| 00:01.16 | Katty | :< |
| 00:01.31 | Katty | boyfriend's watching a scary movie |
| 00:07.31 | *** join/#asterisk sahafeez (n=sahafeez@65-119-47-100.dia.static.qwest.net) |
| 00:10.12 | *** join/#asterisk Yuda-israel1984 (n=Yuda-isr@94.159.199.233) |
| 00:11.28 | Yuda-israel1984 | hi guys does anyone have the correct sip config for rnk accounts?? |
| 00:11.44 | *** join/#asterisk sircolin (n=sircolin@my83-216-68-241.mynow.co.uk) |
| 00:13.16 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-218-155-145.cablep.bezeqint.net) |
| 00:15.56 | pawpro | Could anybody elaborate on the following warning: channel.c:3445 set_format: Unable to find a codec translation path from 0x4 (ulaw) to 0x8 (alaw)? |
| 00:17.44 | tzafrir_laptop | pawpro, hmmm.... any chance you actually miss some basic codec_* modules? |
| 00:18.15 | tzafrir_laptop | Do you load modules manually or automatically? |
| 00:19.51 | radic | all variables difened in [global] of the extension.conf still global if change them in an context? |
| 00:22.03 | pawpro | core show codecs shows all codecs including ulaw alaw gsm g729 etc |
| 00:24.26 | *** join/#asterisk manxpower (n=ewieling@24.42.221.26) |
| 00:24.44 | manxpower | ~answers |
| 00:24.45 | infobot | answers is, like, Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 00:25.17 | Katty | hi manx. |
| 00:25.19 | pawpro | i do load modules manually thou |
| 00:25.30 | pawpro | tzafrir_laptop: I load modules manually |
| 00:27.34 | radic | hmm |
| 00:27.48 | tzafrir_laptop | see: codec_ulaw.so, codec_alaw.so, codec_a_mu.so |
| 00:29.03 | tzafrir_laptop | 'core show codecs' shows all codec types known at build-time. Regardless of whether they are actually supported |
| 00:30.08 | pawpro | tzafrir_laptop: obiously i wasnt loading alaw. ehhh Thanks man! |
| 00:32.31 | *** join/#asterisk cosmicwombat (n=cosmicwo@69.7.44.68) |
| 00:36.47 | *** join/#asterisk geneticx (n=geneticx@adsl-10-114-216.mia.bellsouth.net) |
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| 01:02.07 | Katty | hi jaytee |
| 01:02.44 | jaytee | hi Katty |
| 01:02.48 | Katty | what'd ya have for dinner? |
| 01:02.57 | jaytee | fettucini alfredo |
| 01:03.23 | Katty | yum. |
| 01:04.09 | jaytee | feeling sad and listening to Seal |
| 01:04.21 | Katty | hugs jaytee |
| 01:04.25 | Katty | i will listen to seal with you. |
| 01:05.07 | jaytee | got home late after fight that trojan that got past our filters and had a message from my mom telling me by uncle passed early this evening |
| 01:05.12 | *** join/#asterisk chendy (n=chatzill@58.251.102.26) |
| 01:10.00 | loather-work | i like seal. i've got almost all his albums. |
| 01:11.07 | *** join/#asterisk Sandheaver (n=jeremiah@98.214.112.102) |
| 01:11.43 | Sandheaver | wikipedia down for anyone else? |
| 01:11.49 | Sandheaver | never saw wikipedia go down before. |
| 01:11.53 | *** join/#asterisk GameGamer43 (n=GameGame@CPE-65-27-78-156.new.res.rr.com) |
| 01:12.45 | *** join/#asterisk Kumbang (n=kumbang@125.163.83.153) |
| 01:15.21 | loather-work | works for me. |
| 01:15.31 | *** join/#asterisk dmz (n=dmz@64.203.233.195.dyn-cm-pool35.hargray.net) |
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| 01:48.49 | jblack | <PROTECTED> |
| 01:52.51 | *** part/#asterisk alevy (n=aaron@75-101-48-125.dsl.static.sonic.net) |
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| 02:00.50 | Katty | hey jblack (- |
| 02:00.52 | Katty | (= |
| 02:02.47 | raden_work | :P |
| 02:03.01 | raden_work | Katty, why you never say hi to me anymore :( |
| 02:06.03 | *** join/#asterisk dmz (n=dmz@64.203.233.195) |
| 02:32.17 | *** part/#asterisk sircolin (n=sircolin@my83-216-68-241.mynow.co.uk) |
| 02:33.15 | *** join/#asterisk alexshell (n=abc@unaffiliated/alexshell) |
| 02:33.35 | jblack | How is callcentric these days? |
| 02:33.55 | *** join/#asterisk mchou (n=quassel@unaffiliated/mchou) |
| 02:42.02 | *** join/#asterisk tjz (n=tjz@unaffiliated/tjz) |
| 02:42.16 | *** join/#asterisk sfr33man (n=sfreeman@64.183.147.98) |
| 02:44.06 | sfr33man | anyone seen asterisk segfault when running 'stop gracefully' at the cli? |
| 02:45.39 | *** join/#asterisk slinksh0t (n=slinksh0@c-76-108-31-194.hsd1.fl.comcast.net) |
| 02:50.49 | loather-work | i'd just like to find a free reporting interface for asterisk |
| 02:51.00 | loather-work | something that provides some basic reporting features and call center features. |
| 02:51.40 | Katty | Basic Reporting? |
| 02:51.49 | Katty | i know a nice php wrap around for Call Logs |
| 02:51.54 | Katty | but that might not be what you're talking about |
| 02:52.06 | loather-work | yeah, just something that'll read the CDR and do some basic per-extension reports, call details, etc. |
| 02:52.24 | loather-work | number of calls per hour, per day, per month, that kind of thing |
| 02:52.26 | Katty | google asterisk-stat |
| 02:53.30 | loather-work | this might be exactly what i'm looking for. |
| 02:55.37 | *** join/#asterisk fish-bulb (n=cstewart@nat/digium/x-nimfwfvlelsjvtjm) |
| 02:56.21 | KavanS | man I feel fucking l33t.. |
| 02:56.37 | KavanS | I hacked out this macro to navigate verizon and at&t voicemail systems...my followme was leaving "press 1 to accept" voicemails |
| 02:56.55 | KavanS | it's cheezy, but works...no matter if you press ignore, have phone off...or it just rings till it hits voicemail |
| 02:57.10 | KavanS | <3's * |
| 02:57.21 | Katty | smiles |
| 02:57.45 | KavanS | amd wasn't working for my needs... |
| 02:58.35 | jblack | I've been fighting with t-mobile for 3 days to reject rejected/unanswered calls. |
| 02:59.29 | loather-work | Katty: *perfect* -- thanks. this is exactly what I needed. |
| 02:59.35 | KavanS | jblack, I let the asterisk system leave the voicemail, then I have it navigate out by choosing to delete the message...send some garbage keys, and then it asks you, "press 1 to disconnect" for at&t |
| 02:59.37 | Katty | loather-work: cheers (= |
| 02:59.54 | Katty | now if only my ears would shut up |
| 03:00.07 | loather-work | ringing? |
| 03:00.16 | *** join/#asterisk mnicholson (n=mnichols@nat/digium/x-bhxdktvbbyhuiehf) |
| 03:00.16 | KavanS | did the exact same for verizon...tested both phones in on/off and reject modes |
| 03:00.22 | Katty | otherwise known as tinnitus |
| 03:00.24 | jblack | my phone is in a group. |
| 03:00.34 | KavanS | jblack, ring group? |
| 03:00.43 | loather-work | yah, i've got it too :( |
| 03:00.43 | jblack | my office phone, then all the phones in the house, then my cell phone. Then, I want the call to go back to *'s voicemail. |
| 03:01.15 | KavanS | jblack, I do that now...want me to share some snippets? |
| 03:01.17 | loather-work | i was a live sound engineer for a long time. my right ear has it worse than my left, but it's manageable. As long as there's ambient noise I don't notice it. |
| 03:01.26 | KavanS | my work is somewhat hackish as I've been toying with * for a couple months to get it the way I want it |
| 03:01.32 | KavanS | but it works I can tell you that... |
| 03:01.34 | jblack | kavans: With t-mobile? Becuase like I said, I can't get them to turn voicemail off. |
| 03:01.54 | KavanS | jblack, no need to turn off voicemail...call your tmobile with landline, leave a voicemail, press pound... |
| 03:01.56 | jblack | which means the call _always_ gets answered. |
