomph | depending on what you do with it |
| 07:10.00 | oomph | its not bad |
| 07:10.08 | oomph | what country you looking for. |
| 07:10.20 | oomph | they got a few sister sites |
| 07:10.24 | oomph | that have different countries |
| 07:10.29 | Ciber311 | i need DID's in the philippines heh |
| 07:10.44 | budmang | anyone good with queues? When ever I add more then 3 static agents we get all circuts are busy now. |
| 07:11.59 | *** join/#asterisk Maroder (n=drago@fanatici.net) |
| 07:13.32 | oomph | budmang: sorry, i dont currently have an asterisk installation to reference with. I've only used it here and there as a hobby |
| 07:14.00 | oomph | Ciber311: let me know if you happen to get your polycom to work with voipbuster |
| 07:14.14 | Ciber311 | you haven't? |
| 07:14.23 | Ciber311 | why would the phone matter? |
| 07:14.54 | oomph | because i dont have an asterisk install atm :P |
| 07:14.59 | Ciber311 | oh |
| 07:15.00 | Ciber311 | lol |
| 07:15.09 | Ciber311 | i don't do anything direct with my phones :P |
| 07:15.13 | Ciber311 | all goes through asterisk |
| 07:15.21 | oomph | yeah, gotcha |
| 07:15.24 | oomph | i would too |
| 07:15.35 | oomph | if i had a pbx to interface with |
| 07:15.59 | oomph | dont get me wrong |
| 07:16.49 | oomph | ive installed it before and used it like that |
| 07:17.30 | oomph | mostly for experimenting |
| 07:18.21 | oomph | Ciber311: which axvoice plan do you use? |
| 07:19.01 | Ciber311 | the unlimited one |
| 07:19.17 | Ciber311 | usa/can |
| 07:19.31 | Ciber311 | i only need to call the US and CAN |
| 07:19.49 | oomph | why dont u just use vonage for that |
| 07:19.53 | oomph | since you already got it |
| 07:20.04 | Ciber311 | vonage won't let me use my 501 |
| 07:20.06 | Ciber311 | :P |
| 07:20.16 | oomph | good point... |
| 07:20.30 | oomph | i kinda like having a cordless at home though |
| 07:21.05 | Ciber311 | i can't stand cordless phones |
| 07:21.10 | Ciber311 | i'm always at my desk anyway |
| 07:21.15 | Ciber311 | or near my bed |
| 07:21.19 | Ciber311 | both have 501's |
| 07:21.33 | oomph | yeah i got 2 601s and a grandstream |
| 07:21.36 | Ciber311 | my wife is one of those soft spoken people |
| 07:21.50 | Ciber311 | i have no idea what she's saying on anything but the 501's |
| 07:21.59 | oomph | hehe |
| 07:22.09 | Ciber311 | with the volume maxed of course |
| 07:22.28 | oomph | i like cordless phones at home. its nice to get up walk around with |
| 07:23.06 | oomph | why do you have vonage then if you have no use for it> |
| 07:23.07 | Ciber311 | i don't do that lol |
| 07:23.15 | Ciber311 | i like sitting |
| 07:23.16 | Ciber311 | lol |
| 07:23.26 | Ciber311 | oh i do use vonage |
| 07:24.46 | oomph | for what? |
| 07:24.53 | oomph | a second line? |
| 07:25.23 | Ciber311 | pretty much |
| 07:25.40 | oomph | for your cordless P |
| 07:25.47 | Ciber311 | yep lol |
| 07:26.02 | oomph | lol |
| 07:26.10 | Ciber311 | women :P |
| 07:27.15 | *** part/#asterisk nailbags (n=nailbags@c220-237-123-137.randw1.nsw.optusnet.com.au) |
| 07:28.28 | *** join/#asterisk dlynes_laptop (n=dlynes@S01060016b6c052ee.vc.shawcable.net) |
| 07:28.50 | Ciber311 | dlynes_laptop: !!!!!!!!!! |
| 07:28.59 | dlynes_laptop | Heya Ciber311 |
| 07:30.03 | Ciber311 | Aug 6 03:29:19 NOTICE[8052]: chan_iax2.c:6983 socket_read: Rejected connect attempt from 65.39.204.69, requested/capability 0x2/0xff03 incompatible with our capability 0xf80c. |
| 07:30.03 | Ciber311 | <PROTECTED> |
| 07:30.49 | dlynes_laptop | Ciber311: it's quite true |
| 07:30.50 | Ciber311 | lets try that again |
| 07:31.01 | dlynes_laptop | 0x2 is incompatible with 0xf80c |
| 07:31.09 | russellb | d'uh :-p |
| 07:31.34 | dlynes_laptop | wherein lies the disbelief? |
| 07:31.43 | hads|home | :) |
| 07:31.51 | Ciber311 | what the heck does the error actually mean? codec issue right? |
| 07:31.57 | russellb | i mean, i could explain what that means if anyone cares :) |
| 07:31.58 | dlynes_laptop | Ciber311: correct |
| 07:32.45 | dlynes_laptop | Ciber311: You've got it marked as a capability of 0xff03, it's requesting 0x02, but you can't support 0x02 because you can only support 0xf80c |
| 07:32.52 | dlynes_laptop | Ciber311: it's a simple bitmask |
| 07:32.59 | russellb | they are hex representations of a number that represents codec capability. each bit in the number represents a different codec |
| 07:33.11 | russellb | "show codecs" shows you the value of each codec |
| 07:33.19 | russellb | add up the ones you support and you get your bit mask |
| 07:33.23 | dlynes_laptop | Ciber311: 0x02 is gsm |
| 07:33.30 | dlynes_laptop | Ciber311: one side doesn't support it |
| 07:33.31 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
| 07:34.31 | Ciber311 | hmm |
| 07:34.38 | dlynes_laptop | but those both look like impossible capabilities to me |
| 07:34.40 | Ciber311 | someone on the forums said voxbone allows gsm |
| 07:35.04 | dlynes_laptop | 0x0FFF is possible, and 0xF0FFF is possible, but not 0xFFFF |
| 07:35.54 | dlynes_laptop | 1<<11 through 1<<15 do not exist |
| 07:36.52 | dlynes_laptop | hell |
| 07:37.19 | dlynes_laptop | even anthing between 0x0700 and 0x10000 isn't possible |
| 07:37.28 | dlynes_laptop | s/anthing/anything/ |
| 07:37.39 | dlynes_laptop | erm |
| 07:37.43 | dlynes_laptop | 0x07ff i mean |
| 07:38.06 | Assid | my binary sucks :( |
| 07:38.18 | dlynes_laptop | well |
| 07:38.29 | dlynes_laptop | it's possible to specify, but asterisk doesn't support those codecs |
| 07:38.48 | Ciber311 | lol |
| 07:38.52 | Ciber311 | damned thing |
| 07:41.25 | *** join/#asterisk tempest1 (n=asf@adsl-144-57-36.chs.bellsouth.net) |
| 07:41.40 | Ciber311 | lol |
| 07:41.43 | Ciber311 | set it to allow all |
| 07:41.46 | Ciber311 | and it works |
| 07:43.44 | dlynes_laptop | well, you told it not to allow gsm, and it was attempting to send gsm |
| 07:44.09 | Ciber311 | actually it had gsm as allow lol |
| 07:44.18 | Ciber311 | along with ulaw and alaw |
| 07:44.24 | dlynes_laptop | Ciber311: on both sides? |
| 07:44.31 | Ciber311 | set allow all under general and now it works |
| 07:44.48 | dlynes_laptop | Ciber311: were you disallowing gsm under general anywhere? |
| 07:45.02 | Ciber311 | no |
| 07:45.31 | dlynes_laptop | Ciber311: or maybe you were allowing gsm for voxbone, and disallowing it for somewhere else |
| 07:45.43 | dlynes_laptop | Ciber311: and you only had format_gsm.so loaded and not codec_gsm.so |
| 07:45.57 | *** join/#asterisk frodonbash (n=frodon@AMarseille-153-1-84-237.w86-209.abo.wanadoo.fr) |
| 07:46.41 | Ciber311 | well |
| 07:46.51 | Ciber311 | just removed allow all from general |
| 07:46.57 | Ciber311 | and added allow gsm |
| 07:47.00 | Ciber311 | and it works the same |
| 07:47.21 | dlynes_laptop | Ciber311: default == disallow=all, allow=ulaw,alaw |
| 07:47.25 | Ciber311 | guess it wasn't allowing gsm even though i didn't specifically disallow it |
| 07:47.54 | *** join/#asterisk af_ (n=af@ip-192-212.sn2.eutelia.it) |
| 07:48.11 | dlynes_laptop | Ciber311: oh yeah...and voice...can't remember which codec number that is |
| 07:49.33 | Ciber311 | wow |
| 07:49.40 | Ciber311 | iax is so much less annoying than sip :P |
| 07:50.22 | dlynes_laptop | Ciber311: those codec defaults are the same for all channels...not just iax |
| 07:50.35 | Ciber311 | i meant in general :P |
| 07:50.42 | dlynes_laptop | how so? |
| 07:50.58 | dlynes_laptop | you mean for traversing firewalls? |
| 07:50.59 | Ciber311 | i've been trying to get incoming sound on voxbone for hours |
| 07:51.03 | Ciber311 | yeah |
| 07:51.07 | Ciber311 | ports are open |
| 07:51.19 | Ciber311 | i can get two way sound from other providers |
| 07:51.29 | Ciber311 | that voxbone just wasn't working |
| 07:51.38 | Ciber311 | was working before in a different setup |
| 07:51.45 | Ciber311 | set to iax and its working :P |
| 07:56.13 | oomph | yeah i use iax whenever possible |
| 07:56.47 | oomph | what codec does vonage use |
| 07:59.24 | Ciber311 | who knows :P |
| 08:00.53 | *** join/#asterisk potsboy (n=chrisg@dsl-145-231-33.telkomadsl.co.za) |
| 08:03.27 | docelmo | oomph ulaw |
| 08:15.01 | *** join/#asterisk Qwell_ (n=north@unaffiliated/qwell) |
| 08:15.01 | *** mode/#asterisk [+o Qwell_] by ChanServ |
| 08:21.15 | queuetue | I have a digium TDM400P board in a machine with two FXO cards in it. zap show status sees it, and zap show channels shows three entries - pseudo, 3 and 4. 4 has a local PSTN line plugged into it, and when I call that line, nothing happens on the console, and the line just rings. If I issue a Dial(Zap/g4/<myhomenumber>), I get "Unable to create channel of type 'Zap' (cause 0 - Unknown)" ... Is there some important step I've missed |
| 08:21.15 | queuetue | ? Has this hardware gone south? |
| 08:24.19 | dlynes_laptop | queuetue: pb your zapata.conf and zaptel.conf files (without all the commented out lines) |
| 08:24.55 | oomph | docelmo: what type of bandwidth does ulaw take up again? |
| 08:25.08 | docelmo | 80k |
| 08:25.11 | oomph | how come they dont use something like g729 |
| 08:25.15 | dlynes_laptop | about 100K or so with overhead |
| 08:25.16 | docelmo | fax |
| 08:25.56 | oomph | no wonder vonage has issues for my uncle when he calls me from south america |
| 08:26.16 | oomph | he's on a slower connection there |
| 08:26.32 | oomph | skpe and some sip providers sound fine |
| 08:26.41 | oomph | but vonage can be flakey for em |
| 08:26.49 | *** join/#asterisk adorah (n=Administ@84.94.208.77.cable.012.net.il) |
| 08:27.14 | Maroder | how i can use my asterisk as gatekeeper ? |
| 08:27.20 | Maroder | h323 |
| 08:28.50 | dlynes_laptop | Maroder: i don't know if h323 under asterisk can even be configured for that, but there's a number of h323 modules for asterisk |
| 08:29.09 | dlynes_laptop | Maroder: the one in asterisk-addons may or may not be capable of gatekeeper mode |
| 08:29.29 | dlynes_laptop | Maroder: from what i understand, none of the h323 modules for asterisk are actually stable, however |
| 08:29.42 | Maroder | in the moment my ast connect to gnugk for the h323 connections |
| 08:30.02 | Maroder | but i want to remove gnugk ... |
| 08:37.24 | queuetue | dlynes_laptop: Ok, I will - sorry, got pulled away (And did not expect a quick answer at this hour :) ). |
| 08:37.48 | *** join/#asterisk pengyong (n=lala@222.188.132.35) |
| 08:41.10 | *** join/#asterisk threat2 (n=threat@60-240-43-214.static.tpgi.com.au) |
| 08:41.14 | threat2 | hey |
| 08:41.20 | queuetue | dlynes_laptop: http://pastebin.ca/118268 |
| 08:42.21 | dlynes_laptop | queuetue: Dial(Zap/3/18005551212) or Dial(Zap/4/18005551212) |
| 08:42.36 | dlynes_laptop | queuetue: You haven't defined any groups, so there won't be a g3 or g4 |
| 08:47.01 | queuetue | dlynes_laptop: Oh! When I get rid of the "g", it does indeed dial, but all I hear is silence. Thanks for getting me one step closer. |
| 08:47.30 | queuetue | I suppose it's time to check cabling for the 50'th time. :) |
| 08:47.42 | dlynes_laptop | I suspect you don't have dial tone |
| 08:48.09 | dlynes_laptop | Have you tried hooking a butt set or an analog phone directly up to the analog line? |
| 08:48.32 | budmang | anyone good with queues? When ever I add more then 3 static agents we get all circuts are busy now. |
| 08:50.43 | queuetue | dlynes_laptop: Yes, we've been using these lines for a year with an A@H box and an external fax machine. |
| 08:50.57 | dlynes_laptop | Just check for dial tone |
| 08:50.59 | dlynes_laptop | humor me :) |
| 08:51.11 | dlynes_laptop | but chekc for dial tone while the digium card is hooked up to it |
| 08:51.21 | hads|home | Yeah, either that or you are dialling some digits before it's ready. |
| 08:51.39 | queuetue | dlynes_laptop: It'll be a wile - it's only 5:00 am here and physical access is past my partner's wife's sleeping body. :) |
| 08:52.03 | dlynes_laptop | tell her to drag her ass out of bed then :) |
| 08:52.23 | queuetue | dlynes_laptop: Working on it, but the partner doesn't seem as willing as I'd like him to be . ;) |
| 08:52.40 | dlynes_laptop | find a new partner :) |
| 08:52.52 | queuetue | This one has the money. :) |
| 08:53.18 | dlynes_laptop | minor problem |
| 08:53.40 | hads|home | How about trying a Dial(Zap/4/) just to see what happens. |
| 08:53.48 | queuetue | hads: Ok. |
| 08:53.55 | hads|home | (excuse me butting in halfway through) |
| 08:54.22 | queuetue | I can't do that straight from the CLI can I? It had to be done through a utility extension or a real dialplan? |
| 08:54.37 | queuetue | hads|home: Butt away, I can use all the help I can get. |
| 08:55.00 | hads|home | Yeah, just put a dummy extension in the dialplan |
| 08:55.40 | queuetue | It says "Called 4/" and then silence. |
| 08:56.04 | hads|home | Then you'll have to do what dlynes_laptop said. |
| 08:56.24 | hads|home | I was thining that if it gave you a dialtone it would save you having to do the waking up |
| 08:56.35 | hads|home | s/thining/thinking/ |
| 08:57.24 | queuetue | hads|home: The "waking up" is just a pipedream of mine. ;) I'll be three hours before I can get into that closet. :) |
| 08:58.09 | queuetue | hads|home: dlynes_laptop: But the config seems solid? If the cabling, hardware, and telco is working, this should be answering? |
| 08:58.32 | dlynes_laptop | queuetue: is it a new card? |
| 08:58.55 | queuetue | dlynes_laptop: It's an old card, with a new second FXO added. |
| 08:59.13 | dlynes_laptop | Ok, so the card's worked in the past? |
| 08:59.51 | queuetue | dlynes_laptop: Yes, but now it's also in a new computer with a fresh linux install, although I don't think that could matter. |
| 09:00.05 | dlynes_laptop | ok, so supposedly channel 3 works then, right? |
| 09:00.09 | dlynes_laptop | i.e. it worked before? |
| 09:00.30 | queuetue | Channel 4 has worked fine for a year, yes. |
| 09:00.36 | dlynes_laptop | oh |
| 09:00.40 | dlynes_laptop | channel 3 is new then? |
| 09:00.43 | queuetue | Yes. |
| 09:00.55 | dlynes_laptop | but it's channel 4 that's not working? |
| 09:01.13 | queuetue | Chennel 3 is not attached atm - still feeding fax machine until this starts to answer. |
| 09:01.43 | queuetue | I assume channel 3 and 4 are of the same status, and this is a config issue. |
| 09:02.04 | queuetue | I will have him cable both lines up and test independantly, though. |
| 09:02.52 | queuetue | Basically, the A@H to running my own dialplan process was more involved than I expected. :) |
| 09:02.55 | dlynes_laptop | have you tried configuring it for channel 1, 2, 3, and 4, in case someone took the modules out, and plugged them into channel 1 and 2 instead? |
| 09:03.22 | dlynes_laptop | also, are you loading wctdm, or wcfxo? |
| 09:03.56 | dlynes_laptop | lsmod | grep w |
| 09:03.56 | dlynes_laptop | lsmod | grep wc |
| 09:03.56 | queuetue | dlynes_laptop: I have done a physical inspection to insure that did not happen. The card is labelled on the outside, by the ports, as 3 + 4, and those leds are lit. |
| 09:04.01 | dlynes_laptop | ah...never used a digium tdm400p, so wouldn't know |
| 09:04.31 | queuetue | dlynes_laptop: zaptel 191748 16 wcusb,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wct4xxp,tor2,wctdm |
| 09:04.40 | dlynes_laptop | holy crap |
| 09:04.46 | dlynes_laptop | why so much garbage loaded? |
| 09:04.55 | dlynes_laptop | you only need wctdm and zaptel loaded |
| 09:04.59 | *** join/#asterisk my007ms (n=noor@217.139.224.194) |
| 09:05.00 | dlynes_laptop | and crc_ccitt |
| 09:05.34 | dlynes_laptop | also, what do you get when you type zap show status? |
| 09:05.39 | queuetue | The zaptel install did this by itself. |
| 09:05.48 | queuetue | Shoudl I comment the other out? |
| 09:06.22 | dlynes_laptop | hads|home: He should only get one line on zap show status for a tdm400, right? |
| 09:06.38 | dlynes_laptop | hads|home: not one for every channel, plus a pseudo? |
| 09:06.58 | hads|home | Yeah correct |
| 09:07.47 | dlynes_laptop | queuetue: comment out everything except wctdm (it'll autoload zaptel on its own) |
| 09:08.52 | hads|home | Yeah, what he said. Try getting rid of the extra modules to clean things up a bit. |
| 09:09.04 | queuetue | lsmod |grep zaptel == zaptel 191748 9 wctdm \ crc_ccitt 2304 3 hisax,irda,zaptel |
| 09:10.17 | dlynes_laptop | queuetue: good boy |
| 09:10.28 | dlynes_laptop | queuetue: now do a ztcfg -vvvvvvvvvvvv |
| 09:10.33 | dlynes_laptop | queuetue: and restart asterisk |
| 09:10.41 | dlynes_laptop | queuetue: then do a zap show status |
| 09:11.26 | queuetue | Wildcard TDM400P REV I Board 1 OK 0 0 0 |
| 09:11.43 | dlynes_laptop | queuetue: and zap show channels? |
| 09:12.05 | queuetue | pseudo 3 and 4 |
| 09:12.16 | queuetue | (Same as pastebin) |
| 09:12.22 | dlynes_laptop | queuetue: try Dial(Zap/4/18005551212) again |
| 09:12.38 | dlynes_laptop | queuetue: if that doesn't work, try Dial(Zap/3/18005551212) |
| 09:12.46 | queuetue | dead air, both channels. |
| 09:13.00 | dlynes_laptop | queuetue: the phone lines must be unplugged, then |
| 09:13.14 | hads|home | Does sound like it |
| 09:13.49 | hads|home | You should also get 'starting simple switch' at least when you dial in too. |
| 09:14.25 | queuetue | Will I hear call progress with zap, or will audio only come through when the call is connected? |
| 09:14.30 | queuetue | I just don't remember. |
| 09:15.09 | hads|home | The first thing you should hear is ringing audio from the far wnd |
| 09:15.29 | queuetue | I do not see any "switch" comments in CLI. I do see "Executing Dial("SIP/306-081970e0", "Zap/3/") in new stack" |
| 09:16.48 | hads|home | Just to be sure, do a 'set verbose 10' and 'set debug 10' and pastebin the CLI output from when you try and place a call. So we can see for sure what's going on. |
| 09:17.42 | dlynes_laptop | hads|home: set debug 10 isn't going to do squat, unless debug mode is on for something |
| 09:17.44 | *** join/#asterisk sav_mcfly (n=R00T@pergamo.zonaz.net) |
| 09:17.56 | *** part/#asterisk sav_mcfly (n=R00T@pergamo.zonaz.net) |
| 09:18.33 | hads|home | Yeah, there is 'general' debug output isn't there? |
| 09:19.37 | dlynes_laptop | yeah, but i think it's either on or off |
| 09:19.46 | dlynes_laptop | because even at debug level 0 it still gets logged |
| 09:20.51 | hads|home | Maybe you're right. I'm sure I remember seeing debug output on the console after turning debug up. No matter :) |
| 09:26.33 | queuetue | I have this section: [outgoing-theworld] \n exten => _.,1,Macro(call-ext,${EXTEN}) in place, and it matches *everything* ... how can I rewrite this to only match 7+ digits? (exten => _1NXXNXXNXXXX,1,Macro(call-ext,${EXTEN}) , which I expected to work at least for 10-digits, gets ignored.) |
| 09:26.42 | *** join/#asterisk EyeCue (n=eyecue@unaffiliated/eyecue) |
| 09:27.24 | queuetue | (This firs after every call to my utility extensions, I've noticed. I just want to eliminate it as a problem before more testing.) |
| 09:27.47 | hads|home | That depends on what 7+ digits you are trying to match _XXXXXXX. will match 7+ |
| 09:28.10 | hads|home | You should use _X. rather than _. anyway. |
| 09:28.36 | queuetue | hads: I tried that first, and it seemed to stop matching. |
| 09:29.41 | queuetue | Seems t work fine. I think I replaced it with _X, not _X. |
| 09:30.40 | hads|home | Even so _X. will match _everything_ with a digit in it. You only want to do that if you really want to do that. |
| 09:30.45 | queuetue | _XXXXXX. works to keep local extensions from triggering "world" script, too. |
| 09:31.50 | queuetue | script? section? what are [foo]'s called in asterisk dialplan? |
| 09:31.58 | hads|home | context |
| 09:32.23 | queuetue | There is no zap debug options, right? |
| 09:33.42 | dlynes_laptop | queuetue: not for analog lines, no |
| 09:33.58 | hads|home | There isn't much to debug :) |
| 09:33.59 | dlynes_laptop | queuetue: there is for pri lines though |
| 09:34.34 | queuetue | What is pri? I've seen that a few times now. is that "big boy" T1 stuff that does not apply to me? |
| 09:34.44 | macTijn | queuetue: yes |
| 09:35.17 | macTijn | queuetue: PRI (Primary Rate Interface) is T1 in north-america and E1 everywhere else ;) |
| 09:35.28 | queuetue | Ok, thanks. |
| 09:35.29 | macTijn | T1 has 24 channels, E1 has 30 |
| 09:36.11 | queuetue | hads|home: http://pastebin.ca/118301 |
| 09:36.32 | queuetue | This is from full log. |
| 09:37.11 | queuetue | CLI Showed -- Executing Dial("SIP/306-0819dad8", "Zap/4/18005551212") in new stack \ -- Called 4/12129378857 \ -- Hungup 'Zap/4-1' |
| 09:37.26 | queuetue | Oops, ignore that #. :) Forgot to fix one. :) |
| 09:39.46 | hads|home | Can't see anything weird there, except don't know about queues. |
| 09:40.25 | queuetue | I odn't either, and am not intentionally using any. |
| 09:41.15 | queuetue | Does app_queue maybe just monitor all channels, whether queues are configured or not? |
| 09:41.28 | queuetue | SIP/306 is me. |
| 09:45.16 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
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| 10:11.41 | *** join/#asterisk TypMic (n=TypMic@c-68-49-188-94.hsd1.md.comcast.net) |
| 10:12.06 | TypMic | hello, any audio experts out there |
| 10:12.57 | coppice | i can hear, if that's what you mean |
| 10:13.45 | hads|home | hah |
| 10:14.05 | puppet | i can hear too |
| 10:14.09 | puppet | i can even like sing but bad |
| 10:14.38 | coppice | seems we've enough folk here to start an audio consultancy |
| 10:15.20 | TypMic | good joke but looking for someone with experience getting a digium tdm work with a phoneline connecting phones thru a public PSTN to a astrisk softphone |
| 10:15.54 | coppice | then why didn't you ask a meaningful question in the first place. what doesn't work right now? |
| 10:16.15 | *** join/#asterisk UlbabraB (n=filippo@host241-43-static.72-81-b.business.telecomitalia.it) |
| 10:16.15 | puppet | and what does have to do with audio expert? |
| 10:16.23 | puppet | doesnt that have to do with like ZAP |
| 10:18.14 | TypMic | point taken...anyway...audio is very very low between the legacy phone system (PSTN) and any softphone, I suspect either mis-matched codex or lost packets (have not had time to test for lost packets yet). I was wondering if anyone has experienced this as well and found a solution |
| 10:18.40 | hads|home | rxgain? |
| 10:18.43 | puppet | by memory ( i dont have zp self) there is an option to increase the volume |
| 10:18.49 | puppet | hads|home: thanks |
| 10:20.31 | TypMic | ok, so a asterisk newbie question.... is rxgain a asterisk option, and which file is modified...extensions.conf? |
| 10:21.01 | hads|home | TypMic: It's a zaptel option, so it's in zapata.conf |
| 10:21.19 | puppet | http://www.google.se/search?hl=sv&q=zap+asterisk+low+volume&btnG=Google-s%C3%B6kning&meta= |
| 10:21.26 | hads|home | More correctly a chan_zap option. |
| 10:22.14 | puppet | http://www.voip-info.org/wiki/index.php?page=Asterisk+config+zapata.conf |
| 10:23.39 | puppet | typmic: voip-info is your friend |
| 10:24.12 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
| 10:26.53 | TypMic | ok, I'll give that a try, yeah I totally agree voip-info is great, so are the folks on this IRC for those, like ME who are so very green behind the gills with this world....and not sure what we are reading and learning....learning curve is long and slow dont you know........thanks... |
| 10:27.17 | puppet | typmic: you learn more if you try to search self first |
| 10:27.40 | *** join/#asterisk BurtyB (n=chris@cpc2-nwrk1-0-0-cust1009.nott.cable.ntl.com) |
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| 10:29.57 | [LucaS] | Ola |
| 10:30.58 | [LucaS] | Which cidsignalling I've to use in Italy? |
| 10:31.38 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
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| 10:38.37 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 11:02.40 | *** join/#asterisk numbshot (n=numbshot@189.129.71.64) |
| 11:05.20 | numbshot | hello, i recently got a did number (voxbone) and now some extensions don't work, for example, i can dial 1 with success but if i dial "331" asterisk tries to dial the first digit "3" instead of the full 331 extension, any ideas please¿? (this only happens through voxbone did, if i dial this same context from another asterisk sip extension, then the full 331 extension is dialed) :| |
| 11:09.26 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust944.leed.cable.ntl.com) |
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| 11:47.09 | kFuQ | numbshot: kinda sounds like ur dialplan may be a little screwey.. |
| 11:49.02 | RoyK | numbshot: pastebin config and so on... |
| 11:52.52 | *** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net) |
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| 12:06.28 | numbshot | RoyK: pastebin.com is not responding me but please take a look at http://www.atenea.info/paste.html |
| 12:20.41 | RoyK | now that host= line in sip.conf looks funny |
| 12:21.17 | RoyK | anyway |
| 12:21.23 | *** join/#asterisk potsboy (n=chrisg@c2-68-2.vic.dial.mweb.co.za) |
| 12:21.24 | RoyK | ah |
| 12:21.25 | RoyK | i see |
| 12:21.36 | RoyK | the extensions matching logic in asterisk is fscked |
| 12:21.40 | RoyK | by design |
| 12:22.32 | RoyK | hm. but you don't have a 3 context there |
| 12:23.55 | numbshot | it seems that i can only use 1 digit extensions |
| 12:24.13 | numbshot | in larger extensions only the first digit is used |
| 12:24.13 | RoyK | numbshot: well, that's the first match |
| 12:24.29 | RoyK | perhaps use Read instead |
| 12:24.41 | RoyK | Read and then Dial(Local/..... |
| 12:35.54 | docelmo | doubt it |
| 12:36.16 | RoyK | :P |
| 12:37.08 | *** join/#asterisk Ebola (n=Ebola@81-86-155-65.dsl.pipex.com) |
| 12:42.15 | *** join/#asterisk saftsack (n=saftsack@p54A7F03F.dip.t-dialin.net) |
| 13:00.26 | coppice | which is RFC2550? is that pancakes over IP? |
| 13:02.01 | *** join/#asterisk beyond (n=beyond@201-1-206-248.dsl.telesp.net.br) |
| 13:05.35 | *** join/#asterisk tsurk0 (n=tsurko@85.187.160.157) |
| 13:10.20 | RoyK | coppice: "Y10K and Beyond" |
| 13:10.26 | RoyK | http://www.faqs.org/rfcs/rfc2550.html |
| 13:11.23 | coppice | i'm sure they'll still be eating pancakes then |
| 13:13.04 | SwK_ | I still have been meaning to implement PFC2549 for rural areas |
| 13:14.59 | coppice | you can't achieve QoS over such an open network. that RFC is just a bogus addition to the well researched RFC1149 |
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| 13:46.31 | my007ms | it's asterisk but i can not knwo what is defualt damn username/password for freePBX |
| 13:55.38 | *** join/#asterisk rogier (n=rogier@83-67-dsl.ipact.nl) |
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| 14:00.52 | *** join/#asterisk evilbit (i=hhoffman@gateway/tor/x-7578481d394409f2) |
| 14:01.38 | evilbit | hi, so I'm having lots of trouble with teliax and I am thinking about getting a new provider. Are DID numbers portable? |
| 14:02.27 | my007ms | is there any othere file othere /etc/asterisk/modules.conf make modules lod |
| 14:02.58 | *** join/#asterisk Yoda-BZH (n=yoda-bzh@solene.yoda-bzh.net) |