| 03:02.04 | KavanS | then tell it to delete message and press some garbage keys |
| 03:02.07 | KavanS | make it disconnect you |
| 03:02.14 | jblack | ?? |
| 03:02.16 | KavanS | then use "SendDTMF" to emulate this |
| 03:02.28 | KavanS | I have it repeat the same routine twice...seems to work flawlessly |
| 03:02.32 | KavanS | let me pastebin... |
| 03:02.44 | jblack | You want me to tell people that are trying to call me, to do a fake voice mail, delete it, and call back my main number, and go straight to voicemail? |
| 03:02.58 | KavanS | jblack, no...your asterisk system does this |
| 03:03.11 | KavanS | jblack, you just don't notice it...this is only related to your tmobile voicemial |
| 03:03.28 | *** join/#asterisk dlewis (n=dlewis@about/security/staff/dlewis) |
| 03:03.35 | jblack | Yeah, I'd like to see what you're doing. |
| 03:03.38 | dlewis | greetings |
| 03:03.46 | KavanS | jblack, my system rings my desk phone...no answer for 5 secs, rings deskphone, cellphone, sipphone... |
| 03:03.55 | Katty | hi dlewis |
| 03:03.57 | KavanS | during cellphone ringing, it waits for user to press 1 to accept call |
| 03:04.13 | jblack | user being you? |
| 03:04.20 | KavanS | during that time, at the end of the loop...asterisk sends DTMF to tmobile/verizon/at&t to disconnect/delete the message |
| 03:04.31 | KavanS | yep, user being person who's cell phone it is |
| 03:04.37 | KavanS | in this case it is me, but I use this for other people as well :) |
| 03:04.37 | jblack | i understand what you're doing. |
| 03:04.46 | KavanS | ok, lemme share |
| 03:04.50 | jblack | how do you inject dtmf after the fact? I thought * couldn't do that |
| 03:04.57 | *** join/#asterisk chendy (n=chatzill@113.91.36.164) |
| 03:05.09 | dlewis | gonna open up a can of worms here |
| 03:05.16 | Katty | oh boy! |
| 03:05.19 | Katty | i'll get my bug spray |
| 03:05.24 | dlewis | but, what are yalls opinion on the best phone for asterisk (regardless of price)? |
| 03:05.42 | Katty | i prefer polycom |
| 03:05.44 | jblack | best phone is an email client. |
| 03:05.51 | Katty | infobot: phones? |
| 03:05.52 | infobot | While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. |
| 03:05.52 | loather-work | polycom, aastra, snom are all good |
| 03:06.07 | dlewis | ok |
| 03:06.15 | KavanS | jblack, hell yeah, I'm injecting DTMF to the party on the other end ;) |
| 03:06.19 | dlewis | I was looking at then Aastra 57i CT |
| 03:06.23 | Katty | ^_- |
| 03:06.26 | KavanS | jblack, pasting you my stuff...it'll take me a sec |
| 03:06.30 | Katty | interesting. right ear is louder than the left. |
| 03:06.36 | Katty | dlewis: what price range are you working with? |
| 03:06.45 | Katty | dlewis: the polycom 320/330 is very good for it's price. |
| 03:06.54 | Sandheaver | yes |
| 03:07.01 | Sandheaver | 320 = $85 |
| 03:07.03 | dlewis | Katty: cost isn't an issue for this client |
| 03:07.18 | dlewis | he wants the best available |
| 03:07.21 | Sandheaver | howcome wikipedia is down for me |
| 03:07.49 | *** join/#asterisk manxpower (n=ewieling@24.42.221.26) |
| 03:07.55 | Katty | well in that case i'd go with color LCD polycoms |
| 03:08.00 | Sandheaver | dlewis: if he wants to brag to his college buddies, cisco phones. but you must pay for a Call Manager license even though you're using Asterisk |
| 03:08.21 | Sandheaver | and at that point it's just a small jump to just use Cisco all around i guess |
| 03:08.28 | dlewis | Sandheaver: outside of Cisco |
| 03:08.33 | Sandheaver | Polycom then |
| 03:08.58 | Sandheaver | Katty: wikipedia working for you? http://en.wikipedia.org/ |
| 03:09.47 | manxpower | ~phones |
| 03:09.48 | infobot | While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. |
| 03:09.48 | Katty | Sandheaver: works for me |
| 03:09.48 | Sandheaver | hrm. |
| 03:09.48 | dlewis | ok |
| 03:09.48 | Katty | dns problems? |
| 03:09.49 | Sandheaver | I get Invalid URL The requested URL "/", is invalid. |
| 03:09.54 | manxpower | Smells like a proxy to me |
| 03:10.04 | Sandheaver | probably, manxpower |
| 03:10.06 | Sandheaver | but not mine |
| 03:10.18 | Katty | can you bring up http://208.80.152.2 |
| 03:10.34 | Sandheaver | Katty: yes, says "wiki does not exist" |
| 03:10.39 | Katty | ^_- |
| 03:10.58 | Sandheaver | lots of links to various wikipedias |
| 03:12.13 | Katty | weird. |
| 03:12.27 | KavanS | jblack, http://pastey.net/126641 |
| 03:14.23 | jblack | Oh, you're using zap. |
| 03:14.29 | KavanS | jblack, and SIP |
| 03:14.42 | jblack | I believe SendDTMF only works for zap. I don't believe it works inband for sip |
| 03:14.42 | KavanS | jblack, I use failover logic in there to make it so if Zap is congested use my SIP channel(s) |
| 03:14.59 | KavanS | ahh no shit? |
| 03:15.03 | KavanS | christ... |
| 03:15.41 | KavanS | http://www.voip-info.org/wiki/view/Asterisk+cmd+SendDTMF |
| 03:15.48 | KavanS | hrm doesn't seem to say anything about not working on SIP lines |
| 03:16.10 | jblack | Try it. |
| 03:16.21 | KavanS | testing now |
| 03:16.24 | KavanS | blocking up the zap line... |
| 03:17.18 | *** join/#asterisk dssman (n=no@CPE001d7e602900-CM0011aec52a9c.cpe.net.cable.rogers.com) |
| 03:17.48 | dssman | hey there... anyone know of a tapi client that works with vista x64? |
| 03:18.13 | KavanS | jblack, so far so good...got some interesting output, but I think the SIP channel went dead as expected (SendDTMF executed successfully) |
| 03:18.25 | KavanS | waiting for any possible voicemail now... |
| 03:19.23 | alexshell | does someone provides paid support? estabilished company? |
| 03:20.06 | *** part/#asterisk dlewis (n=dlewis@about/security/staff/dlewis) |
| 03:20.24 | KavanS | alexshell, www.digium.org best support around |
| 03:20.40 | KavanS | jblack, yep no voicemails either on SIP line...sendDTMF looks like it works |
| 03:21.30 | alexshell | KavanS, I want it to get support to integrate asterisk with MS-SQL database |
| 03:21.35 | jblack | test by calling a line you set up to inject dtmf in the middle. |
| 03:22.06 | alexshell | KavanS, AGI expertise perhaps |
| 03:22.17 | KavanS | alexshell, pretty sure digium would be a good source for this :) |
| 03:22.35 | jblack | Hrrr. callcentric is dropping packets |
| 03:22.46 | alexshell | ok, will try KavanS thankyou! |
| 03:23.36 | alexshell | KavanS, you mean digium.com right? |
| 03:24.10 | KavanS | alexshell, yes I do! heh sorry it is late for me :) |
| 03:24.22 | KavanS | alexshell, http://www.digium.com/en/supportcenter/ I'm sure they'd help you integrate with MS-SQL |
| 03:25.02 | alexshell | np KavanS, thank you again :) |
| 03:25.17 | KavanS | jblack, I've tested the DTMF in a SIP call before by calling myself...so I know it's sending them |
| 03:25.44 | KavanS | alexshell, no worries, good luck :) |
| 03:28.37 | jblack | kavans: Ok |
| 03:34.41 | *** join/#asterisk lowlevel (n=Stuart@lowlevel.ca) |
| 03:51.01 | jblack | One of these sides is screwing up. I can't tell if it's callcentric or sipdroid |
| 03:53.08 | jblack | callcentric. |
| 03:57.58 | *** join/#asterisk effigee (n=jenshens@S0106001d7e52bc10.vc.shawcable.net) |
| 03:58.13 | *** join/#asterisk r0oter (n=gerardoj@66-191-135-178.static.roch.mn.charter.com) |
| 04:01.40 | jblack | found it. |
| 04:01.44 | jblack | pfm options. |