| 14:04.32 | evilbit | don't believe so |
| 14:06.30 | RoyK | my007ms: no, why? |
| 14:09.21 | *** join/#asterisk sip_me (n=ask@80.179.11.31.static.012.net.il) |
| 14:11.18 | Corydon76-home | No, DIDs are not portable |
| 14:11.46 | sip_me | Hi, Is there a way to get asterisk to register ALL users without defining them one by one in the sip.conf (a security free configuration)? |
| 14:12.15 | EyeCue | might be better to use some form of backend sql integration |
| 14:12.22 | EyeCue | that way you can insert a record in per user |
| 14:12.47 | Corydon76-home | sip_me: you cannot register them, no, but you could allow all SIP calls through |
| 14:12.57 | Corydon76-home | but that won't allow them to be peers |
| 14:13.15 | *** join/#asterisk zigman (i=zigman@irc.zigman.de) |
| 14:13.41 | RoyK | http://www.youtube.com/watch?v=gyuwPetrmaY |
| 14:13.56 | Corydon76-home | Just set up a user with insecure=invite |
| 14:14.23 | sip_me | I am trying to get asterisk to server as a SIP peoxy - anyone that registers should from that moment all be able to generate and accept calls. |
| 14:16.03 | sip_me | Corydon76-home, I am not sure I follow. My clients start with REGISTER. When I was using Partysip - I could ask it to simply auto-register these clients. Can I do the same here? |
| 14:16.26 | Corydon76-home | sip_me: no, you can't. |
| 14:16.47 | Corydon76-home | There's no support for doing that |
| 14:17.40 | sip_me | Corydon76-home, OK. So I need to setup all users with a simple: [username] type=friend, host=dynamic |
| 14:17.45 | sip_me | Anything else? |
| 14:17.55 | Corydon76-home | Correct |
| 14:18.08 | sip_me | Great. |
| 14:18.36 | *** part/#asterisk Yoda-BZH (n=yoda-bzh@solene.yoda-bzh.net) |
| 14:28.25 | sip_me | mmm. I can register, but the invite fails with 404 |
| 14:28.46 | sip_me | I am sending in the invite the same URL that I used for registration |
| 14:29.36 | sip_me | In both cases I am using P01@192.168.0.173 whereas P01 is port 01 ( I also have port 02) and 192.168.0.173 is the IP address of the Asterisk server |
| 14:30.34 | sip_me | SIP Show Users indicates two registered users (P01 and P02) |
| 14:30.43 | queuetue | dlynes_laptop: hads|home: The cable shave been confirmed with an analog phone - both ports *are* properly connected. Still no audio coming from pots lines, though. |
| 14:30.58 | queuetue | file: Are you around by any chance? |
| 14:31.30 | sip_me | 192.168.0.173 is a registered domain in SIP Show Domains |
| 14:31.33 | sip_me | any clue? |
| 14:33.51 | *** join/#asterisk The_LightSide (n=DialT@dsl-145-200-161.telkomadsl.co.za) |
| 14:33.58 | The_LightSide | hi all! |
| 14:34.17 | evilbit | what does "[0 online, 0 offline, 1 unmonitored]" in iax2 show peers mean? |
| 14:34.26 | The_LightSide | can any1 tell me how to add audio before answer on a pri system? |
| 14:36.09 | The_LightSide | is any1 awake? ;p |
| 14:36.23 | E-bola | patience is a virtue ... |
| 14:36.36 | The_LightSide | indeed :) |
| 14:37.07 | Corydon76-home | evilbit: it means you have a single peer without state information |
| 14:37.21 | *** join/#asterisk ionix (n=ionix@p3101-ipbfp05miyazaki.miyazaki.ocn.ne.jp) |
| 14:37.24 | ionix | Hello everyone |
| 14:37.32 | ionix | something weird with the AGI on an outgoing call. |
| 14:37.33 | ionix | Aug 6 10:37:10 WARNING[18824112]: pbx.c:4072 ast_pbx_run_app: No such application 'call.php' |
| 14:38.01 | ionix | /var/lib/asterisk/agi-bin/call.php exists and is executable |
| 14:38.23 | RoyK | can you run it manually? |
| 14:39.54 | ionix | yes |
| 14:39.59 | *** join/#asterisk l-fy (n=pchitesc@yate/developer/l-fy) |
| 14:40.30 | ionix | I even tried to load one of the asterisk's example and it failed with the same error |
| 14:40.44 | ionix | is there a way to check asterisk's env to see where it takes the scripts/ |
| 14:41.18 | The_LightSide | does any1 know anything bout libpri? |
| 14:42.15 | l-fy | The_LightSide > shoot |
| 14:42.24 | *** join/#asterisk Trakkasure (n=Sgemtum@69-163-145-203.atlsfl.adelphia.net) |
| 14:42.54 | evilbit | Corydon76-home: thx, do peers normally maintain state? |
| 14:43.13 | The_LightSide | thanks l-fy, i need to add in "audio before answer" and was wonderiong how i do that? |
| 14:43.24 | The_LightSide | on a pri system |
| 14:43.59 | l-fy | The_LightSide > early media? |
| 14:44.24 | *** join/#asterisk iq (n=iq@unaffiliated/iq) |
| 14:44.56 | The_LightSide | the problem ive got is that i dial the number, the pri device dials etc, but the progress tones dont come thru |
| 14:45.08 | The_LightSide | until it gets a answer condition |
| 14:45.18 | l-fy | The_LightSide > i don't know how is this done in asterisk |
| 14:45.19 | The_LightSide | not sure what you mean by early media |
| 14:45.24 | *** join/#asterisk |dennis| (n=dennis@200.32.215.82) |
| 14:45.29 | l-fy | i had a similar problem like 3 years ago with asterisk |
| 14:45.37 | The_LightSide | lol |
| 14:45.41 | l-fy | early media is the media you get before the call |
| 14:45.56 | The_LightSide | such as progress tones |
| 14:46.00 | l-fy | yes |
| 14:46.03 | l-fy | exactly |
| 14:46.07 | l-fy | is also working on sip and h323 |
| 14:46.29 | The_LightSide | sip & iax work :/ |
| 14:48.42 | The_LightSide | well, thanks for the info... will do some more googling :) |
| 14:49.01 | *** join/#asterisk X-Gen (n=X-Gen@dsl-145-212-201.telkomadsl.co.za) |
| 14:50.39 | ionix | so, anyone knows why a script will be reported as not found even though it's there? |
| 14:51.45 | *** join/#asterisk linlin (n=linlin@c-67-184-230-25.hsd1.il.comcast.net) |
| 14:51.56 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
| 14:54.21 | *** join/#asterisk rogier (n=rogier@83-67-dsl.ipact.nl) |
| 14:57.39 | l-fy | hi RoyK |
| 15:00.55 | sip_me | Hi, I have a new Asterisk installation. I am succesful in registration. However my Invite fails with 404. Can anyone guide me with checking what is wrong? |
| 15:01.16 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 15:01.16 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:01.27 | Assid | try and set canreinvit=no for your sip accounts |
| 15:01.35 | Assid | canreinvite=no |
| 15:02.54 | sip_me | Assid, that does not seem to change the situation. |
| 15:03.05 | Assid | did you reload sip after that? |
| 15:03.33 | sip_me | I did "reload" fro the command line |
| 15:03.44 | Assid | what does verbose show |
| 15:04.32 | l-fy | 404 - can't find the extension |
| 15:04.48 | sip_me | It shows the packets in and out but I do not see any error or warning ot notice message (I am using SIP Debug) |
| 15:04.56 | queuetue | Are there any other applications that can talk to zapata cards? So I can try to confirm if it's a problem with the driver, hardware, or something broken in asterisk? |
| 15:05.08 | Assid | what extension sare you dialling.. is that there? |
| 15:05.41 | sip_me | I am dialing P01@192.168.0.173 (192.168.0.173 is my Asterisk server IP) |
| 15:05.44 | l-fy | queuetue > yate |
| 15:05.51 | l-fy | queuetue > http://yate.null.ro |
| 15:06.08 | l-fy | and also openpbx and freeswitch but those are based on the same modules as asterisk |
| 15:06.10 | Assid | the location you are dialling it from is within NAT ? |
| 15:06.13 | l-fy | so is not really a test |
| 15:06.24 | *** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135) |
| 15:06.48 | sip_me | The server and client PC are conected to the same local Switch - No NAT |
| 15:07.15 | Assid | and what does your sip context look like |
| 15:08.19 | queuetue | l-fy: Whats[ the setup process like to get it to actually talk to a zaptel card and shwo progress>? |
| 15:08.20 | *** join/#asterisk logicwrath (n=dfgaeefa@c-68-60-121-112.hsd1.mi.comcast.net) |
| 15:08.26 | sip_me | Where can I find my sip context? Is this the name I used for registration? |
| 15:08.40 | *** join/#asterisk af_ (n=af@ip-192-212.sn2.eutelia.it) |
| 15:08.45 | Assid | in your sip.conf where you have P01 |
| 15:09.18 | sip_me | [general] context=default |
| 15:09.32 | Assid | that isnt what i asked for |
| 15:09.51 | Assid | i said in your sip.conf .. the context where you defined P01 |
| 15:10.33 | sip_me | I also have regcontext=sipregisrations - no other context definitions |
| 15:10.39 | Assid | well |
| 15:10.45 | Assid | that answers your question |
| 15:10.50 | Assid | you dont have P01 defined |
| 15:10.53 | Assid | you cant invite it |
| 15:11.04 | sip_me | I have P01 and P01 defined. |
| 15:11.24 | l-fy | queuetue > can we please move to #yate for that? |
| 15:11.38 | Assid | well.. then you arent giving me the context then are you |
| 15:11.39 | sip_me | Each with type=friend, host=dynamic, |
| 15:12.06 | Assid | im asking you for the P01 definition.. and you still havent shown it to me |
| 15:13.32 | sip_me | [P01] type=friend, host=dynamic, insecure=port,invite regexten=P01, canreinvite=no |
| 15:13.54 | sip_me | (same for P02) |
| 15:16.49 | sip_me | Assid, is that what you ment? - This is the P01 definition that I added. |
| 15:17.21 | *** join/#asterisk Jimme (n=Jimbo@host-80-194-73-131.static.telewest.net) |
| 15:19.07 | Jimme | Hi, i am having some problems getting a siemens legacy pbx to dial out via asterisk. I have ths siemens connected to asterisk using a crossed T1 cable to span 2 of its PRI card. I am not trying to figure out how to get asterisk to give the siemens a dial tone to start the outgoing call process |
| 15:19.49 | Jimme | at the moment i just get "-- Extension '' in context 'inbound-from-siemens' from '692202' does not exist. Rejecting call on channel 0/31, span 2" when I hit 9 on the siemens to get an outside line |
| 15:29.44 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 15:31.45 | sip_me | Can anyone help with a 404 problem? My configuration is: [P01] type=friend, host=dynamic, insecure=port,invite regexten=P01, canreinvite=no |
| 15:32.14 | sip_me | and the same for a second user P02 - I am trying to call from P01 to P02 and gets a 404 to my Invite. |
| 15:32.59 | file | have you written dialplan logic to call them? |
| 15:33.24 | sip_me | NO! |
| 15:33.41 | file | then how can you call them? |
| 15:34.17 | sip_me | file, I am use to working with a proxy that have an auto-registration and does all this without my knowledge... |
| 15:34.41 | sip_me | where can I define the dialplan logic? |
| 15:34.51 | file | you won't get that with Asterisk because an extension is not a phone, it's a set of instructions that execute a set of applications |
| 15:34.57 | file | extensions.conf |
| 15:35.33 | queuetue | Hi file. I'm still not able to get that digium zap card to work. Do you have any ideas? |
| 15:36.20 | queuetue | (It was the TDM400 that needed you to patch * for me to get gsm audio working.) |
| 15:37.08 | file | queuetue: no but I suggest calling technical support tomorrow |
| 15:37.30 | queuetue | file: Ok, can do. |
| 15:37.57 | queuetue | file: (What's that likely to cost me?) |
| 15:38.11 | file | queuetue: you get free installation support with your Digium card |
| 15:38.51 | queuetue | file: I bought it through a reseller quite a while ago. But I'll give it a shot. |
| 15:39.15 | queuetue | (It's been in use for a year. Upgrade to 1.2.10 seems to have made it stop working.) |
| 15:39.26 | file | queuetue: plus addition of another module? |
| 15:40.07 | queuetue | It was working with the other module in place. Not sure if the other module was functional at the time, we weren't using it yet. |
| 15:40.36 | queuetue | It was added about a month before the upgrade. |
| 15:45.47 | *** join/#asterisk xtr (n=94752345@S0106000c41ed11e1.vf.shawcable.net) |
| 15:48.35 | *** join/#asterisk beyond (n=beyond@201-1-206-248.dsl.telesp.net.br) |
| 15:53.20 | dlynes_laptop | queuetue: you didn't mention that to me earlier |
| 15:53.44 | dlynes_laptop | queuetue: you led me to believe it had been in use for a year |
| 15:54.11 | dlynes_laptop | queuetue: oh...nvm |
| 15:54.42 | queuetue | dlynes_laptop: I wasn't trying to mislead anyone. :) Just trying to simplify. |
| 15:54.46 | dlynes_laptop | queuetue: btw....have you tried putting the two modules in slot 1 and 2, instead of 3 and 4? |
| 15:55.24 | queuetue | dlynes_laptop: No, that is on the list of configurations to try. I have tried each module in #4, an neither works alone. |
| 15:55.26 | dlynes_laptop | queuetue: and don't you go to sleep? |
| 15:55.39 | queuetue | Not much. |
| 15:56.00 | dlynes_laptop | queuetue: You've been on here since about 5am or so |
| 15:56.43 | queuetue | Closer to 3:30. |
| 15:56.44 | dlynes_laptop | queuetue: well, mind you, you're somewhere near toronto so....it must be the humidity...you probably couldn't sleep :) |
| 15:57.34 | dlynes_laptop | you couldn't pay me enough to move back to southern ontario and put up with that god awful humidity again :) |
| 15:57.45 | queuetue | My phones are also going to be down tomorrow morning for 4 days, so I'm trying to be here whenever someone who can help is. :) |
| 15:57.58 | *** join/#asterisk parag_ast (n=root@dxb-b110696.alshamil.net.ae) |
| 15:58.09 | queuetue | dlynes_laptop: My first summer in Montreal, and I agree. I'd much rather deal with February than July. |
| 15:58.27 | file | cookies for all! |
| 15:58.29 | queuetue | Cold is a manageble problem. |
| 15:58.47 | dlynes_laptop | yeah...but southern ontario winters suck, too |
| 15:58.55 | dlynes_laptop | the humidity makes -10 feel like -30 |
| 15:58.58 | parag_ast | Hi kindly tell me that if i don't want to play the sound on "All circuts are busy , kindly try after some time" what i need to do |
| 15:59.20 | dlynes_laptop | parag_ast: are you on a pri? |
| 15:59.29 | parag_ast | no |
| 15:59.35 | dlynes_laptop | analog? |
| 15:59.36 | parag_ast | just for VOIP route |
| 15:59.48 | parag_ast | suppose VOIP route is showing busy |
| 15:59.52 | dlynes_laptop | ah |
| 15:59.55 | parag_ast | so it plays this sound |
| 15:59.57 | dlynes_laptop | then play whatever you feel like playing |
| 16:00.03 | parag_ast | all circutes are busy |
| 16:00.06 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 16:00.10 | parag_ast | i don't want to play anyting |
| 16:00.13 | dlynes_laptop | the voip end doesn't send you back a recording |
| 16:00.20 | parag_ast | no |
| 16:00.22 | dlynes_laptop | it sends you back a sip or iax error code |
| 16:00.24 | file | it could |
| 16:00.26 | parag_ast | i got a error |
| 16:00.27 | *** join/#asterisk bmg505 (n=leon@c1-223-1.rndf.isadsl.co.za) |
| 16:00.31 | queuetue | parag_ast: You just want them to guess what the silence means? |
| 16:00.35 | parag_ast | everyone is busy/congested |
| 16:00.44 | Dr-Linux | dlynes_laptop: hey :) |
| 16:00.46 | parag_ast | yes |
| 16:00.50 | dlynes_laptop | parag_ast: so play whatever recorfding you think is appropriate |
| 16:00.58 | parag_ast | how can i do that |
| 16:01.20 | dlynes_laptop | parag_ast: or use playtones(congestion) ; |
| 16:01.28 | dlynes_laptop | parag_ast: or playtones(busy) |
| 16:01.38 | parag_ast | where |
| 16:01.54 | dlynes_laptop | parag_ast: Playback(/var/lib/asterisk/sounds/all-circuits-are-busy) |
| 16:01.55 | parag_ast | or i can set someting in DIAL |
| 16:01.57 | parag_ast | command |
| 16:02.03 | jart | hmm |
| 16:02.16 | dlynes_laptop | parag_ast: nope |
| 16:02.30 | jart | jart: this is a test |
| 16:02.35 | file | test failed |
| 16:02.36 | dlynes_laptop | parag_ast: do it based on ummmm...trying to remember the built in variable |
| 16:02.52 | file | DIALSTATUS? |
| 16:03.09 | parag_ast | DIALSTATUS = busy |
| 16:03.47 | jart | file: talk to me and say jart at the beginning |
| 16:03.51 | file | jart: moo |
| 16:03.54 | dlynes_laptop | Oh yeha |
| 16:03.59 | dlynes_laptop | DIALSTATUS...that's what it was :) |
| 16:04.04 | jart | test did fail :( |
| 16:04.09 | parag_ast | where |
| 16:04.18 | Dr-Linux | dlynes_laptop: one of cisco 7940 is on same LAN where asterisk box located. this phone user complaining me that voice is breaking .. i really don't understand while there is no VOIP. any clue? |
| 16:04.57 | dlynes_laptop | Dr-Linux: same LAN? how about same subnet? |
| 16:06.19 | evilbit | is there a site for agi scripts? I'm looking to write one to allow a user to call in, enter their vmail box/pin and then get a line to dial back out on |
| 16:06.32 | Dr-Linux | dlynes_laptop: yeah on same subnet |
| 16:06.43 | dlynes_laptop | evilbit: check the documentation for voicemail.conf |
| 16:06.45 | Jimme | sorry to reask so soon, but does anyone have any ideas on my problem, i have run out of things to try |
| 16:06.56 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 16:06.56 | dlynes_laptop | evilbit: no point writing an agi script for it....app_voicemail already does it |
| 16:07.11 | file | VMAuthenticate and DISA will work too |
| 16:07.19 | evilbit | dlynes_laptop: and it provides a line back out? |
| 16:07.23 | dlynes_laptop | evilbit: yep |
| 16:07.27 | dlynes_laptop | evilbit: VoicemailMain() |
| 16:07.29 | evilbit | wow! thanks :-) |
| 16:07.37 | dlynes_laptop | evilbit: it allows you to check your voicemail and dial out |
| 16:07.47 | dlynes_laptop | evilbit: sorta like disa inside of it |
| 16:07.52 | evilbit | that's perfect. |
| 16:07.53 | dlynes_laptop | evilbit: and what file said, too |
| 16:07.58 | evilbit | hehe |
| 16:08.21 | dlynes_laptop | evilbit: it's just an option you can add to each voicemail box |
| 16:08.32 | *** join/#asterisk trbldwine (i=troubled@71.194.161.170) |
| 16:08.33 | dlynes_laptop | evilbit: i wouldn't suggest using it as a global option, however |
| 16:09.03 | evilbit | dlynes_laptop: yeah, I'd like to use it on a very select basis |
| 16:09.03 | parag_ast | dlynes_laptop, if i m renaming the file all-circuits-busy-now.gsm to all-circuits-busy-now1.gsm ...will it help me in not playing this sound at all ?? |
| 16:09.25 | dlynes_laptop | parag_ast: that's not a sound...it's a voice recording |
| 16:09.34 | Dr-Linux | dlynes_laptop: thoughts? |
| 16:09.36 | parag_ast | yes will it help me |
| 16:09.57 | dlynes_laptop | parag_ast: oh wait |
| 16:10.08 | dlynes_laptop | parag_ast: forgot...you're using asterisk@home |
| 16:10.14 | parag_ast | es |
| 16:10.15 | parag_ast | yes |
| 16:10.16 | dlynes_laptop | parag_ast: maybe it plays it automatically on you |
| 16:10.21 | dlynes_laptop | parag_ast: i have no idea |
| 16:10.24 | parag_ast | yes |
| 16:10.27 | dlynes_laptop | parag_ast: ask on #trixbox |
| 16:10.36 | parag_ast | join #trixbox |
| 16:10.42 | dlynes_laptop | parag_ast: they'll probably be able to give you a definitive answer |
| 16:11.08 | *** join/#asterisk exBobster (n=Miranda@staticline2218.toya.net.pl) |
| 16:11.42 | *** part/#asterisk trbldwine (i=troubled@71.194.161.170) |
| 16:11.54 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 16:12.23 | exBobster | hi, i'm trying to install asterisk 1.2.10 on debian sarge, but still get 'segmentation fault' errors when compiling. can anybody help me solve this problem? |
| 16:13.22 | dlynes_laptop | exBobster: do you have an unstable glibc or unstable gcc installed? |
| 16:13.55 | exBobster | no, i don't |
| 16:14.29 | exBobster | is this safe to install unstable packages on stable-distro-box? |
| 16:15.25 | exBobster | i had many dependency problems in the past, after doing such thing |
| 16:15.44 | *** join/#asterisk MacWeenie (n=chatzill@cpe-74-65-228-134.nyc.res.rr.com) |
| 16:16.30 | *** join/#asterisk droops (n=root@adsl-065-005-212-128.sip.jan.bellsouth.net) |
| 16:16.32 | dlynes_laptop | exBobster: no idea....#debian would be able to help you out with that question |
| 16:16.43 | dlynes_laptop | exBobster: last time I used debian was back before v1.0 |
| 16:16.57 | exBobster | ;) |
| 16:17.51 | *** join/#asterisk |dennis| (n=dennis@200.32.215.82) |
| 16:18.20 | *** join/#asterisk bkw__ (n=brian@asterisk/friend-and-developer/bkw) |
| 16:18.20 | exBobster | dlynes_laptop: anyway, thanks for an answer :) |
| 16:18.52 | dlynes_laptop | parag_ast: Now, after getting phillipel's explanation, do you understand why we're not able to help in here with a@h questions? |
| 16:19.10 | dlynes_laptop | parag_ast: a@h setup is quite a bit different from vanilla asterisk |
| 16:19.34 | parag_ast | understood sir |
| 16:20.13 | Dr-Linux | :S |
| 16:20.23 | dlynes_laptop | I had no idea a@h provided default recordings for all the sip/iax error codes, myself...after the way you were explaining it though, I started to have my suspicions...that's why i joined #freepbx to see if my suspicions were correct |
| 16:20.59 | parag_ast | ha ha ha |
| 16:21.08 | dlynes_laptop | anyways |
| 16:21.09 | Dr-Linux | he he he |
| 16:21.15 | dlynes_laptop | phillipel's got another response for you |
| 16:23.15 | *** join/#asterisk ronaldgibson (n=ronaldgi@adsl-66-218-62-5.dslextreme.com) |
| 16:27.13 | dlynes_laptop | omg |
| 16:27.28 | dlynes_laptop | someone's planning to use freepbx in a reseller environment? |
| 16:27.29 | dlynes_laptop | scary |
| 16:27.57 | file | dlynes_laptop: people do dit |
| 16:28.01 | file | erm do it |
| 16:28.30 | dlynes_laptop | file: yeah...like i said...scary thought |
| 16:30.01 | dlynes_laptop | Just doesn't seem terribly maintainable |
| 16:32.08 | *** join/#asterisk viperdude (n=viperdud@195.74.96.120) |
| 16:33.03 | evilbit | so, I'm having some connection problems with my provider, teliax, and I can't tell if it's on my end or theirs... Many times I will try and dial my DID, and it will run several times and then I get a busy signal. When this happens I don't see any packets (via tcpdump) coming from them to me. Other times, I dial my DID and it will go thru and I'll hear my voiceprompts... If I use teliax and set IAX plus a follow-me timeout for 10secs t |
| 16:33.03 | evilbit | hen my follow-me number gets called after the timeout... anyone have any pointers? |
| 16:33.04 | *** join/#asterisk vaddineni (n=vaddinen@cpe-72-181-71-206.houston.res.rr.com) |
| 16:33.37 | file | evilbit: behind NAT? |
| 16:34.06 | queuetue | freepbx is the same as AAH? |
| 16:34.50 | file | queuetue: AMP became FreePBX and Asterisk@Home became Trixbox, I *think* |
| 16:34.51 | dlynes_laptop | no |
| 16:34.57 | dlynes_laptop | file: correct |
| 16:35.32 | queuetue | Ah, I didn't realize there was a difference between AMP and AAH. |
| 16:35.39 | dlynes_laptop | aah uses amp |
| 16:35.42 | file | and then Asterisk@Home/Trixbox uses AMP/FreePBX for web management so you have to go to them for support if you're doing that |
| 16:35.45 | AndyCap | what was the name of the asterisk fork? |
| 16:36.02 | AndyCap | ah, that was open, not free |
| 16:36.03 | dlynes_laptop | aah is a linux distribution with amp, flash operator panel and a bunch of other stuff preinstalled |
| 16:36.17 | dlynes_laptop | AndyCap: OpenPBX |
| 16:36.38 | dlynes_laptop | AndyCap: /join #openpbx |
| 16:36.53 | file | don't forget the TelcomjoshvoxmartPBX |
| 16:37.14 | dlynes_laptop | file: that's something josh just made up, right? |
| 16:37.28 | queuetue | What is "truly" open source supposed to imply? :) |
| 16:37.40 | file | I've forgotten where exactly, but I think a Matt in support came up with part of it |
| 16:37.43 | AndyCap | dlynes_laptop: yes, I found it. had that and freepbx mixed up |
| 16:37.55 | file | so I use it every chance I can now |
| 16:37.55 | dlynes_laptop | queuetue: it means the source is freely available |
| 16:38.22 | evilbit | file: no, route-able ip addr |
| 16:38.30 | AndyCap | queuetue: Hehe, they made it up, so I guess they have an answer to it as well. |
| 16:38.35 | file | evilbit: so that eliminates one thing... how odd |
| 16:38.36 | dlynes_laptop | queuetue: well, not necessarily freely available, but that the full source code is obtainable upon request, without extra charge |
| 16:38.49 | dlynes_laptop | queuetue: or sometimes it's included with the binary |
| 16:38.59 | dlynes_laptop | queuetue: it doesn't necessarily mean it's free as in beer, however |
| 16:39.01 | queuetue | I know what open source means (I'm the OSS laiason for the Free State Project. ). I was asking why they referred to it as "truly" open. :) |
| 16:39.17 | dlynes_laptop | ah |
| 16:39.33 | dlynes_laptop | queuetue: openpbx? |
| 16:39.37 | queuetue | It sounds like a dig at asterisk for some perceived slight. |
| 16:39.43 | queuetue | dlynes_laptop: Yeah. |
| 16:40.14 | Nugget | It's in reference to Diguim's requirement that all code contributors allow Digium to dual-license asterisk |
| 16:40.29 | dlynes_laptop | queuetue: because the guys that started openpbx have a problem with Marc's licensing model and code release waivers |
| 16:40.41 | dlynes_laptop | erm mark's |
| 16:41.15 | *** join/#asterisk russellb_ (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 16:41.15 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 16:41.16 | dlynes_laptop | It's kinda silly, I think |
| 16:41.29 | Nugget | In order to contribute code to asterisk you either have to release it as public domain or BSD, or you have to sign a waiver allowing Digium to release that code under non-GPL licenses. |
| 16:41.35 | dlynes_laptop | Asterisk isn't the only project with dual licensing, and I'm sure it's not the last |
| 16:41.41 | queuetue | Most forks are. Non-silly forks pretty much take over instantly. |
| 16:41.49 | dlynes_laptop | but they seem to have targetted digium for some reason |
| 16:42.11 | AndyCap | queuetue: of course it is. that's why I said they probably had the answer. |
| 16:42.25 | queuetue | Does Digium return that right? Contributors can sell non-free versions? |
| 16:42.28 | Nugget | no |
| 16:42.35 | queuetue | Ah, I see the problem now. |
| 16:42.43 | dlynes_laptop | Personally, I think freeswitch is a much more mature response to their issue with the digium licensing, than openpbx |
| 16:43.07 | file | dlynes_laptop: freeswitch is targetting a different sector though, softswitch |
| 16:43.17 | dlynes_laptop | file: nah...they have a pbx module now, too |
| 16:43.24 | queuetue | Its a good reason for a fork, I guess, but you would need more than one or two devs who really care. And vitriol isn't a good way to recruit. |
| 16:43.56 | AndyCap | still since the fork is gpl, they're still not selling non-free version. |
| 16:44.08 | queuetue | AndyCap: Good point. :) |
| 16:44.09 | file | dlynes_laptop: it was originally designed to be a softswitch though, where Asterisk wasn't - it was for someone who didn't want to pay for a PBX |
| 16:44.09 | bkw__ | Also since freeswitch is MPL you're free to exploit it commercially without having the dual license issue.. or any patent issues with say g.729 |
| 16:44.24 | Nugget | "exploit" is a charged word. |
| 16:44.31 | queuetue | AndyCap: So they gain nothing except being able to not help the people who started the project in the first place. |
| 16:44.45 | bkw__ | well they are free to build commercial applications on it without the requirement to give back changes to their code they wrote. |
| 16:44.57 | bkw__ | Whats mine is mine and whats yours is yours... |
| 16:45.06 | bkw__ | so you're free to build binary modules and sell them for freeswitch |
| 16:45.12 | queuetue | Sounds pretty "open" to me. :) |