| 04:06.33 | r0oter | hey everybody, I have setup my asterisk server and my spa3102 adapter. So far when I make a call to the asterisk server, asterisk pick the call and I get dial tone to dial the extension. Im trying to get the menu right when asterisk get the call. Could u guys help me out with the dialplan? |
| 04:09.42 | *** join/#asterisk effigee (n=jenshens@S0106001d7e52bc10.vc.shawcable.net) |
| 04:10.28 | *** join/#asterisk xuser_ (n=xuser@unaffiliated/xuser) |
| 04:12.10 | drmessano | jblack: Using Callcentric now? |
| 04:12.48 | dssman | *** anyone around... I jsut ran "conary update sendmail" and my manager.conf file seems to have disappeard |
| 04:13.02 | drmessano | uh WUT? |
| 04:13.07 | *** join/#asterisk chendy (n=chatzill@58.250.8.198) |
| 04:13.20 | dssman | that at me? |
| 04:13.33 | drmessano | I would imagine you did more than that |
| 04:13.40 | dssman | I updated sendmail and restarted the process and she dont work :S |
| 04:13.41 | *** join/#asterisk JKac3BEq (n=JKac3BEq@173-20-71-143.client.mchsi.com) |
| 04:13.53 | drmessano | The file is gone? |
| 04:13.56 | dssman | I can send you my shell session! |
| 04:13.58 | dssman | it appears so |
| 04:14.12 | dssman | I did "locate manager.conf" and I get nothin |
| 04:14.21 | drmessano | I dont want your shell session, I dont accept EXE attachments, KTHX |
| 04:14.27 | dssman | lolz |
| 04:14.35 | drmessano | How about nano /etc/asterisk/manager.conf? |
| 04:14.57 | dssman | hmm |
| 04:15.02 | dssman | its there, but permission denied |
| 04:15.07 | dssman | lemme see rights |
| 04:17.08 | dssman | -rw------- |
| 04:18.37 | p3nguin | You'll want to be root, of course. |
| 04:19.03 | dssman | I sudo'd iut |
| 04:19.17 | p3nguin | sudo vim /etc/asterisk/manager.conf? |
| 04:19.30 | p3nguin | You're using ubuntu? |
| 04:19.30 | *** join/#asterisk gmarsh (n=gmarsh@c-98-223-193-198.hsd1.in.comcast.net) |
| 04:19.38 | dssman | Im running *now |
| 04:19.51 | p3nguin | Does it even have sudo configured for you to be able to do that? |
| 04:20.08 | p3nguin | Better just become root and try again. |
| 04:22.24 | dssman | shits weird... I just tried to restore and all my restore files are gone... there were 3 there yesterday |
| 04:23.50 | dssman | I sent the box down for a reboot and I have my manager.conf access back again |
| 04:23.57 | dssman | but shit is weirddddddd |
| 04:24.54 | p3nguin | Like rebooting will magically fix the problem? |
| 04:25.05 | drmessano | ZOMG hard rock asterisk covers |
| 04:25.44 | drmessano | Got the best idea for hold music, ever.. Except all that RIAA stuff |
| 04:25.45 | dssman | that was the hope and for some reason it did |
| 04:25.55 | drmessano | Rebooting fixes nothing.. YOU LIE |
| 04:26.07 | dssman | it fixes all in windows :D |
| 04:26.18 | dssman | but yea, I have no idea |
| 04:27.07 | dssman | hmmm now my backup engine is still dead |
| 04:27.52 | *** join/#asterisk Mango (n=Mango@96.49.67.94) |
| 04:28.27 | Mango | If Asterisk loses and then regains its internet connection, does it attempt to reregister at intervals by default? |
| 04:28.38 | Mango | I got the network back up remotely but |
| 04:28.43 | Mango | Asterisk isn't coming online. |
| 04:30.44 | dssman | as far as I know it does |
| 04:34.00 | Mango | Heh. Then maybe I'm in for a drive. |
| 04:34.53 | *** join/#asterisk sahafeez (n=sahafeez@65-119-47-100.dia.static.qwest.net) |
| 04:50.04 | mchou | drmessano: you happen to know if flowroute does cname lookups on inbound callerids? |
| 04:50.25 | drmessano | Dunno |
| 04:50.54 | mchou | drmessano: I gather you dont use their DIDs? |
| 04:51.02 | drmessano | Nope |
| 04:51.44 | mchou | drmessano: who do you use for DIDs (aside from google voice/ipkall)? |
| 04:52.36 | drmessano | Told you, I have no friends.. My google voice number gets the most action. I get maybe 10 calls a month. 3 are telemarketers, 5 are turning me down for job interviews, 1 is a wrong number, and the last is my monthly call from my ex-wife wanting her flowbee back |
| 04:53.02 | Katty | :< |
| 04:53.15 | drmessano | ITS MY FLOWBEE NOW |
| 04:53.17 | drmessano | U HEAR ME |
| 04:53.24 | mchou | lol |
| 04:53.37 | drmessano | WITCH WITH A "B" |
| 04:53.40 | drmessano | TAKE THAT! |
| 04:54.18 | mchou | bah |
| 04:54.27 | drmessano | mchou: I used Les.net a lot for a while |
| 04:54.45 | drmessano | mchou: Somehow all my calling got shifted to my cell, and I am in the process of fixing that |
| 04:55.30 | drmessano | I prefer flowroute for termination, and really, if I could get Gvoice direct to Asterisk, my life would be complete |
| 04:56.01 | mchou | I dunno. I need to find an ITSP that doesnt meter for incoming (and doesnt charge an arm & leg either) |
| 04:56.22 | drmessano | Flowroute has $6.95 unlimited inbound |
| 04:56.27 | *** join/#asterisk nsgn (n=brandonb@cpe-24-27-49-209.austin.res.rr.com) |
| 04:56.35 | mchou | yeah, that's tied per DID |
| 04:56.48 | drmessano | and VPRI |
| 04:56.50 | nsgn | goodevening. haven't been here for a while. last asterisk box i put together has been running 6 months without a peep, though. |
| 04:57.09 | nsgn | here to ask recommendations for a reputable online shop to get digium cards |
| 04:57.17 | nsgn | specifically the TDM808E |
| 04:57.18 | drmessano | digium.com |
| 04:58.17 | nsgn | they don't carry that specific configuration |
| 04:58.38 | nsgn | and i thought i remember that the configuration that number indicates was cheaper than buying the components separately |
| 04:58.49 | nsgn | though i cant recall where the heck i ended up getting the card |
| 04:59.21 | drmessano | http://store.digium.com/productview.php?product_code=1TDM808EF |
| 05:00.35 | mchou | bah |
| 05:00.39 | nsgn | weird, their search didnt find that card. unfortunately their price is pretty bad compared to the one other place i trust, telephonydepot |
| 05:00.53 | mchou | google was able to find a lower price |
| 05:01.04 | drmessano | You asked for reputable |
| 05:01.14 | drmessano | Perhaps you need to refine your terms |
| 05:01.21 | nsgn | yeah, i didnt want to just buy cheapest on google. i need the thing in a timely manner, in good condition, etc |
| 05:01.26 | drmessano | "Cheapest possible where I wont get screwed" |
| 05:01.34 | nsgn | i've used and trust telephony depot, and would trust digium |
| 05:01.38 | mchou | nsgn: http://www.voiplink.com/Digium_TDM808E_p/digium-tdm808e.htm |
| 05:01.43 | nsgn | i just wanted to see if there were others people had good word for |
| 05:01.52 | mchou | that's courtesy of google |
| 05:02.54 | nsgn | has anyone used that store? my criteria was a store someone has used and had a smooth experience with. if i'm gonna buy over a grand of stuff it's not gonna be from someone's yahoo store :) |
| 05:03.23 | drmessano | Christ |
| 05:03.53 | mchou | the problem has been overconstrained |
| 05:04.02 | nsgn | is it too much to simply ask for a trusted store? |
| 05:04.13 | mchou | no |
| 05:04.30 | nsgn | no issue if nobody knows of one, but i figured you would be the people to know |
| 05:04.40 | nsgn | was just interested in shopping around beyond telephonydepot |
| 05:04.42 | mchou | but trust means nothing when the govt has to bail out Goldman Sachs |
| 05:05.01 | mchou | that's not a yahoo store |
| 05:05.32 | drmessano | Yeah, they're one of the biggest telephony stores |
| 05:05.42 | drmessano | Your google foo = fail |
| 05:06.01 | mchou | drmessano: you talking to me? |
| 05:06.08 | nsgn | probably me. i found that store earlier when searching for that card, but wanted to know if someone has had a good experience with them |
| 05:06.19 | drmessano | No, mchou, DUH |
| 05:06.37 | drmessano | nsgn: Someone probably has, someone probably hasnt |
| 05:06.38 | nsgn | that's what i was asking. for a store someone had had a plesant transaction with before |
| 05:06.48 | nsgn | * someone being one of you in here |
| 05:06.53 | drmessano | newegg owes me $42 |
| 05:06.54 | mchou | lol |
| 05:06.58 | nsgn | hah |
| 05:07.01 | *** join/#asterisk chendy (n=chatzill@58.250.11.120) |
| 05:07.02 | drmessano | Does that mean they suck? |
| 05:07.04 | mchou | you trust people on IRC |
| 05:07.21 | mchou | that's like blind leading the blind :) |
| 05:07.22 | nsgn | i've gotten fine advice from kind people in here many a time before |
| 05:07.35 | nsgn | but though you get what i'm asking you clearly don't desire to help. thanks for the time though |
| 05:07.38 | drmessano | I dont find it particulary funny they fucked me over on an RMA, but if it amuses you, feel free to continue laughing |
| 05:07.41 | Mango | mchou: I'm late to the party, but you may want to consider Callcentric's Dirt Cheap DID or VoIP.ms. |
| 05:08.02 | nsgn | drmessano: was amusing in the sense that it was a good example. i enjoy newegg but i'm sure they have experiences like that |
| 05:08.07 | mchou | Mango: Callcentric and "cheap" is an oxymoron |
| 05:08.26 | drmessano | I also bought my laptop from them and have had no problems otherwise.. |
| 05:08.35 | mchou | Mango: see you don't get my trust |
| 05:08.45 | mchou | mango: :) |
| 05:09.02 | nsgn | thanks all, i'm out |
| 05:09.19 | drmessano | Fuckwit |
| 05:09.30 | drmessano | Ah,did I type that? |
| 05:09.32 | drmessano | . |
| 05:09.34 | drmessano | Yeah |
| 05:09.52 | mchou | the dude missed: Govt and educational Accts on that page |
| 05:10.07 | mchou | dont mess with Uncle Sam |
| 05:10.41 | drmessano | He missed the part about looking for google hits on a retailer |
| 05:10.41 | mchou | IRS screw you over with broomstick (that's tem being kind) |
| 05:10.49 | mchou | them* |
| 05:11.11 | Mango | mchou: Dirt Cheap DIDs are $3/month, 2 channels, unlimited incoming. |
| 05:11.28 | Mango | Callcentric and "cheap" ain't always an oxymoron :) |
| 05:11.47 | Mango | They're not available in all rate centres though. Hence cheap. |
| 05:12.14 | mchou | Mango: the 2 channel limitation is a sick joke |
| 05:12.21 | Mango | Oh? |
| 05:12.52 | mchou | ideally I'd have a pool of DIDs... |
| 05:14.00 | Mango | Continue... |
| 05:14.54 | mchou | how many ITSPs limit 2 channels/DID? |
| 05:15.08 | Mango | For $3/month? ;) |
| 05:15.40 | Mango | VoIP.ms starts at $0.99/month for 25 channels but they charge per minute. |
| 05:15.43 | mchou | diamondcard.us is $2/mo (if you prepay yr.) |
| 05:16.07 | Mango | how many channels? |
| 05:16.11 | mchou | of course diamondcard also meters per minute inbound |
| 05:16.35 | mchou | as many channels as you account balance will alow |
| 05:16.42 | mchou | allow* |
| 05:17.04 | Mango | I still don't see how 2 channels is a sick joke, assuming normal residential usage. |
| 05:17.38 | mchou | It's not residential. I's actually a doctor's practice |
| 05:17.44 | mchou | it's* |
| 05:18.01 | Mango | Okay. |
| 05:18.10 | mchou | that's as close as you can get to residential w/o being residential |
| 05:18.32 | mchou | in terms of usage |
| 05:20.50 | mchou | AMA should just have an ITSP for all members |
| 05:21.23 | mchou | Doctors to freaking dumb to come up with that kind of plan |
| 05:21.30 | mchou | too* |
| 05:23.17 | *** join/#asterisk soman (n=somnath@115.119.41.110) |
| 05:23.41 | mchou | I find it incredible that hospitals have gone all voip but the doctors have NO idea how the stuff works |
| 05:23.59 | Mango | ...why? |
| 05:24.46 | mchou | cause they use it every day and can't figure out how to bring it into their own practive |
| 05:25.02 | mchou | practice* |
| 05:25.02 | drmessano | They have no concept of transports |
| 05:27.15 | mchou | and hospitals have gone whole hog with thin clients.... |
| 05:27.16 | *** join/#asterisk Tim_Toady (n=moi@adsl45-72.kln.forthnet.gr) |
| 05:27.50 | mchou | Doctors havent caught up on that either in their own practice |
| 05:28.22 | Mango | Every doctor/lawyer/etc I've ever worked for hasn't cared about technology, so long as they can do their job as easily as possible. |
| 05:28.35 | mchou | oh, they care |
| 05:28.41 | effigee | with those people, upgrading takes money out of their own pockets |
| 05:29.02 | effigee | so they point at the phone they have, which is some merdian pbx being run by a dinosaur and they go "look its a phone, why do we need new ones" |
| 05:29.03 | mchou | Voip is not an "upgrade" per se |
| 05:29.46 | mchou | effigee: seriously, I dont see many doctors with key systems |
| 05:30.38 | effigee | i work at a law firm, i guess thats where i was speaking from |
| 05:30.46 | effigee | it just depends on the size of the place |
| 05:30.51 | mchou | effigee: indeed |
| 05:31.18 | drmessano | Small doctors offices typically have key systems |
| 05:32.08 | mchou | I havent priced it out, but I'd think voip would be cheaper than installing a key system |
| 05:32.39 | mchou | I mean if you're just starting a practce |
| 05:32.39 | drmessano | Removing the key system costs them money |
| 05:32.53 | drmessano | Thats why they dont switch |
| 05:33.05 | drmessano | New offices I have seen go heavy into voip |
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| 07:07.56 | xrmx__ | how can i disable a specific protocol (mgcp, dundi, skinny) support? put an empty bindaddr in conf file? |
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| 07:09.53 | Tim_Toady | xrmx__ dont load the modules |
| 07:11.26 | xrmx__ | Tim_Toady, right, thanks |
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| 07:34.30 | *** join/#asterisk chodorenko (n=chodoren@ext.one.by) |
| 07:34.35 | chodorenko | Hi All |
| 07:35.24 | chodorenko | please answer howe to i can add in sip.conf two subnet for parametr localnet ? localnet=192.168.0.0/16 10.9.0.0/24 is not work |
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| 07:53.23 | fiddur | chodorenko: http://www.voip-info.org/wiki/view/Asterisk+SIP+localnet |
| 07:55.21 | chodorenko | fiddur: THX |
| 07:56.21 | fiddur | chodorenko: Or maybe I should have said http://www.voip-info.org/wiki/view/Asterisk+SIP+localnet ;) |
| 07:56.24 | fiddur | oops |
| 07:56.50 | *** join/#asterisk DelphiWorld (n=Miranda@196.20.95.121) |
| 07:56.51 | fiddur | i mean http://tinyurl.com/ygblp5p |
| 07:56.52 | DelphiWorld | hi all |
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| 07:57.18 | ChannelZ | greetings citizen |
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| 09:09.00 | garymc | WOWSER im having some hassle here |
| 09:09.13 | garymc | anyone in the uk and use ISDN 30 with Asterisk? |
| 09:09.22 | garymc | pri |
| 09:10.01 | kaldemar | just ask if you have a specific issue |
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| 09:34.44 | mattboll | Hi all, after a few minutes, between 1 and 15, the conversation stops. Actually I haven't got the tone but we can't here each other… any idea ? I use Asterisk SVN-branch-1.4-r184842 and here is the cli with debug : http://pastebin.com/d14826b39 but I don't see anything usefull in there :( |