| 16:45.22 | queuetue | Oh, wait, no - the other thing. :) |
| 16:45.27 | file | lol |
| 16:45.38 | bkw__ | once we stamp the version number the API is frozen for that release. |
| 16:45.46 | bkw__ | we are not allowed to change the API |
| 16:45.48 | dlynes_laptop | bkw__: but regardless of whether your base is freeswitch or asterisk, it's still in your best interest (usually) to offer your changes back to the community |
| 16:46.00 | bkw__ | dlynes_laptop, right but sometimes you can't |
| 16:46.05 | bkw__ | like if you use a closed PRI stack |
| 16:46.08 | bkw__ | or a closed sip stack |
| 16:46.09 | dlynes_laptop | bkw__: because then you don't have to keep patching the code every time a new version comes out |
| 16:46.12 | bkw__ | the MPL allows this |
| 16:46.23 | bkw__ | it gives you more options as an end user |
| 16:46.24 | *** part/#asterisk parag_ast (n=root@dxb-b110696.alshamil.net.ae) |
| 16:46.33 | dlynes_laptop | bkw__: like i said....'usually' |
| 16:46.38 | bkw__ | right |
| 16:47.07 | queuetue | It also gives you more options as a developer ... mostly those that let you remove options from the end user. :) |
| 16:47.08 | bkw__ | we had a great conference too |
| 16:47.25 | bkw__ | I learned how to build synthetic voices for Festival |
| 16:47.33 | bkw__ | and how to make acoustic models for sphinx2 |
| 16:47.44 | bkw__ | I already built a talking clock using my voice :P |
| 16:48.01 | file | that's creepy, sure you want that? |
| 16:48.14 | bkw__ | next I have to record 1132 sentences from the CMU Arctic DB for a full clunits voice |
| 16:48.24 | dlynes_laptop | bkw__: you mean ummmmmm....cluecon? |
| 16:48.29 | Nugget | bkw is the new allison. |
| 16:48.36 | bkw__ | dlynes_laptop, yes |
| 16:48.40 | dlynes_laptop | ah |
| 16:48.42 | bkw__ | also i fixed speech_tools for the Mac while I was there |
| 16:48.56 | bkw__ | Kevin lenzo and I spent like 5 or 6 hours talking and he showed me much of what I needed to know |
| 16:49.08 | Nugget | lenzo! the author of infobot! ;) |
| 16:49.10 | dlynes_laptop | yeah...i wish i had free time so i could attend things like that |
| 16:49.12 | bkw__ | yes |
| 16:49.18 | bkw__ | and CTO of Cepstral |
| 16:49.19 | dlynes_laptop | unfortunately, i've got negative free time |
| 16:49.45 | *** join/#asterisk javar (n=javar@200.118.174.253) |
| 16:49.48 | dlynes_laptop | anyways...gotta run |
| 16:49.52 | AndyCap | bkw__: so how many credits would it take for a alison festival voice. :P |
| 16:49.52 | bkw__ | I'm going to be building a personal digital assistant on the phone using my voice.. sphinx2 and interface with my addresbook and ical on the mac. |
| 16:49.53 | dlynes_laptop | talk to you all later |
| 16:50.07 | *** join/#asterisk vaddineni (n=vaddinen@cpe-72-181-71-206.houston.res.rr.com) |
| 16:50.16 | bkw__ | AndyCap, first you have to ask permission. Then get the Arctic Database from CMU and have them all recorded. |
| 16:50.27 | bkw__ | http://festvox.org/cmu_arctic/cmuarctic.data |
| 16:50.33 | bkw__ | its 1132 sentences |
| 16:50.46 | bkw__ | it takes months to hand correct the voice |
| 16:51.04 | bkw__ | automation can get you started but you'll still have to go back and fix pitchmarks and cut points |
| 16:51.47 | bkw__ | and the voice you build out of the arctic database will sound great if you tweak it. |
| 16:51.55 | bkw__ | almost flawless |
| 16:52.11 | AndyCap | :) not bad. |
| 16:52.12 | bkw__ | the voices that are used with festival are diphone or uniphone voices.. which don't give you much coverage |
| 16:52.21 | bkw__ | if you build a full voice it can sound great |
| 16:52.41 | bkw__ | it will take me 7-8 hours to record all this stuff |
| 16:53.14 | bkw__ | but then I never have to go back.. and I can really mess with people on the phone :P |
| 16:53.47 | AndyCap | Hmm, maybe one should take up speechwriting. :-P |
| 16:53.59 | bkw__ | voice building |
| 16:54.24 | AndyCap | bkw__: no, that comes later, first one has to get the head of state to say what you need. |
| 16:54.39 | bkw__ | the acoustic model for sphinx will take me about 50-60 people reading the arctic database to get a good sample |
| 16:54.52 | bkw__ | but those need to be read over the phone :P |
| 16:54.57 | bkw__ | with various background noise |
| 16:55.05 | bkw__ | then I can release the 500-600 meg database to the public |
| 16:55.13 | bkw__ | its a HUGE endeavor |
| 16:56.37 | bkw__ | "The Virtual bkw_(tm)" |
| 16:56.38 | bkw__ | haha |
| 16:58.46 | file | geez |
| 16:59.32 | file | all I want to do is apply this credit I have to my order... is that so hard? |
| 17:00.46 | file | "I'm sorry sir, we can not do that." |
| 17:00.50 | file | "Yes you can. Do it." |
| 17:02.13 | file | "Okay Joshua, I've got it here. It's all taken care of." |
| 17:02.14 | *** join/#asterisk Soul (n=Soul@195-23-161-68.net.novis.pt) |
| 17:02.18 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.239.225.Dial1.SanJose1.Level3.net) |
| 17:03.06 | E-bola | :) |
| 17:03.08 | file | that call went from Toronto, to India, back to Toronto, and then to Dieppe... |
| 17:03.24 | E-bola | i hate all asian phone ppl on support lines |
| 17:03.38 | E-bola | they always want to know 117 things about you and ur stuff before u get any help |
| 17:03.42 | evilbit | that's great |
| 17:04.12 | file | E-bola: okay, I need to know your exact coordinates plus what time it is oh and what channels you subscribe to on your TV before I can help you sir |
| 17:04.34 | E-bola | dont forget about describing all ur shit in details for many minutes |
| 17:04.42 | E-bola | to then get transfered to a guy who cant see what u just told another guy |
| 17:04.45 | E-bola | thats the worst :) |
| 17:04.57 | E-bola | or when support lines make u enter a client or phone number to know who u are in advance |
| 17:05.04 | E-bola | and then when u get connected they ask u anyway lol |
| 17:05.19 | bkw__ | ya that pisses me off |
| 17:05.24 | file | how about entering your date of birth? (but they don't tell you how to enter it, and then when you enter it wrong it complains) |
| 17:05.26 | bkw__ | why did they ask if they were going to ask again |
| 17:05.42 | *** join/#asterisk mafkees (n=michiel@vanbaak.xs4all.nl) |
| 17:05.45 | mafkees | hello all |
| 17:05.48 | E-bola | ohh well the world isnt perfect :) |
| 17:06.00 | mafkees | I'm trying to get the RFC ENUM macro working |
| 17:06.08 | E-bola | I think it sux that xlite works better than a 100$ linksys spa922 btw |
| 17:06.08 | E-bola | :) |
| 17:06.10 | mafkees | but for some reason I cant get it to run |
| 17:06.13 | mafkees | <PROTECTED> |
| 17:06.13 | E-bola | Xlite is free |
| 17:06.13 | SantaRosaMark | all the tech support ???s can be answered here: http://www.deadtroll.com/index2.html?/video/helldeskcable.html~content |
| 17:06.23 | E-bola | and it doesnt give me any echo, while the linksys do give me a bad echo |
| 17:06.27 | mafkees | that line is always going to the failed label |
| 17:06.38 | mafkees | no matter what I put in E164NETWORKS |
| 17:07.40 | mafkees | I replaced the line by a Verbose(1,len is ${LEN(${E164NETWORKS})} for content ${E164NETWORKS}) |
| 17:07.45 | mafkees | and that works fine |
| 17:07.51 | mafkees | of course it comes in an endless loop |
| 17:08.01 | mafkees | because it's not doing any checking anymore |
| 17:08.28 | mafkees | help please |
| 17:08.57 | file | today is getting weird |
| 17:09.12 | mafkees | file: gheh, why ? |
| 17:09.34 | file | the radio station seems to be doing a gay call-in thing |
| 17:09.48 | file | "yeah I totally made out with this guy named Shaun" |
| 17:10.08 | E-bola | change the station |
| 17:10.08 | E-bola | hehe |
| 17:11.14 | file | so how is everyone doing on this beautiful day? |
| 17:11.21 | mafkees | file: doing ok |
| 17:11.30 | mafkees | trying to get the enum macro to work |
| 17:11.39 | mafkees | and you ? |
| 17:12.02 | file | E-bola: easy, pretend the product doesn't exist |
| 17:12.33 | E-bola | file: hard when we bought 4 of them |
| 17:12.33 | mafkees | file: care to help me a little ? |
| 17:12.33 | E-bola | hehe |
| 17:13.58 | mafkees | if I do: exten => s,n,Verbose(1, len is ${LEN(${E164NETWORKS)}) |
| 17:14.02 | mafkees | it outputs: 18 |
| 17:14.23 | mafkees | if I do: exten => s,n,GotoIf($["${LEN(${E164NETWORKS})}" < "2"]?failed) |
| 17:14.33 | mafkees | it's ALWAYS going to 'failed' |
| 17:15.19 | file | erm? |
| 17:15.37 | file | exten => s,n,GotoIf($[${LEN(${E164NETWORKS})} < 2]?failed) |
| 17:16.14 | mafkees | 6. GotoIf($[${LEN(${E164NETWORKS})} < 2]?failed) |
| 17:16.19 | file | might need to space it out a bit |
| 17:16.32 | mafkees | I now have that |
| 17:16.34 | mafkees | same result |
| 17:16.34 | file | it can be picky |
| 17:17.12 | file | what do you have in your dialplan right now? |
| 17:17.30 | mafkees | <PROTECTED> |
| 17:17.57 | *** join/#asterisk alexcf (i=alexcf@lexi.me.uk) |
| 17:18.00 | file | I mean in the .conf |
| 17:18.07 | alexcf | hey gusy |
| 17:18.09 | alexcf | guys too |
| 17:18.17 | mafkees | exten => s,n(begin),GotoIf($[${LEN(${E164NETWORKS})} < 2]?failed) |
| 17:18.31 | *** join/#asterisk _DAW (n=bob@adsl-222-55-211.msy.bellsouth.net) |
| 17:19.22 | mafkees | I copied the macro from here |
| 17:19.22 | file | what's the console output look like for that line when you use it? |
| 17:19.24 | mafkees | http://www.voip-info.org/wiki/view/RFC+Compliant+ENUM+Macro |
| 17:19.45 | alexcf | currently, my voip provider reads my callerid using "From: <sip:xxx@yyy.com>" (the xxx), but the only place i can see the set the "fromuser" is in the sip.conf |
| 17:19.53 | mafkees | none, because it jumps to the failed label |
| 17:19.53 | alexcf | is there any way i can set that in the extensions.conf? |
| 17:19.57 | mafkees | that is working fine |
| 17:20.07 | file | mafkees: no, you're not understanding what I'm asking |
| 17:20.07 | alexcf | (i tried using setcallerid()) |
| 17:20.20 | mafkees | sorry |
| 17:20.30 | file | mafkees: the gotoif line will be shown on the Asterisk console |
| 17:20.34 | mafkees | ah |
| 17:20.37 | mafkees | in verbose mode |
| 17:20.39 | mafkees | hang on |
| 17:20.44 | file | alexcf: fromuser overrides callerid set via that method |
| 17:20.54 | file | alexcf: if fromuser isn't set, then you can put the callerid into the From header |
| 17:20.59 | alexcf | ok |
| 17:21.03 | mafkees | <PROTECTED> |
| 17:21.25 | alexcf | file: ill give it a go now |
| 17:21.26 | alexcf | ta |
| 17:22.37 | mafkees | <PROTECTED> |
| 17:22.39 | mafkees | there |
| 17:22.40 | mafkees | I have it |
| 17:22.50 | mafkees | ah man |
| 17:22.51 | mafkees | hit me |
| 17:22.54 | file | what was it? |
| 17:23.02 | mafkees | with the biggest axe you have |
| 17:23.14 | mafkees | a ; |
| 17:23.17 | mafkees | before the dial |
| 17:23.20 | mafkees | DUH |
| 17:23.31 | file | moving swiftly on |
| 17:23.38 | mafkees | sorry |
| 17:25.27 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust944.leed.cable.ntl.com) |
| 17:25.41 | alexcf | file: ok, removing the fromuser, and just using user=, the provider is still using my username as the caller id |
| 17:25.44 | alexcf | :| |
| 17:26.22 | file | alexcf: need to see a sip show peer, sip.conf entry, plus sip debug |
| 17:26.27 | alexcf | k |
| 17:26.52 | file | you might also try setting sendrpid=yes |
| 17:26.55 | file | and seeing if they support RPID |
| 17:26.57 | *** join/#asterisk Qwell_ (n=north@unaffiliated/qwell) |
| 17:26.57 | *** mode/#asterisk [+o Qwell_] by ChanServ |
| 17:26.58 | alexcf | k |
| 17:27.01 | alexcf | ill give that a go first, sec |
| 17:27.20 | file | why look who it is... |
| 17:27.27 | Qwell | meh |
| 17:27.32 | file | Qwell: ITS' YOU |
| 17:27.32 | Qwell | I think my router is hosed |
| 17:27.43 | file | shopping spree time! |
| 17:27.47 | Qwell | eeps |
| 17:28.56 | alexcf | file: prefer the spam here? (i've pm'd it to you already) |
| 17:29.17 | file | erm? I ignore pms from random people, since they end up pasting full configs and stuff |
| 17:29.26 | file | throw it up on a pastebin |
| 17:29.26 | alexcf | heh np |
| 17:29.28 | alexcf | ok |
| 17:29.42 | file | full configs over IRC make me sad :( |
| 17:29.48 | alexcf | tell me about it |
| 17:29.50 | file | you heard me, sad! |
| 17:33.46 | alexcf | hmm |
| 17:33.50 | alexcf | file: pastebin seems b0rked |
| 17:33.54 | alexcf | i've uploaded it |
| 17:33.56 | alexcf | http://cruz.co.uk/~alexcf/asterisk.txt |
| 17:35.06 | file | 02079931703 is the callerid? |
| 17:35.23 | *** join/#asterisk logicwrath (n=dfgaeefa@c-68-60-121-112.hsd1.mi.comcast.net) |
| 17:35.44 | alexcf | yea |
| 17:35.48 | alexcf | hit refresh |
| 17:35.55 | file | it's being sent. |
| 17:36.01 | *** join/#asterisk lin00bies (n=jsharryp@210.213.199.191) |
| 17:36.03 | alexcf | mmm |
| 17:36.09 | alexcf | callerid comes up at 0<username> |
| 17:36.37 | file | they could be changing it |
| 17:36.47 | alexcf | mhmm |
| 17:36.47 | alexcf | ok |
| 17:37.49 | *** join/#asterisk topping (n=topping@207.47.6.192.static.nextweb.net) |
| 17:37.59 | alexcf | thanks file, ill go beat them with a stick |
| 17:38.09 | file | ok |
| 17:38.56 | *** join/#asterisk saftsack (n=saftsack@p54A7C8E6.dip.t-dialin.net) |
| 17:41.39 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 17:42.55 | *** join/#asterisk ESCulapio__ (n=ESCulapi@188stb68.codetel.net.do) |
| 17:50.14 | *** join/#asterisk sumasuma (n=chumma@cm222.omega183.maxonline.com.sg) |
| 17:50.26 | sumasuma | How can i check the errorlog of AGI ?? |
| 17:52.42 | *** join/#asterisk Skinzy (i=tom@217.151.104.169) |