| 09:35.02 | mattboll | s/here/ear/ |
| 09:35.32 | mattboll | thanks infobot :p |
| 09:43.44 | kaldemar | that debug is completely useless. it only shows a few OPTIONS-OK dialogs. |
| 09:48.15 | mattboll | kaldemar: yeah it's what I saw :( but what should I do ? |
| 09:49.40 | kaldemar | show the relevant dialplan, show CLI output with sip debug when the call is made, and try to grab sip debug when the conversation stops. start with the first two. |
| 09:49.50 | garymc | kaldemar i spoke to TKD-fender showed him my log file and he said that BT was disconnecting my outgoing call. I get incoming calls fine. Take a look at my log file. http://pastebin.ca/1622345 |
| 09:50.08 | garymc | lines 99- 107 |
| 09:50.59 | garymc | So i called BT back , they cant look at PRI debug info they just tell me that the line is open and working and thats all they know. Any ideas what could be causing this? |
| 09:51.12 | kaldemar | garymc: < Message type: DISCONNECT (69) -- Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal Event (0) |
| 09:51.21 | garymc | yes |
| 09:52.05 | garymc | Thats me making an outgoing call to an existing number. I just get a asterisk voice message "all circuits are busy now" |
| 09:53.44 | mattboll | kaldemar: the pastebin contains "sip set debug peer BNobody_116" in CLI |
| 09:55.33 | kaldemar | mattboll: that pastebin doesn't contain a CALL |
| 10:01.03 | garymc | kaldemar : So what does that line mean above^^ |
| 10:01.16 | garymc | i need help with this |
| 10:01.19 | kaldemar | garymc: you're having a numbering/dialplan issue. bt is rejecting the call to that number, so you need to find out what you can dial and build your dialplan accordingly. for building the dialplan, you're making these questions in the wrong place since you use freepbx. |
| 10:01.52 | kaldemar | unallocated number means that the number you sent to BT is unallocated according to them. |
| 10:01.54 | garymc | ok so you think it is a dial plan issue? How would i find out the dial plan required |
| 10:02.11 | kaldemar | i just told you. |
| 10:02.53 | kaldemar | or did i? no, maybe not. ask BT about numbering if you don't know what numbers you can dial. |
| 10:04.48 | drcarumas | Hi everyone. I'm trying to append to a variable several results. Something like this $var1=resultfromcounter1,resultfromcounter2,resultfromcounter3 . I just cant find a good way to do this in the dialplan. I have a counter that gives me results a then i have variables with values from that counter like var1=100 var2=333 , but i can have any number of vars from that counter and than i want to append those results to one single vari |
| 10:04.48 | drcarumas | able. I just dont see any function that allows me to append in this dynamic way. |
| 10:05.05 | drcarumas | Thanks in advanced. |
| 10:06.37 | mattboll | this one contains a full call http://pastebin.com/db2c5da2 |
| 10:12.38 | kaldemar | drcarumas: i'd say you can do that with apps While, Endwhile, Set and possibly func EVAL |
| 10:14.33 | drcarumas | kaldemar, thanks for your reply. Yes i'm using while endwhile. I was trying with arrays but asterisk cant read all values from array , exept with a while. Hmm. let me check that EVAL. I'm running all the functions in asterisk manual so i can see what i can do. |
| 10:16.45 | drcarumas | kaldemar, hmm i think EVAL wont do : "Using EVAL basically causes a string to be evaluated twice." |
| 10:17.25 | drcarumas | kaldemar, what i need is to append something that in php or in bash should be easy. using "." to append values. |
| 10:20.43 | kaldemar | mattboll: looks like the NAT router in the other end is dropping it, not sure though |
| 10:22.24 | kaldemar | drcarumas: you can append with Set(var1=${var1}${nextvar}) |
| 10:22.39 | kaldemar | appending is the easiest part. |
| 10:22.50 | *** join/#asterisk MaliutaLap (n=biteme@203.171.195.112) |
| 10:23.39 | kaldemar | for nextvar, you might need eval: var1=${var1}${EVAL(var${counter})} |
| 10:23.41 | drcarumas | kaldemar, yes i can do that the thing is that this ${nextvar} is dinamic. Like nextvar1, nextvar2 . And i can have 10 , 5 , any number of them |
| 10:24.20 | kaldemar | i'd try that first. |
| 10:24.31 | drcarumas | ohh okk |
| 10:24.53 | drcarumas | kaldemar, i'll try that. :) |
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| 10:31.36 | Katty | :< |
| 10:32.45 | Katty | Dear Universe, I understand that life is not always cheery and bright, but it would be nice to at least be able to face these things on a full night's sleep instead of 3 hours. Think we could work out a peaceful resolution? Love, Katty |
| 10:36.00 | Faustov | seconds that |
| 10:37.42 | drcarumas | kaldemar, i think this is it. i'm just not getting the right value for now, trying to ajust to what i have. Thanks again . Brb |
| 10:39.46 | *** join/#asterisk TSM2 (n=the_soft@fw-lon1.wenn.com) |
| 10:40.21 | TSM2 | is there anyway to make it so that when you park calls it returns a SIP header to the phone, polycoms have the ability to show messages on the screen from sip headers |
| 10:43.28 | mattboll | kaldemar: thanks for looking at my problème. Any idea what I should do to monitor/debug/whatever ? |
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| 11:13.48 | TSM2 | is it possable on polycoms to only show certian buttons depending on who is calling? |
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| 11:23.01 | rabbit7 | hello, i have some problem with ring groups ? |
| 11:25.34 | rabbit7 | i use ring strategy hunt, i just dont understand how long "hunt" will call the phones in the extensionlist |
| 11:26.59 | Faustov | does it make sense not to load all kinds of modules as most tutorials say but select only the ones that are required? |
| 11:27.09 | Faustov | or is it more trouble than possible gain? |
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| 11:55.39 | TSM2 | is it normal for * not to hangup after you transfer a call to park? |
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| 12:01.56 | TSM2 | im trying to find out where the park lines are in the conf files, is it all done in the app |
| 12:05.29 | *** join/#asterisk zmitya (n=mitya@gw.gammatelecom.hu) |
| 12:05.35 | zmitya | hi everybody |
| 12:06.22 | zmitya | guys, how many RTP session can be passed through by asterisk *without* codec translation ? |
| 12:06.36 | zmitya | lets say I have a _really_ powerful server |
| 12:06.50 | *** join/#asterisk baijum (n=baiju@122.166.46.113) |
| 12:07.09 | zmitya | lets say with G711 codec in 20ms packetisation |
| 12:08.41 | qu1ckkkk | many |
| 12:10.03 | zmitya | qu1ckkkk: in numbers please ? |
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| 12:11.24 | qu1ckkkk | Read : http://www.voip-info.org/wiki/view/Asterisk+dimensioning |
| 12:11.57 | zmitya | qu1ckkkk: thanks |
| 12:12.11 | qu1ckkkk | zmitya: np |
| 12:12.29 | pawpro | Hi, what is the preferred way in SIP to let the destination asterisk/applience know what number did the caller dial (DNID/Access Number). Now SIP header to naturally shows username@host |
| 12:12.59 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:14.49 | qu1ckkkk | pawpro: not sure if i understand correctly but try ${EXTEN} |
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| 12:17.15 | pawpro | qu1ckkkk: the call comes to asterisk with username (in the "to" field) the call should be sent to. This username is a sip trunk peer. but the call comes from pstn and i want the the sip trunk end point to know that the call originated through certain DNID number |
| 12:18.53 | [TK]D-Fender | pawpro: "core show function SIP_HEADER" <- parse out the "To:" |
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| 12:22.15 | pawpro | [TK]D-Fender: I know what the to is but my customer does not get the DNID from my only his own username in the TO field. I can set a custom header but want to find how others do it |
| 12:22.57 | pawpro | [TK]D-Fender: Sorry, he does not get the DNID from me only the username |
| 12:23.36 | [TK]D-Fender | pawpro: pastebin SIP DEBUG of a call... your wording is gettin worse... |
| 12:23.40 | [TK]D-Fender | ~pb |
| 12:23.41 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 12:26.58 | *** join/#asterisk robl^laptop (n=robl@m4c5336d0.tmodns.net) |
| 12:29.10 | pawpro | [TK]D-Fender: http://pastebin.com/d5176b8a1 |
| 12:29.20 | qu1ckkkk | lol @[TK]D-Fender |
| 12:31.06 | drcarumas | kaldemar, hi again. Just to say thanks. It's working with EVAL. :) |
| 12:33.54 | pawpro | qu1ckkkk: In the SIP proxy I would change the To header and send it to the right IP but i don't know how to do it in asterisk where the to is the user. |
| 12:36.24 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 12:36.27 | [TK]D-Fender | pawpro: what is the real # we should be looking at in there? |
| 12:36.50 | *** join/#asterisk Pazzo (n=ugelt@reserved-225136.rol.raiffeisen.net) |
| 12:39.54 | pawpro | [TK]D-Fender: The number isn't there because this is HIS SIP debug. Ideally instead of sending him his username in "To:" I would like to send him the string (I.E.0800xxxxxxx) which I find during processing the dialplan (from the custom header X-Access-Number). |
| 12:40.22 | *** join/#asterisk lftsy (n=lftsy@88.191.80.8) |
| 12:43.09 | *** join/#asterisk DelphiWorld (n=Miranda@196.20.95.121) |
| 12:43.40 | DelphiWorld | asterisk do SCTP? |
| 12:45.07 | *** join/#asterisk retentiveboy (n=pdugas@72.54.144.26) |
| 12:46.54 | *** join/#asterisk chendy (n=chatzill@58.250.10.224) |
| 12:47.12 | [TK]D-Fender | pawpro: "core show application SIPAddHeader" |
| 12:48.02 | [TK]D-Fender | DelphiWorld: Nope |
| 12:48.17 | pawpro | [TK]D-Fender: Again my original question was that I know how to add a custom header what I was asking about was the common header name or other technique |
| 12:48.50 | *** join/#asterisk chendy (n=chatzill@58.251.103.163) |
| 12:49.10 | [TK]D-Fender | pawpro: You normally never have to do this. |
| 12:49.39 | [TK]D-Fender | pawpro: Remote-Party-ID: "07849742558" <sip:07849742558@212.6.11.84>;privacy=off;screen=no |
| 12:50.02 | [TK]D-Fender | pawpro: Here we send the callerID, and I am not seeing what YOU are sending on your side |
| 12:50.10 | [TK]D-Fender | pawpro: the dial issued |
| 12:50.16 | [TK]D-Fender | pawpro: provide debug from the OTHER side |
| 12:52.29 | pawpro | [TK]D-Fender: There is 4 asterisks involved and it would take to much time as they are all busy and the debug would be messy. On one end you have PSTN GW on the other end you have SBC/Sip Gateway asterisk and i need to send through the original DID (I already have it on the SIP GW I'm just not sure where to stick it to)/. |
| 12:52.34 | [TK]D-Fender | pawpro: Well you'd need to isolate the one that had more info but didn't pass it |
| 12:52.45 | [TK]D-Fender | pawpro: saying "looking is hard" doesn't really cut it... |
| 12:53.42 | pawpro | [TK]D-Fender: I pass it all the way in the custom header X-Access-Number but before it is sent to the customer I wanted to standarize it or maybe overvrite To so it will work regardless whether it's asterisk or not |
| 12:54.57 | [TK]D-Fender | pawpro: do they register to the server this debug came from? |
| 12:56.00 | pawpro | [TK]D-Fender: no this is what the user gets on his box outside of my system |
| 12:56.27 | *** part/#asterisk DelphiWorld (n=Miranda@196.20.95.121) |
| 12:57.53 | [TK]D-Fender | pawpro: PB the peer you dialed for that call. |
| 12:59.00 | *** join/#asterisk wakdep (n=wakdep@81.134.137.201) |
| 12:59.27 | wakdep | Hi All - I have an issue compiling Asterisk 1.6.1.6 on Debian Lenny. I have the build deps installed but I am getting the following error: |
| 12:59.39 | wakdep | <PROTECTED> |
| 12:59.39 | wakdep | alaw.c: In function ‘ast_alaw_init’: |
| 12:59.39 | wakdep | alaw.c:186: warning: implicit declaration of function ‘ast_log’ |
| 12:59.39 | wakdep | alaw.c:186: error: ‘LOG_WARNING’ undeclared (first use in this function) |
| 12:59.39 | wakdep | alaw.c:186: error: (Each undeclared identifier is reported only once |
| 12:59.40 | wakdep | alaw.c:186: error: for each function it appears in.) |
| 12:59.44 | wakdep | alaw.c:190: error: ‘LOG_NOTICE’ undeclared (first use in this function) |
| 12:59.55 | wakdep | Would anyone have come across this before and know how to fix please? |
| 13:00.04 | [TK]D-Fender | wakdep: PASteBIN. |
| 13:00.09 | [TK]D-Fender | wakdep: Do NOT spam in here |
| 13:00.11 | [TK]D-Fender | ~pb |
| 13:00.12 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 13:00.13 | [TK]D-Fender | ^^^^^^^^^^^ |
| 13:00.19 | wakdep | Ah - sorry chaps. |
| 13:00.29 | qu1ckkkk | PASTEBIN dude |
| 13:02.29 | wakdep | Sorry - http://paste.lisp.org/+1WGT |
| 13:03.47 | [TK]D-Fender | wakdep: I see this with 1.6.1.4.... |
| 13:03.52 | [TK]D-Fender | wakdep: But not yours. |
| 13:04.01 | [TK]D-Fender | https://issues.asterisk.org/view.php?id=15697 |
| 13:04.53 | wakdep | so this would be with the new G711 Algorithm then? I'll try without too... |
| 13:05.46 | wakdep | it almost looks like a header file missing.... *ponders* |
| 13:08.55 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 13:09.33 | *** join/#asterisk DelphiWorld (n=Miranda@196.20.95.121) |
| 13:09.45 | DelphiWorld | hapy diwali to all person from india;) |
| 13:11.17 | *** join/#asterisk seanmh (n=johndoe@69.254.131.168) |
| 13:13.36 | *** join/#asterisk knobo (n=user@90.149.4.182) |
| 13:15.43 | wakdep | Can I ask where I search for issues? I seem to be having an dissue with the make target for /tests/modules.link now! :-) |
| 13:17.19 | *** join/#asterisk devyll (n=paul@89.36.24.2) |
| 13:17.42 | *** part/#asterisk DelphiWorld (n=Miranda@196.20.95.121) |
| 13:18.29 | [TK]D-Fender | wakdep: [09:03]<[TK]D-Fender>https://issues.asterisk.org/view.php?id=15697 <-- good place to start... |
| 13:18.33 | [TK]D-Fender | wakdep: repoen it |
| 13:18.37 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:18.39 | devyll | hello. How can I make the Read command to hide the string entered by the user ? I don't want that string to appear in the loggs. I'm trying to build a password authentification dialplan. |
| 13:19.27 | [TK]D-Fender | devyll: Do you actually see it? |
| 13:25.30 | *** join/#asterisk baijum (n=baiju@122.166.46.113) |
| 13:32.28 | *** join/#asterisk errr (n=errr@fedora/errr) |
| 13:38.35 | *** join/#asterisk ManxPower-work (n=EWieling@24.42.221.26) |
| 13:38.43 | *** join/#asterisk slinksh0t (n=slinksh0@98.64.206.62) |
| 13:38.45 | ManxPower-work | ~answers |
| 13:38.46 | infobot | rumour has it, answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 13:39.23 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 13:40.09 | [TK]D-Fender | Wow... somone seriously tried pounding my home server for SIP peers last month... |