| 17:53.39 | mafkees | on the cli |
| 17:53.42 | mafkees | agi debug |
| 17:54.57 | sumasuma | mafkees: thanks |
| 17:55.58 | *** join/#asterisk ThePlot (i=ThePlot@202.164.38.210) |
| 17:57.45 | ThePlot | Anyone aware of any registration problems with Polycom phones and Asterisk 1.2.10? I have a hard time trying to get them to register, tried everything like removing passwords, configurations.. the debug info just says that it fails to register.. any way to turn on more verbose output that shows username coming in and the value with it is being compared? |
| 17:58.51 | ThePlot | I did set in the address field to match the username too |
| 17:59.15 | *** join/#asterisk mcreedjr (n=mcreedjr@68.72.158.2) |
| 18:01.13 | mcreedjr | Hey guys and gals. When a SIP phone on my network issues a 302 Redirect with when rang, I want it to grab a differnet outside line. Anyone have an idea how to do this? |
| 18:01.57 | mcreedjr | Should I just add a special pattern in my dialout context to treat the call differently? Meaning, when a user redirects his/her phone, should they just prefix the redirected number with say 65... |
| 18:08.22 | *** join/#asterisk |dennis| (n=dennis@200.32.215.82) |
| 18:10.39 | *** join/#asterisk spackle (n=spackle@ip207-199-243-35.static.ishsi.com) |
| 18:14.01 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
| 18:15.38 | my007ms | how can i use exten => s,3,Dial |
| 18:15.41 | my007ms | to call many |
| 18:15.42 | *** join/#asterisk Tili (n=tili@202.133.67.218) |
| 18:16.01 | jart | many? |
| 18:16.29 | jart | exten => s,1,Dial(Zap/1&SIP/somone&Zap/2) ??? |
| 18:18.01 | *** join/#asterisk tempest1 (n=asf@adsl-230-225-133.chs.bellsouth.net) |
| 18:18.07 | Skinzy | Anyone recomend any tutorial / info sites except voip-info? |
| 18:18.34 | jart | for? |
| 18:18.42 | jart | general * usage? |
| 18:19.06 | Skinzy | Yeah, i'm trying to setup a * server for sip usage, to connect to two different voip providers (sipgate and voiptalk.org) |
| 18:19.28 | jart | well, you may want to try the asterisk documentation project |
| 18:19.31 | jart | or the starfish book |
| 18:19.46 | Skinzy | Aight, thanks |
| 18:19.50 | jart | * isn't known for being extensively documented |
| 18:20.02 | Skinzy | Yeah, i noticed |
| 18:20.35 | jart | http://www.oreilly.com/catalog/asterisk/ |
| 18:20.43 | Skinzy | thanks |
| 18:20.46 | mafkees | http://www.oreilly.com/catalog/asterisk/ |
| 18:20.48 | mafkees | ah |
| 18:20.49 | mafkees | lol |
| 18:20.49 | *** join/#asterisk blainett (n=kvirc@87.217.91.184) |
| 18:21.04 | mcreedjr | Skinzy: I highly recommend that O;Reilly book... |
| 18:21.14 | mcreedjr | Skinzy: all around good stuff |
| 18:21.26 | Skinzy | cool |
| 18:22.02 | jart | i had the pleasure of hanging out with the guy who wrote it at clue con, he's pretty cool |
| 18:22.18 | *** join/#asterisk spackle (n=spackle@ip207-199-243-35.static.ishsi.com) |
| 18:22.18 | mcreedjr | jart: i liked the way it was written - the little jokes and whatnot |
| 18:22.28 | mcreedjr | jart: easy, yet informative read... |
| 18:22.45 | jart | mcreedjr: i agree |
| 18:22.48 | Assid | ~thebook |
| 18:22.50 | jbot | thebook is probably a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 18:23.03 | mcreedjr | Good man Assid |
| 18:25.08 | russellb_ | my copy is signed by all the authors and mark :-p |
| 18:25.32 | Assid | hrmm.. i want a digitally signed copy |
| 18:25.37 | file | russellb_: ooh |
| 18:26.18 | russellb_ | and then i signed their own copies |
| 18:26.21 | russellb_ | and fun times were had by all |
| 18:26.37 | file | my internet is being sad |
| 18:26.42 | file | it no work! |
| 18:26.44 | russellb | file: mine too |
| 18:27.21 | Assid | sounds like my connection |
| 18:27.34 | Assid | hrmm whats a solo processor ? |
| 18:27.46 | Assid | Intel® Core Solo Proc |
| 18:27.59 | russellb | the internet needs bigger tubes |
| 18:28.22 | spackle | marketspeak |
| 18:28.41 | russellb | as in, *not* core duo |
| 18:28.52 | file | spackle: !!! |
| 18:29.00 | Assid | so its still a centrino? |
| 18:29.19 | spackle | file !!!! |
| 18:30.58 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 18:30.59 | *** join/#asterisk ManxPower (n=ewieling@71-8-11-111.dhcp.leds.al.charter.com) |
| 18:31.33 | JunK-Y | ohh no, its jart ! |
| 18:31.40 | jart | hi there junky |
| 18:31.45 | JunK-Y | whats up? |
| 18:31.57 | jart | just trying to get ekiga to stop being a pain |
| 18:32.16 | JunK-Y | just dont use it ? :P |
| 18:32.25 | *** join/#asterisk bkw__ (n=brian@asterisk/friend-and-developer/bkw) |
| 18:32.45 | *** join/#asterisk Keysernz (n=kaiser@pinkerton98.plus.com) |
| 18:32.54 | jart | well, i kind of don't feel like calling the pbx with my cell phone every time i test this app |
| 18:33.06 | jart | or setting up a sip phone |
| 18:33.34 | JunK-Y | im mainly use xten for that kind of needs. |
| 18:33.57 | jart | i'm on linux, xten isn't an option for me |
| 18:34.08 | JunK-Y | xten works on ubuntu too. |
| 18:34.28 | jart | not for me, besides i think my problems are nat related |
| 18:35.53 | Assid | hrmm anyone know any nice laptops on sale ? toshiba/dell .. ? |
| 18:36.06 | jart | i recommend teh 500 dollar hp laptop on newegg |
| 18:38.11 | Mike | hey guys |
| 18:38.20 | Mike | i think call-limit doesnt work |
| 18:38.21 | Qwell | jart: link? |
| 18:38.35 | Assid | whats better.. turion or mobile athlon? |
| 18:38.42 | Qwell | turion I thought |
| 18:40.12 | tsurk0 | hello everybody |
| 18:40.23 | jart | Qwell: i think they're out of stock now :( |
| 18:40.39 | file | Qwell: you need to get more t-shirts |
| 18:40.43 | jart | Qwell: but it's an HP Compaq nx6110, if you upgrade teh ram it's awesome |
| 18:40.46 | file | Qwell: and then... give them to me |
| 18:40.51 | Qwell | I wanna get that acer barebones laptop |
| 18:40.53 | Qwell | file: ...yeah... |
| 18:41.06 | file | I like the color orange! |
| 18:41.08 | file | okay?!? |
| 18:41.09 | Qwell | :D |
| 18:41.15 | tsurk0 | i have make asterisk place calls to a local mobile operator, but i've no ide what kind of hardware i need |
| 18:41.19 | tsurk0 | could you help me? |
| 18:41.24 | *** join/#asterisk Frogdude (n=FroggerD@c-24-16-72-159.hsd1.mn.comcast.net) |
| 18:41.25 | Dr-Linux | Qwell: hey :) |
| 18:41.43 | Assid | would AMD be better for laptops or intels? |
| 18:41.45 | Dr-Linux | Qwell: they sent me third email :) |
| 18:41.59 | Qwell | Assid: If you like battery life, go with a turion |
| 18:42.20 | Assid | know a decent one ? |
| 18:42.31 | *** join/#asterisk fnordus (n=dnall@s142-179-111-243.bc.hsia.telus.net) |
| 18:42.38 | spackle | what is the standing guidance for analog echo cancellation these days? |
| 18:42.40 | Qwell | the barebones acer.. |
| 18:42.44 | Assid | a friend of mine is moving to NY, and she needs a laptop.. i said i'd help her find a decent one |
| 18:42.46 | Qwell | or was is asus? whichever |
| 18:42.56 | Qwell | sec |
| 18:43.25 | Qwell | asus |
| 18:43.43 | Qwell | there are 3. one was the a6u |
| 18:43.58 | Qwell | Z92Km was it, I think |
| 18:44.29 | Assid | dont see it in newegg |
| 18:45.06 | Qwell | they don't carry it |
| 18:45.20 | Assid | hrmm.. hows this: http://www.newegg.com/Product/Product.asp?Item=N82E16834115257 ? |
| 18:45.22 | Qwell | find a place that sells it, where they also customize it |
| 18:45.47 | Qwell | that's not too shabby |
| 18:45.49 | mcreedjr | <PROTECTED> |
| 18:45.57 | *** join/#asterisk russellb_ (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 18:45.57 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 18:46.34 | *** join/#asterisk tempest1 (n=asf@adsl-153-56-119.chs.bellsouth.net) |
| 18:47.09 | Assid | trust me.. gotta make it as easy as possible for her to get it.. dont think shes gonna know where to go and do whatever that needs to be done |
| 18:47.27 | Qwell | get a mac |
| 18:47.39 | Assid | hahaa.. no way... |
| 18:47.52 | russellb_ | yes way~ |
| 18:48.12 | Assid | i think im gonna send her these urls.. ask her to choose whichever one she wants |
| 18:48.28 | Assid | theres no way shes gonna learn to use a mac.. |
| 18:48.47 | spackle | yeah, they are pretty complicated |
| 18:48.51 | *** part/#asterisk budmang (n=budman@12.206.134.162) |
| 18:49.02 | mcreedjr | lol spackle |
| 18:49.05 | mcreedjr | sarcasm? :) |
| 18:49.30 | mcreedjr | powerbooks don't have an IPOD scroll wheel though... |
| 18:49.36 | mcreedjr | that greatly simplifies things |
| 18:49.37 | mcreedjr | heh |
| 18:49.46 | Assid | i want a scroll wheel! |
| 18:50.12 | spackle | mcreedjr yup. sorry, couldn't resist. |
| 18:50.14 | mcreedjr | who needs a keyboard when you've got a scroll wheel |
| 18:50.21 | mcreedjr | lol |
| 18:50.25 | Assid | just wrote a letter to these telephone guys to get them to launch this stupid unlimited plan here.. they got it in another city .. not here and im getting tired of my isp |
| 18:50.29 | mcreedjr | you can scroll those letters in! |
| 18:50.45 | spackle | I want a sip/puff tube so I don't have to move. |
| 19:00.14 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 19:04.09 | Nugget | Get a helper monkey like Homer Simpson did. |
| 19:16.38 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.215) |
| 19:22.42 | *** join/#asterisk TwoZero (n=twozero@2001:888:10:616:feed:feed:feed:feed) |
| 19:26.01 | *** join/#asterisk Chunks (n=none@ool-43536ea0.dyn.optonline.net) |
| 19:27.30 | *** join/#asterisk Un1x (i=Sean@CPE0016ec7e9cb3-CM0011ae8a7b0a.cpe.net.cable.rogers.com) |
| 19:30.00 | Chunks | I have a new install, new setup, and generally a newbie to *. I've googled/wiki'd but cannot figure out how to troubleshoot a SIP client unnable to authenticate with * |
| 19:30.23 | *** join/#asterisk smackus (n=ckwall@c-67-169-248-217.hsd1.ut.comcast.net) |
| 19:30.23 | Chunks | anyone have any suggestions? it's a simple authentication failure it seems. but the configs are incredibly simplistic. |
| 19:30.38 | jart | try turning on sip debug |
| 19:31.21 | smackus | does a tutorial for DUNDi setup exist? I have been trying to get my head around this, and I am not understanding it. |
| 19:31.50 | Chunks | I'm looking at that now. Sip phone makes a call, * asks for auth.... but I don't see any authentication coming from the phone, not that I'd know what that looks like, tbh |
| 19:32.39 | smackus | Chunks: what phone are you using? |
| 19:32.49 | Chunks | Cisco 7960 with 7.4 firmware |
| 19:34.58 | Chunks | I see a realm coming back |
| 19:35.28 | Chunks | but I don't remember configuring one. checking configs. not sure what a realm is :) |
| 19:37.32 | Chunks | if there's any relevant logs that might help trouble shoot... I can throw 'em on my site... if that'd help.. |
| 19:41.57 | Chunks | http://chunks.net/sip/forbidden.txt |
| 19:42.00 | ManxPower | Chunks, "sip debug" |
| 19:42.26 | Chunks | I'll get the whole thing.. |
| 19:42.41 | ManxPower | Chunks, do you have a [1001] section in sip.conf with a username=1001 in that section? |
| 19:43.07 | ManxPower | and a type=friend in that section as well? |
| 19:44.09 | Chunks | I'm using FreePBX so it's in sip_additional.conf, but yes, I have username=1001, type=friend |
| 19:44.23 | ManxPower | in the [1001] section? |
| 19:44.43 | Chunks | yes, there's only that 1 section. 1002 does not exist, I hadn't got that far yet :) |
| 19:44.51 | ManxPower | I can't help yu with FreePBX (see the /topic) because they create such bizarre config files. |
| 19:44.59 | Chunks | ahh |
| 19:45.29 | Chunks | hehe well, ya got out of that one :p :) no problem, I understand. I appreciate the help though. |
| 19:45.51 | ManxPower | "FreePBX/AMP/Asterisk@Home/Trixbox support in #freepbx" |
| 19:46.16 | Chunks | yea, I'm already on my way. I'm sorry for wasting your time |
| 19:46.21 | trelane | ManxPower, they aren't bizarre, they just think that everything requires a Macro |
| 19:46.55 | ManxPower | trelane, Yes, as I said, bizarre. |
| 19:47.12 | trelane | I consider that to be more "confused" than bizarre |
| 19:47.53 | ManxPower | For example, he put the [1001] section in sip_addtional.conf or whatever. Perhaps he used the GUI to configure it, in which case there could be a [1001] in sip.conf as well. |
| 19:48.26 | Chunks | it didn't, but I understand your point :) |
| 19:49.37 | *** join/#asterisk fourcheeze (n=rich@82.153.23.79) |
| 19:52.14 | Chunks | since I am just learning this stuff.. is there any gui you'd recommend? or should I just bite the bullet and try to learn CLI style? |
| 19:53.58 | spackle | bite the bullet(s) and get a good dentist! |
| 19:55.00 | Chunks | do you know any good freeware/opensource dentists? |
| 19:55.20 | fourcheeze | I'm contemplating using SRV records in DNS - can clients use these to choose a registrar - or is it only for making calls? |
| 19:57.29 | fourcheeze | e.g. if I have this in the DNS for mydomain.com: |
| 19:57.29 | fourcheeze | _sip._udp        SRV 0 1 5060 sip.otherdomain.com |
| 19:57.29 | fourcheeze | and I tell clients to register to mydomain.com, will they register there or with sip.otherdomain.com ? |
| 19:57.49 | fourcheeze | s/sip.otherdomain.com/sip.otherdomain.com. :-) |
| 20:08.32 | JunK-Y | these guys rocks: http://video.google.com/videoplay?docid=-5628733099857233320 |
| 20:11.28 | *** join/#asterisk [hC] (n=hardcore@190.10.9.191) |
| 20:13.28 | *** join/#asterisk SkunK (n=SkunK@oxford-bb-occam9-ws-25.dsl.maqs.net) |
| 20:14.18 | SkunK | can anyone offer soem assistance or poitn me in the direction of a howto and getting x-lite to dial into an asterisk box the x-lite phone registers on my box but i am unable to dial into the system with it |
| 20:24.16 | *** part/#asterisk smackus (n=ckwall@c-67-169-248-217.hsd1.ut.comcast.net) |
| 20:28.59 | *** join/#asterisk spackle[work] (n=spackle[@ip207-199-243-35.static.ishsi.com) |
| 20:30.38 | docelmo | SkunK here's a thought. Fundamentals of troubleshooting your own problem. If your going to come in here and ask just a lame question, you will never get an answer. Make sure when you dial it actually reaches the box. Your context is setup correctly with the right extensions and such. Make sure you have Xlite setup correctly.. |
| 20:31.17 | *** join/#asterisk Splas (n=jwb@brooklyn.paravolve.net) |
| 20:37.53 | *** join/#asterisk networkjedi (n=networkj@f3c121.gpcom.net) |
| 20:41.52 | *** join/#asterisk backblue (n=moo@87-196-102-213.net.novis.pt) |
| 20:44.22 | *** join/#asterisk ctooley (i=keefejoh@seraph.techwareit.com) |
| 20:44.37 | ctooley | Is there a way to make SayDigits interruptable? |
| 20:45.02 | ctooley | I need to be able to accept DTMF while playing the audio |
| 20:49.52 | JunK-Y | ctooley: use SAY DIGITS in an agi? |
| 20:50.39 | hads | ctooley: I saw somewhere a macro that would loop through the digits and background them all. Just been trying to find it. |
| 20:51.46 | ctooley | hads if you found that, that would be perfect |
| 20:55.47 | *** join/#asterisk Tili (n=tili@202.133.67.178) |
| 20:59.53 | *** join/#asterisk oomph (n=jimmy@ip68-98-166-99.dc.dc.cox.net) |
| 21:03.15 | hads | ctooley: That took a while: http://threebit.net/mail-archive/asterisk-users/msg00247.html |
| 21:03.31 | hads | You should be able to work that into a macro |
| 21:04.45 | *** join/#asterisk LoneShadow (n=duh@59.92.128.65) |
| 21:05.22 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 21:05.32 | LoneShadow | There are references about 1.4 being released in July 2006, any updates on when it is actually being released ? |
| 21:05.54 | hads | 'When it's ready' I think is the official word. |
| 21:06.11 | LoneShadow | thats no update, thats the golden rule :D |
| 21:06.39 | hads | I'm prettty sure that's the only answer you'll get from anyone at the moment :) |
| 21:07.07 | hads | (although I am just a luser) |
| 21:07.47 | LoneShadow | I saw this article about asterisk+jabber/googletalk http://www.voip-info.org/wiki/view/Asterisk+Speaks+with+Google+Talk |
| 21:07.54 | LoneShadow | would rather wait for 1.4 though |
| 21:15.40 | ctooley | hads, thank you |
| 21:16.01 | *** join/#asterisk test34 (n=test34@unaffiliated/test34) |
| 21:16.57 | LoneShadow | how do I fix my IAX settings, so that I dont get " Accepting UNAUTHENTICATED call from xxx.xxx.xxx.xxx" ? It ends up using [default] context on 2nd asterisk box ? |
| 21:17.48 | LoneShadow | I tried with and without qualify=yes, insecure=very, host=dynamic/xxx.dyndns.org |
| 21:20.42 | *** join/#asterisk RoyK (n=roy@ti211310a080-8252.bb.online.no) |
| 21:23.12 | wunderkin | wow the voipstreet guy needs to learn how to trim his posts ;D |
| 21:23.14 | dlynes_laptop | LoneShadow: context=somecontextyouhaventdefined |
| 21:24.12 | dlynes_laptop | hads: nah...that's the official word on #asterisk-dev, too |
| 21:24.43 | hads | lo dlynes_laptop :) |
| 21:24.49 | dlynes_laptop | afternoon |
| 21:24.54 | dlynes_laptop | too damned ho |
| 21:24.56 | dlynes_laptop | too damned hot |
| 21:25.08 | dlynes_laptop | I'll be so happy when this heat wave is over |
| 21:25.11 | dlynes_laptop | I'm starting to feel old |
| 21:25.28 | hads | Hah. It's 0925 and cold and raining here |
| 21:25.33 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 21:26.08 | hads | If you specify nonexistantcontext, won't it try then then fall back to default? |
| 21:26.38 | hads | I think you may need to specify existantcontext with a Hangup() in it. |
| 21:27.49 | dlynes_laptop | hads: yeah...good point |
| 21:28.46 | LoneShadow | I have the valid context mentioned |
| 21:28.58 | LoneShadow | its probably my broken trunk |
| 21:29.18 | dlynes_laptop | 25C here today |
| 21:29.19 | *** part/#asterisk bmonty (n=bmontgom@ubuntu/member/bmonty) |
| 21:29.24 | *** join/#asterisk docelm0 (n=Snake@55-65.126-70.tampabay.res.rr.com) |
| 21:29.29 | dlynes_laptop | too damned hot |
| 21:29.30 | LoneShadow | the 2nd asterisk box is a linksys router running openwrt (2.4 kernel, no ztdummy, no timing..) |
| 21:29.44 | hads | dlynes_laptop: That's not too bad. Besides I thought old people liked the heat ;) |
| 21:29.54 | dlynes_laptop | i'm not old |
| 21:29.57 | dlynes_laptop | i just feel old :) |
| 21:30.03 | hads | :) |
| 21:30.15 | dlynes_laptop | i'm only turning 37 this month |
| 21:30.16 | dlynes_laptop | sheesh |
| 21:30.30 | LoneShadow | what makes Asterisk to say it is an unauthenticated call ? |
| 21:30.30 | hads | Joke! Calm down ;) |
| 21:30.52 | dlynes_laptop | LoneShadow: the caller didn't authenticate using a username and password |
| 21:31.32 | LoneShadow | if I did IAX register from 2nd box to 1st box |
| 21:31.53 | hads | Are they both on static IP? |
| 21:32.04 | LoneShadow | and define the IAX profile on box 1, cant I just say DIAL(IAX/profile_name/exten.. |
| 21:32.07 | *** join/#asterisk JunK-Y (n=junky@modemcable205.175-81-70.mc.videotron.ca) |
| 21:32.16 | LoneShadow | both are dynamic |
| 21:32.23 | hads | Ah, bummer. |
| 21:33.06 | LoneShadow | since I register, I was hoping they know its a valid call :D |
| 21:33.34 | hads | Random! |
| 21:34.40 | hads | I wouldn't have any leftover coins if I had to buy Cisco kit. |
| 21:35.02 | LoneShadow | buy linksys gear :P |
| 21:35.43 | RoyK | iirc cisco bought linksys to allow them to sell cheap stuf |
| 21:37.10 | hads | LoneShadow: pb your iax.conf |
| 21:38.00 | hads | Fine then :) |
| 21:38.03 | *** join/#asterisk LoneShadow (n=duh@59.92.131.250) |
| 21:38.53 | LoneShadow | so whats the problem with dynamic ip ? |
| 21:39.10 | hads | host=dynamic ? |
| 21:39.20 | hads | pb your iax.conf |
| 21:39.23 | LoneShadow | yes |
| 21:39.52 | LoneShadow | its a messy one, since I couldnt get my IAX trunk working :D |
| 21:40.06 | hads | Well tidy it up then paste it. |
| 21:41.02 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 21:41.56 | *** join/#asterisk psymon101 (i=psymon10@24-236-222-55.dhcp.aldl.mi.charter.com) |
| 21:45.20 | LoneShadow | http://pastebin.ca/119465 |
| 21:46.09 | *** join/#asterisk logicwrath (n=dfgaeefa@c-68-60-121-112.hsd1.mi.comcast.net) |
| 21:46.19 | LoneShadow | hads: http://pastebin.ca/119465 |
| 21:47.04 | Un1x | «dlynes_laptop» wusup |
| 21:47.20 | Un1x | wusup |
| 21:47.26 | Un1x | err fucking gay script |
| 21:47.43 | Un1x | err fucking gay script |
| 21:47.44 | Un1x | : wusup |
| 21:47.46 | Un1x | :/ |
| 21:47.54 | Un1x | sup |
| 21:47.59 | LoneShadow | hi ? |
| 21:48.00 | Un1x | : sup |
| 21:48.10 | Un1x | ok brb gonna test elsewhere |
| 21:49.27 | Un1x | there we go fixed it :) |
| 21:49.36 | Un1x | + : wusup? |
| 21:49.41 | Un1x | omg nooo not again :( |
| 21:52.07 | *** join/#asterisk jeffgus (n=jeffgus@shaper.zimage.com) |
| 21:54.32 | *** join/#asterisk jart (n=user@ool-4356f82f.dyn.optonline.net) |
| 21:55.56 | Un1x | dlynes_laptop: you still around :P? |
| 21:56.12 | *** join/#asterisk LoneShadow (n=duh@59.92.128.123) |
| 21:56.22 | LoneShadow | are you there hads ? |
| 22:00.18 | *** join/#asterisk topping (n=topping@207.47.6.192.static.nextweb.net) |
| 22:02.34 | *** join/#asterisk mcreedjr (n=mcreedjr@68.72.158.2) |
| 22:03.08 | mcreedjr | Can anyone point me towards some caller id troubleshooting docs? Asterisk doesn't appear to be receiving CID from the PSTN (via a POTS line). |
| 22:06.53 | mcreedjr | Anyone? |
| 22:07.27 | *** join/#asterisk toerkeium (i=oo@201.216.206.221) |
| 22:07.49 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 22:09.50 | jart | mcreedjr: does it work with a normal phone? |
| 22:09.59 | mcreedjr | jart: aye |
| 22:10.48 | jart | double check your zapata config |
| 22:11.04 | jart | i doubt there are any "docs" on such a thing |
| 22:11.51 | mcreedjr | jart: thanks... i'll keep playing |
| 22:14.37 | Qwell | Do you have usecallerid set? |
| 22:15.01 | mcreedjr | Qwell: yeah, and callerid=asreceived |
| 22:15.29 | mcreedjr | Qwell: a NoOp on s,1 of the inbound context shows empty CID |
| 22:15.37 | Qwell | ahh |
| 22:15.40 | Qwell | do a Wait(2) |
| 22:15.53 | mcreedjr | Qwell: lol... lemme try |
| 22:15.53 | Qwell | welcome to analog |
| 22:16.08 | toerkeium | guys, I need to do some kind of "daemon" to check extensions status, and I know how to do it with php only, but I was told that it's not good to leavea php script running all the time.. any idea how can I do it ? |
| 22:16.29 | jart | what do you mean by extension status? |
| 22:16.48 | mcreedjr | Qwell: still empty :( |
| 22:16.49 | jart | and yes you can write it in php but php isn't meant for that sort of thing |
| 22:17.34 | toerkeium | jart: I will initiate a call, and then I will check with another script if extensions states are 1,2,3,4,5,etc in order to act based on the extension status response |
| 22:18.13 | *** join/#asterisk psymon101 (i=psymon10@24-236-222-55.dhcp.aldl.mi.charter.com) |
| 22:18.14 | jart | well, if you write a script that connects to the manager interface, it can keep track of that stuff |
| 22:18.28 | jart | because events like NewExten will be generated telling you where channels are in the dial plan |
| 22:18.38 | *** join/#asterisk Bobcat_1966 (n=chatzill@cpe-069-132-139-254.carolina.res.rr.com) |
| 22:18.50 | toerkeium | yeah, but how to do it "deamonized" ? |
| 22:18.55 | *** join/#asterisk Shark_y (n=paoloc@host181-54.pool8288.interbusiness.it) |
| 22:19.11 | jart | if you wrote it in C, you would call daemon(0, 0); |
| 22:19.27 | toerkeium | no, it's all php what I can do |
| 22:19.32 | jart | in php, your only real option is to run it in a screen session that is detached to the background |
| 22:19.35 | Shark_y | Anybody there for a help?..... thx |
| 22:19.39 | toerkeium | I mean, not idea how to program in C |
| 22:19.58 | jart | you should learn C, very easy |
| 22:20.17 | toerkeium | jart: what about doing an ajax script which calls to the php script in question? |
| 22:20.20 | toerkeium | what do you think = |
| 22:20.21 | toerkeium | ? |
| 22:20.57 | jart | it has to run under apache? |
| 22:21.01 | toerkeium | yes |
| 22:21.07 | jart | then mod_perl is the only way |
| 22:21.26 | jart | mod_perl modules are loaded in to memory and just keep running |
| 22:21.44 | jart | so you can have a thread in your mod_perl module talking to asterisk, and the other thread could be handling web browser requests |
| 22:21.52 | toerkeium | ok, so I better learn C or perl |
| 22:21.56 | jart | that's how i do my ajax stuff that talks to asterisk |
| 22:21.57 | Shark_y | I have to use an sip line... I can get it working in and out from a phone through a cisco fw with inspection but I cannot recive calls in asterisk, but I can place them |
| 22:22.09 | jart | and then i'll usually write the rest in php |
| 22:23.02 | toerkeium | I readed somewhere that sending about NNNN commands to asterisk through the AMI makes * crash eventually.. is tha true? is there a limit for AMI commands ? |
| 22:23.52 | jart | i don't know, i never stress tested the manager api |
| 22:24.02 | Shark_y | Should at least appear something in the cli if the call is incoming correctly but not configured correctly in the dialplan? |
| 22:24.25 | jart | but you don't have many other options for getting that data |
| 22:24.30 | *** part/#asterisk spackle[work] (n=spackle[@ip207-199-243-35.static.ishsi.com) |
| 22:24.31 | *** join/#asterisk jeffgus (n=jeffgus@shaper.zimage.com) |
| 22:24.47 | Shark_y | Please.... I'm spending days on that problem..... |
| 22:25.02 | jart | Shark_y: what problem |
| 22:25.06 | Mike | someone has a h488 im getting alot of hangups |
| 22:25.13 | Mike | ht488 gs |
| 22:25.37 | Shark_y | Not able to recive calls from a sip account |
| 22:25.52 | jart | you should ask your sip provider for help |
| 22:26.00 | *** join/#asterisk RoyK (n=roy@ti211310a080-8252.bb.online.no) |
| 22:26.02 | toerkeium | well.. thanks jart, gonna try some ideas |
| 22:26.05 | *** join/#asterisk apardo (n=apardo@87.217.144.193) |
| 22:26.19 | jart | toerkeium: good luck with mod_perl modules |
| 22:26.33 | Shark_y | The fw lets pass calls as soon as the sip phone gets calls |
| 22:26.39 | jart | toerkeium: mod_python might also be an easier to learn option |
| 22:30.09 | Shark_y | jart how can I see if it's an asterisk problem or a dialplan issue? should the cli show something, in the log I cannot see nothing significant even with a high verbosity |
| 22:31.39 | Shark_y | What does this mean? "devicestate.c: Changing state for ???? - state 4 (Invalid)" |
| 22:33.28 | *** join/#asterisk sugardave (n=not@cpe-66-68-165-207.austin.res.rr.com) |
| 22:34.25 | sugardave | Anybody playing with Asterisk and Google Talk? I managed to get a semi-working test last night, but am getting no audio |
| 22:36.51 | JunK-Y | sugardave: me neither, no audio. |
| 22:37.11 | *** join/#asterisk trbldwine (i=troubled@71.194.161.170) |
| 22:38.15 | *** part/#asterisk trbldwine (i=troubled@71.194.161.170) |
| 22:38.24 | *** join/#asterisk Dovid (n=dovi5988@pool-71-250-15-227.nwrknj.east.verizon.net) |
| 22:38.54 | *** join/#asterisk type0 (i=type0@148-220-223-66.gci.net) |
| 22:38.57 | Dovid | cab anyone help me with rtcachefriends=yes in asterisk real time ? |
| 22:39.15 | type0 | hey.. im running centos and when i go to compile asterisk im getting this |
| 22:39.15 | type0 | /usr/src/zaptel/zaptel.c:188: warning: 'fcstab' defined but not used |