| 13:41.44 | fiddur | i've seen that too... I reported it to the isp of that ip |
| 13:42.40 | *** join/#asterisk cusco (n=trilili@2001:0:53aa:64c:1864:cb77:2ac0:762d) |
| 13:42.43 | cusco | hi |
| 13:43.21 | cusco | our digium card has two primary sockets |
| 13:43.46 | cusco | 02:01.0 Communication controller: Digium, Inc. Wildcard TE210P dual-span T1/E1/J1 card 3.3V (rev 02) |
| 13:43.46 | cusco | we have all 30 channekls gull, Im about to connect the seccond line in |
| 13:43.55 | cusco | what has to be done on /etc/asterisk/dahdi-channels.conf ? |
| 13:44.04 | cusco | at the moment it reads: |
| 13:44.20 | ManxPower-work | use pastebin! |
| 13:44.22 | ManxPower-work | ~pb |
| 13:44.23 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 13:44.28 | qu1ckkkk | pb pls |
| 13:44.29 | qu1ckkkk | ! |
| 13:44.29 | cusco | http://paste.debian.net/49250/ |
| 13:44.42 | cusco | at the moment it reads: http://paste.debian.net/49250/ |
| 13:45.08 | cusco | it should be ok to plug it in, right? |
| 13:45.13 | cusco | what else needs to be done? |
| 13:45.57 | ManxPower-work | remember, any options set AFTER a channel => line does not get applied |
| 13:45.59 | cusco | how do I activate it |
| 13:46.10 | ManxPower-work | cusco: jut plug it in. |
| 13:46.12 | cusco | so how do I reload? |
| 13:46.42 | ManxPower-work | I assume that 2nd span is also going to the telco, right? |
| 13:48.46 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
| 13:49.26 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:52.27 | *** join/#asterisk zamba (i=marius@flage.org) |
| 13:52.36 | zamba | how do i deactivate the skinny protocol from my *? |
| 13:53.11 | [TK]D-Fender | zamba: noload =>chan_skinny.so |
| 13:53.15 | [TK]D-Fender | zamba: in modules.conf |
| 13:53.48 | zamba | [TK]D-Fender: thanks :) |
| 13:54.13 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 13:54.35 | *** join/#asterisk wonderworld (n=w@62.143.22.226) |
| 13:58.23 | *** join/#asterisk came0 (n=came0@71.42.53.159) |
| 13:59.16 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 13:59.32 | *** part/#asterisk fiddur (n=fiddur@192.121.104.118) |
| 14:00.39 | afink | if I set jbenable=yes and jbforce=yes in sip.conf will all my sip calls be jitter buffered? |
| 14:01.20 | *** join/#asterisk came0_ (n=came0@rrcs-71-42-25-233.se.biz.rr.com) |
| 14:01.26 | afink | or do I need the j option in a dial? |
| 14:02.30 | *** join/#asterisk Naikrovek (n=jjohnson@unaffiliated/naikrovek) |
| 14:03.06 | russellb | afink: if you set it on a sip channel, it will almost never be used |
| 14:03.37 | TSM2 | ok this is polycom related, is there a way to limit certian parameters in the xml conf only for certian models of phone? |
| 14:04.34 | russellb | http://blogs.asterisk.org/2007/02/28/asterisk-1-4-jitterbuffer/ |
| 14:04.38 | Naikrovek | TSM2: yes, but i just woke up and the method isn't revealing itself to me right now |
| 14:05.46 | afink | russellb: thank you |
| 14:06.21 | Naikrovek | TSM2: how many phones do you have in total, and how many models are you using |
| 14:07.15 | [TK]D-Fender | TSM2: like what/ |
| 14:08.39 | TSM2 | [TK]D-Fender: the ip330 and ip450+ have diffrent ammount of softkeys, i want to show diffrent buttons depending on the phone, i know one way of doing it which is including a diffrent config file in the XML file for the phones to load up, but was wandering if it can be done within a single XML file |
| 14:09.13 | *** join/#asterisk shinao1 (n=shinao1@41.219.251.97) |
| 14:09.31 | *** join/#asterisk crazybyte (n=crzp@unaffiliated/crazypenguin/x-000001) |
| 14:09.47 | [TK]D-Fender | TSM2: What file are those in? |
| 14:10.34 | cusco | hi |
| 14:10.46 | cusco | second primary cable is fisically pluggged in |
| 14:10.49 | cusco | how to activate it?» |
| 14:11.03 | cusco | ManxPower-work: yes |
| 14:11.21 | ManxPower-work | cusco: you do not "activate it" You plug it in and it works. |
| 14:11.47 | Naikrovek | i thought he meant "fiscally" for a minute there. |
| 14:11.56 | crazybyte | hi. would somebody be so kind and enlighten me (if he or she can) about the purpose of the following ports used by asterisk http://pastie.org/657552 thank you! |
| 14:11.56 | cusco | ManxPower-work: no |
| 14:11.58 | ManxPower-work | You won't need a T-1/E-1 crossover since you are going from the 2nd port to the telco,. IF you were using the 2nd port to go to a pbx then that would be totally different. |
| 14:11.59 | cusco | pri show span 2 |
| 14:12.04 | cusco | does not show it working |
| 14:12.10 | TSM2 | [TK]D-Fender: first file the phone tries to download is {MAC ADDR}.cfg, in that file is a single line telliing the phone where the main application is, then it tells in what order to load extra config files etc..., currently for me it loads a generic default sip.cfg file to reset the phone, one to define the standard server setup and one to define the phones exention |
| 14:12.19 | ManxPower-work | cusco: PASTEBIN the output |
| 14:12.57 | [TK]D-Fender | TSM2: well you could do a complete single unique file per phone if you really wanted to... |
| 14:13.17 | ManxPower-work | crazybyte: 4569 is IAX2, I assume the others are MGCP and SCCP |
| 14:13.23 | crazybyte | i see |
| 14:13.26 | crazybyte | thx |
| 14:13.56 | crazybyte | i need to put up a firewall and i didn't know what those ports are used for. thank you |
| 14:13.57 | TSM2 | [TK]D-Fender: what i thought was include a seperate file, an adjustment file for softkeys, there are a bunch of standard parameters you can use, phone MAC code, phone model etc... bingo, just thought how i would do it |
| 14:14.08 | ManxPower-work | crazybyte: are you using SIP? |
| 14:14.16 | crazybyte | yes |
| 14:14.18 | crazybyte | only sip |
| 14:14.21 | [TK]D-Fender | TSM2: As long as the bits add up you should be fine |
| 14:14.38 | crazybyte | ManxPower-work, only sip so i don't need to open those ports |
| 14:14.42 | ManxPower-work | then you need ports 5060/UDP and whatever ports are in /etc/asterisk/rtp.conf IF you don't have an rtp.conf then the ports default to 10000/UDP - 20000/UDP |
| 14:14.57 | ManxPower-work | ~sipnat |
| 14:14.57 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 14:14.59 | Naikrovek | TSM2: i think the indended way to do this is to create X number of config files containing the different softbutton configs and add the appropriate one to each macaddress.cfg files |
| 14:15.02 | [TK]D-Fender | crazybyte: MGCP, IAX2, DUNDI, and I don't know 5000 |
| 14:15.08 | crazybyte | ManxPower-work, yes. those are open |
| 14:15.23 | TSM2 | [TK]D-Fender: you might know this. is there any reason why when parking a call it does not automaticly hang up after its done? |
| 14:15.24 | cusco | ManxPower-work: sorry yes let me show tou |
| 14:15.37 | [TK]D-Fender | crazybyte: So NOLOAd those modules in modules.conf |
| 14:15.51 | Naikrovek | TSM2: so if you have 2 phone models, create 2 per-model configs, and include the appropriate one in each macaddress.cfg |
| 14:16.04 | crazybyte | [TK]D-Fender, i will do that also |
| 14:16.04 | crazybyte | thx |
| 14:16.04 | cusco | ManxPower-work: No PRI running on span 2 |
| 14:16.04 | Naikrovek | i think TSM2 has me ignored |
| 14:16.04 | Naikrovek | sigh |
| 14:16.06 | [TK]D-Fender | TSM2: it is an attended transfer. You need to release on your side. its no different |