| 22:39.15 | type0 | make[2]: *** [/usr/src/zaptel/zaptel.o] Error 1 |
| 22:39.15 | type0 | make[1]: *** [_module_/usr/src/zaptel] Error 2 |
| 22:39.15 | type0 | make[1]: Leaving directory `/usr/src/kernels/2.6.9-34.0.2.EL-i686' |
| 22:39.15 | type0 | make: *** [linux26] Error 2 |
| 22:39.31 | Dovid | type0: |
| 22:39.37 | Dovid | ~centosbug |
| 22:39.39 | jbot | rumour has it, centosbug is a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h" |
| 22:40.02 | Dovid | type0: are you trying to compile with ztdummy ?\ |
| 22:40.09 | type0 | yeah |
| 22:40.12 | Dovid | ok |
| 22:40.17 | Dovid | there is a bug |
| 22:40.26 | type0 | but what file do i specify ztdummy? |
| 22:40.31 | Shark_y | Still anybody there? |
| 22:40.38 | Dovid | using 2.6 kernel or 2.4 ? |
| 22:40.45 | Dovid | Shark_y: were alive here |
| 22:40.50 | type0 | 2.6.9 |
| 22:40.52 | Dovid | ok |
| 22:40.57 | Dovid | type0: then nothing |
| 22:41.00 | Dovid | just compile it |
| 22:41.08 | Dovid | but you have to apply the fix first |
| 22:41.17 | Dovid | do what the jbot says |
| 22:41.21 | Dovid | ~centosbug |
| 22:41.23 | jbot | [centosbug] a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h" |
| 22:41.39 | sugardave | JunK-Y: I was just glad to get the connections to work :) |
| 22:41.55 | sugardave | so, at least I'm not the only one who can't get audio |
| 22:41.57 | Dovid | Shark_y: do you have a question ? |
| 22:42.10 | Dovid | sugardave: what problem r u having ? |
| 22:42.41 | sugardave | Dovid: Can't seem to get audio to a Google Talk client |
| 22:42.55 | JunK-Y | sugardave: see the asterisk-dev mailing list, they are few ppl like us. |
| 22:43.09 | sugardave | JunK-Y: cool, will check, thanks |
| 22:43.10 | Dovid | sugardave: never played with that before. sorry. try the usual, NAT etc. |
| 22:43.36 | sugardave | Dovid: Yeah, was wondering if I need to poke some more holes in my firewall, etc...but will check on the dev list |
| 22:43.54 | type0 | how do you enable perl agi's? |
| 22:44.00 | Dovid | try DMZ and see if it works |
| 22:44.05 | Dovid | or try ethera |
| 22:44.08 | Dovid | ehteareal |
| 22:44.14 | Juggie | JunK-Y! |
| 22:44.21 | JunK-Y | Juggie: ! |
| 22:44.27 | Dovid | type0: they are enabled by defualt. u just have to create them |
| 22:44.50 | type0 | ok |
| 22:45.34 | brut- | is there any repository of agi scripts that people have released? |
| 22:46.01 | Dovid | brut-: search voip-info.org as well as google |
| 22:46.04 | Dovid | ~google |
| 22:46.05 | jbot | google is probably a search engine found at http://www.google.com/ |
| 22:46.24 | *** join/#asterisk teknoprep (n=teknopre@unaffiliated/teknoprep) |
| 22:46.38 | teknoprep | whats is trixbox default smtp server... or what server do the php pages use? |
| 22:46.53 | brut- | I've searched both of them, I'm looking for one that specifically deals with hyatt-encore/hobic coded PMS messages |
| 22:47.14 | Dovid | brut-: cant help u there. try the list |
| 22:47.22 | brut- | I'm guessing that one hasn't been developed yet, so I'll have to develop one myself I reckon :) |
| 22:47.38 | brut- | Dovid, thanks m8 |
| 22:47.42 | teknoprep | ? |
| 22:47.54 | JT | teknoprep: ???? |
| 22:47.55 | Dovid | yw |
| 22:48.44 | Juggie | ~seen theplot |
| 22:48.58 | jbot | theplot <i=ThePlot@202.164.38.210> was last seen on IRC in channel #asterisk, 4h 50m 7s ago, saying: 'I did set in the address field to match the username too'. |
| 22:48.58 | Juggie | damn jbot, you suck :) |
| 22:49.18 | Juggie | ahhh slow :) |
| 22:50.10 | teknoprep | whats is trixbox default smtp server... or what server do the php pages use? |
| 22:50.13 | teknoprep | already asked my questin |
| 22:51.23 | *** join/#asterisk Shark_y (n=paoloc@host181-54.pool8288.interbusiness.it) |
| 22:52.18 | JT | teknoprep: then what makes you think asking it again will increase the chances of it being answered? |
| 22:52.30 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 22:52.34 | JT | especially if it's not on topic for the channel |
| 22:52.43 | teknoprep | you are a weirdo |
| 22:52.45 | Mike | guey anyone knows what can cause this message "Received mini frame before first full voice frame" ? |
| 22:52.46 | teknoprep | oh wit |
| 22:52.46 | JT | hardly |
| 22:52.50 | teknoprep | wrong channel lol |
| 22:52.53 | Ciber311 | grrrrr |
| 22:53.00 | Ciber311 | this stupid phone pisses me off |
| 22:53.02 | teknoprep | my bad |
| 22:53.10 | JT | no worries |
| 22:53.46 | Ciber311 | anyone have a polycom 501, and if so what time server are you using? |
| 22:54.16 | JunK-Y | time.apple.com |
| 22:54.35 | Ciber311 | seriously? |
| 22:54.47 | Ciber311 | in ip format or just like that? |
| 22:54.47 | JunK-Y | ya |
| 22:54.53 | JunK-Y | like that. |
| 22:55.06 | type0 | $AGI->exec ("Dial", "IAX2/ntheory\@NuFone/$Digits"); .. thats in my agi |
| 22:55.21 | type0 | what should I replace it with if i've already spec'd the iax provider in my iax.conf |
| 22:58.35 | Qwell | JunK-Y: hey |
| 22:58.46 | Qwell | JunK-Y: Remember that program you had me test, when I got my new CPU? |
| 22:58.55 | JunK-Y | yep |
| 22:59.03 | Qwell | what was that? I wanna run it on this sunfire |
| 22:59.21 | JunK-Y | its Nugget's baby, its dnetc (www.distributed.net) |
| 22:59.26 | Qwell | cool |
| 22:59.38 | Qwell | http://uuoc.com/1558 Think it'll do well? |
| 23:00.07 | JunK-Y | nice! |
| 23:02.58 | Ciber311 | JunK-Y: stupid phone is still blinking the time and date |
| 23:02.59 | Ciber311 | grrrrrrrrr |
| 23:03.42 | wunderkin | Ciber311, are you a 12:00 blinker? hehe |
| 23:04.09 | Ciber311 | 7:03 pm :P |
| 23:04.47 | *** join/#asterisk phonic (i=phonic@antisocial.nu) |
| 23:05.49 | *** join/#asterisk trelane (i=trelane@unaffiliated/trelane) |
| 23:07.25 | Ciber311 | fixed it |
| 23:07.35 | Ciber311 | had to input it through the web interface |
| 23:07.48 | Ciber311 | wasn't accepting the change directly through the phone *rolls eyes* |
| 23:10.45 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 23:10.50 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 23:11.52 | *** join/#asterisk Trakkasure (n=Sgemtum@24-50-26-239.atlsfl.adelphia.net) |
| 23:12.41 | jart | hey junky |
| 23:12.41 | jart | email me those con photos when you get a chance |
| 23:27.27 | *** join/#asterisk supjigatr (n=syslod@152.53.17.26) |
| 23:30.22 | *** join/#asterisk apardo (n=apardo@87.217.146.97) |
| 23:30.46 | supjigatr | Anyone have occasional lost DTMF with sangoma? Any cons to turning on relaxdtmf? |
| 23:31.03 | *** join/#asterisk plexdk (n=plexus@3e6bdb32.rev.stofanet.dk) |
| 23:31.07 | plexdk | hi there.,. |
| 23:31.11 | supjigatr | Hi. |
| 23:31.58 | *** join/#asterisk inv_Arp (i=junya@c-67-191-62-53.hsd1.fl.comcast.net) |
| 23:32.03 | plexdk | I have a bit of a problem... My asterisk works fine for incoming calls.. but when i try to dial out, im getting the dialing tone alright, it just doesnt dial |
| 23:32.26 | supjigatr | What interface are you using? |
| 23:32.32 | *** join/#asterisk RoyK (n=roy@122.80-203-78.nextgentel.com) |
| 23:33.05 | plexdk | SIP |
| 23:33.34 | supjigatr | What SIP are you using? What DTMF? rfc, inband etc? |
| 23:33.41 | supjigatr | What sip device |
| 23:34.04 | *** join/#asterisk RoyK (n=roy@122.80-203-78.nextgentel.com) |
| 23:34.40 | plexdk | im hooked up to an IP ISP |
| 23:34.57 | plexdk | I recieve calls alright, just the outgoing there wont work |
| 23:35.12 | plexdk | exten => _.,1,Dial(SIP/viptel/${EXTEN},120,r) |
| 23:35.19 | plexdk | ^-- is my outgoing |
| 23:36.04 | *** join/#asterisk knarfly (n=bmorris@c-69-180-98-189.hsd1.fl.comcast.net) |
| 23:36.07 | supjigatr | Pastbin the log of the call. |
| 23:36.13 | knarfly | I just got a DID and have trouble...can anyone offer some help here? |
| 23:36.17 | plexdk | ok |
| 23:37.10 | knarfly | Im getting this error Aug 6 19:36:51 NOTICE[73385]: chan_sip.c:10515 handle_request_invite: Failed to authenticate user "3058291080" <sip:3058291080@voipgw1.splitinfinity.net>;tag=as58c287f7 |
| 23:37.24 | *** part/#asterisk phonic (i=phonic@antisocial.nu) |
| 23:37.36 | knarfly | does this mean I'm not registering with them or the other way around? |
| 23:38.04 | supjigatr | Wrong pass or username. |
| 23:38.25 | plexdk | hrm..pastebin is slow |
| 23:38.45 | knarfly | supjigatr: meaning the provider set me up wrong... |
| 23:38.47 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
| 23:39.11 | riddlebox | is this a valid way to use gotoiftime, http://pastebin.ca/119734 ? |
| 23:41.13 | plexdk | supjigatr: http://plex.dk/1/log.txt |
| 23:41.20 | plexdk | ..pastebin is timing out on me |
| 23:43.30 | plexdk | hmm..could it be a firewall problem? no firewlal on outgoing connections, but im behind nat |
| 23:44.49 | Un1x | knarfly |
| 23:44.59 | *** join/#asterisk jblachly (n=james@adsl-70-232-100-143.dsl.ltrkar.sbcglobal.net) |
| 23:45.24 | jblachly | I've a devlopment question ; is this the right channel? |
| 23:45.29 | supjigatr | NAT + SIP = BAD |
| 23:45.33 | supjigatr | usually |
| 23:45.34 | jart | jblachly: what is it |
| 23:45.39 | plexdk | ok |
| 23:45.45 | *** join/#asterisk toerkeium (i=oo@201.216.206.221) |
| 23:46.08 | jblachly | jart: What is the 'trunk' version on SVT? Is it the very latest code which has been approved but not merged into codebase? Reason I ask is |
| 23:46.15 | supjigatr | plexdk: Ur NAT is screwing up the call. |
| 23:46.26 | plexdk | tought it only was the defined port 5060 which needed to be routed? |
| 23:46.29 | jblachly | jart: that I've found and fixed a bug but it looks like while it is not fixed in the 1.2 branch on SVT there is a correction in "trunk" |
| 23:46.31 | plexdk | thought even |
| 23:46.51 | supjigatr | 5060 is just SIP control. The call is another port of RTP |
| 23:46.54 | jart | jblachly: then submit it to the bug tracker and tell them it needs to be merged from trunk to 1.2 |
| 23:47.05 | jart | and be sure to give them the revision to which the fix was committed |
| 23:47.06 | plexdk | ah.. where are those port defined? |
| 23:47.16 | supjigatr | Well your provider. |
| 23:47.26 | jart | then wait about 3 months for it to get in 1.2 svn, then another month for them to make a 1.2 release |
| 23:47.29 | supjigatr | What is you firewall? |
| 23:47.30 | plexdk | ok |
| 23:47.48 | plexdk | i've only opened up for 5060 |
| 23:47.57 | supjigatr | I use 10000 - 20000 |
| 23:47.59 | plexdk | I still do recieve calls without problems |
| 23:48.14 | plexdk | I have no firewall on outgoing connections |
| 23:48.24 | jart | btw, trunk is just the latest development code |
| 23:49.20 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 23:49.34 | shmaltz | hi everyone |
| 23:49.44 | jart | hi shmaltz! |
| 23:49.44 | jblachly | jart: thank you. since someone else has fixed it in trunk, is that destined automatically then to go into a future 1.2 release? I presume it is. In that case, I won't bother anyone. |
| 23:49.54 | plexdk | oh well.. how do i define the range of RTP ports? |
| 23:49.58 | plexdk | in the sip.conf |
| 23:50.04 | Qwell | jblachly: No, it will not. Not unless it was put into 1.2 branch |
| 23:50.15 | Qwell | Which, if it was a bug fix, it should have |
| 23:50.23 | shmaltz | hi jart |
| 23:50.27 | jart | jblachly: it depends on how good the bug marshalls are. Whoever fixed that bug should have addressed whether or not it was a problem in 1.2.... |
| 23:50.39 | jart | jblachly: you should try to checkout the latest 1.2 sources from subversion and see if it was fixes |
| 23:50.40 | shmaltz | plexdk, its in rtp.conf |
| 23:50.42 | jblachly | Qwell: My suspicion is that no one realized that it fixed a bug because the code in trunk was rewritten for a different reason. What is the best way to alert them? |
| 23:50.48 | jart | fixed* |
| 23:50.54 | jblachly | jart: The latest 1.2 from subversion does not have the corrected code |
| 23:50.57 | Qwell | jblachly: What is the bug number? |
| 23:51.12 | Qwell | and fyi; #asterisk-bugs |
| 23:51.16 | jblachly | Qwell: There was no bug report, AFAIK. I filed a new one 0007669 |
| 23:51.20 | jart | jblachly: then chances are the fix won't get in to 1.2 till you make a bug yelling at them |
| 23:52.13 | Qwell | jblachly: You need to test against latest 1.2 branch |
| 23:52.15 | Qwell | 1.2.4 is old |
| 23:52.38 | Juggie | if all else fails, ask russellb |
| 23:52.40 | jblachly | Qwell: Yes, I filed that bug earlier, but when I use the web version of subversion and look at the source for 1.2 , the erroneous code remains |
| 23:52.59 | Qwell | jblachly: Test against 1.2 - don't assume... |
| 23:53.10 | *** join/#asterisk s0lid (n=jlq@210.213.242.41) |
| 23:53.31 | jblachly | Qwell: I've looked at the source and it is identical. |
| 23:53.40 | jart | or else kevin fleming will smite thee |
| 23:53.47 | Qwell | As far as you know, the tone is played elsewhere. |
| 23:54.37 | jblachly | Qwell: I understand your enthusiasm for rigor but I know for a fact it's not. |
| 23:54.53 | jart | gentlemen, take it outside to teh bugtracker |
| 23:56.22 | jblachly | Qwell: final question: When I browse svn.digium.com/view/asterisk/1.2/ is that the latest 1.2 submitted to subversion? |
| 23:56.29 | jart | yes |
| 23:56.57 | jblachly | Alright then, do I need to submit a new bug, as it appears it is impossible to modify the fields in yoru original bug report :-/ |
| 23:57.21 | jart | jblachly: checkout svn checkout http://svn.digium.com/svn/asterisk/branches/1.2 asterisk-1.2 and take it for a run |
| 23:57.33 | *** join/#asterisk niZon (n=ilt@S010600080db4ab60.wp.shawcable.net) |