| 14:16.11 | TSM2 | Naikrovek: just realised the main config can accept [PHONE_MODEL] |
| 14:16.26 | cusco | we are having a GREAT deal of MASS calls |
| 14:16.29 | ManxPower-work | cusco: what is the output of ztcfg -vvv or dahdi_cfg -vvv pastebin the info |
| 14:16.40 | Naikrovek | TSM2: yes, create each model.cfg then include them via that |
| 14:16.41 | cusco | dahdi calls go down when channels reach 30/31 |
| 14:17.14 | ManxPower-work | cusco: are you using zaptel or dahdi? |
| 14:17.41 | cusco | http://pastebin.com/f39d0a92c |
| 14:17.43 | cusco | dahdi |
| 14:18.11 | cusco | 31 first channels seem to be OK |
| 14:18.23 | ManxPower-work | cusco: you cannot add/remove channels from the config while active calls are happening. pastebin the output of dahdi show channels |
| 14:18.23 | cusco | but we are having more thatn 31 calls from dahdi |
| 14:18.29 | TSM2 | [TK]D-Fender: so basicly no way to make it auto hangup, thats annoying, ile program that into the efk stuff on the polycoms |
| 14:19.00 | cusco | you mean in asterisk cli: core show channels |
| 14:19.00 | cusco | ? |
| 14:19.10 | ManxPower-work | cusco: no, I mean dahdi show channels |
| 14:19.29 | ManxPower-work | core show channels only shows channels with active calls. |
| 14:19.57 | cusco | http://paste.debian.net/49261/ |
| 14:20.00 | cusco | ok |
| 14:20.12 | ManxPower-work | cusco: you did not restart asterisk |
| 14:20.33 | cusco | I did after plugging the second span |
| 14:20.36 | cusco | fisically |
| 14:20.46 | cusco | but we are having MASSIVE calls!!1 |
| 14:20.49 | ManxPower-work | you have to restart asterisk when you add/remove channels. You could instead use "module unload chan_dahdi.so" and then "load chan_dahdi.so |
| 14:21.05 | cusco | ahh I did not restart dahdi |
| 14:21.12 | ManxPower-work | cusco: then do this when you do not have massive calls. |
| 14:21.29 | ManxPower-work | running the dahdi_cfg -vvv restarts dahdi |
| 14:22.09 | cusco | ManxPower-work: ok I will waint until I have less calls |
| 14:22.19 | cusco | or asterisk breaks again (calls going down) |
| 14:22.25 | cusco | mean while enlighten me |
| 14:22.37 | cusco | just restart dahdi (reloading the module) will activate the second span |
| 14:22.45 | cusco | is /etc/asterisk7dahdi-channels.conf |
| 14:22.47 | cusco | ok? |
| 14:22.54 | cusco | is /etc/asterisk/dahdi-channels.conf OK? |
| 14:23.33 | *** join/#asterisk moy (n=chatzill@74.12.134.3) |
| 14:23.50 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 14:25.17 | cusco | ManxPower-work: http://paste.debian.net/49250/ |
| 14:25.37 | cusco | that should be OK? I mean, is it even not suposed to be edited by us? |
| 14:26.05 | ManxPower-work | cusco: unless you are using some stupid gui all files should be edited by us. |
| 14:27.00 | cusco | no Im not using anything, it just soudned like that file was generated (it has a comment on red alarm) |
| 14:27.10 | cusco | but it seems to be ok, it start on chan 32 |
| 14:27.49 | *** part/#asterisk sircolin (n=sircolin@my83-216-68-241.mynow.co.uk) |
| 14:29.27 | ManxPower-work | cusco: don't expect it to work with just a dahdi reload. I already said that. |
| 14:34.01 | cusco | ManxPower-work: you said that I could unload and load chan_dahdi.so |
| 14:34.06 | cusco | instead of restarting asterisk |
| 14:34.20 | ManxPower-work | cusco: Correct. But "dahdi reload" is a totally different command. |
| 14:34.33 | ManxPower-work | load/unload is not a reload |
| 14:34.41 | cusco | when I sauid reload dahdi, I was thinking of rmmod && modprobe |
| 14:34.55 | cusco | ok |
| 14:35.04 | cusco | I will unload it and load it via asteirks cli |
| 14:35.43 | cusco | just wait for the right moment |
| 14:37.47 | *** join/#asterisk came0 (n=came0@rrcs-71-42-25-233.se.biz.rr.com) |
| 14:37.52 | ManxPower-work | actually rmmod and modprobe is also not a "dahdi reload" |
| 14:38.11 | cusco | hmm |
| 14:38.17 | cusco | ok... |
| 14:38.31 | cusco | I said reload because /etc/init.d/dahdi restart works that way |
| 14:38.41 | ManxPower-work | When I said "dahdi reload" I meant exactly that. Issuing the command "dahdi reload" in the CLI. |
| 14:39.00 | ManxPower-work | Issuing that command don't add/remove channels. Other stuff will add/remove channels |
| 14:39.08 | cusco | yes ok |
| 14:39.14 | cusco | thanks |
| 14:39.18 | cusco | thansk for helping dude |
| 14:39.36 | cusco | I will unload chan_dahdi.so and load chan_dahdi.so |
| 14:39.42 | cusco | do i have to restart asterisk afterwards? |
| 14:40.03 | ManxPower-work | no, doing those two commands should be enough for Asterisk to see the new channels. |
| 14:40.23 | ManxPower-work | you already took care of the dahdi_cfg earlier |
| 14:40.31 | cusco | cheers |
| 14:40.33 | cusco | ok I only have to wait know.. we are having some calls |
| 14:41.40 | *** join/#asterisk fofware (n=chatzill@190.7.25.160) |
| 14:45.57 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 14:47.04 | *** join/#asterisk G2k (n=1@12.195.18.127) |
| 14:47.23 | cusco | thanks a lot ManxPower-work |
| 14:47.30 | cusco | another question... |
| 14:47.46 | cusco | I heard that the 31st channel is suposed to be for internal monitoring |
| 14:47.53 | cusco | not a actual channel |
| 14:47.56 | cusco | is that true? |
| 14:48.10 | cusco | I saw 30 dahdis comming in, then the 31st came, all calls went down |
| 14:48.17 | cusco | Im still going trough log file |
| 14:50.55 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 14:51.01 | *** join/#asterisk shinao1 (n=shinao1@41.219.249.49) |
| 14:51.34 | cusco | bloody U2, they still a month and half away and people start calling to get their tickets early |
| 14:55.53 | *** join/#asterisk came0 (n=came0@rrcs-71-42-25-233.se.biz.rr.com) |
| 15:01.35 | *** join/#asterisk Firass-z0r (n=asadf@c-67-201-205-34.reshall.wwu.edu) |
| 15:04.59 | *** join/#asterisk DavidR2008 (n=chatzill@208.34.240.2) |
| 15:07.24 | *** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:25fd:d9bd:47bd:158) |
| 15:10.02 | *** join/#asterisk grharry (n=root@ppp-94-65-225-129.home.otenet.gr) |
| 15:10.55 | devyll | [TK]D-Fender: I asked you a question about making Read not display the "password" entered by the caller in the log file or in the CLI when enabling verbose. You asked me if I actually see it. and I had to leave so I'm answering now ... Yes I can see it in the CLI . sorry for the daly |
| 15:11.23 | devyll | actually the question was asked here in the channel .. not to you personally but you answered. |
| 15:12.29 | grharry | hi, I am builing asterisk with mISDN from mISDN.org .. Do I need to build install kernel modules etc for dahdi at all ??? thnx in advance ... |
| 15:12.49 | [TK]D-Fender | devyll: Whoever has access to CLI can do FAR more damage directly. |
| 15:13.11 | [TK]D-Fender | devyll: You don't hand out the keys to your house and the alarm code and then wonder how to secure yourself. |
| 15:14.11 | *** join/#asterisk jlnt (n=jlnt@cisco2.jlmail.com) |
| 15:15.03 | devyll | [TK]D-Fender: you are right. |
| 15:15.54 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 15:16.28 | *** join/#asterisk leif[mobile] (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:16.28 | *** mode/#asterisk [+o leif[mobile]] by ChanServ |
| 15:17.56 | *** join/#asterisk fofware (n=chatzill@190.7.25.160) |
| 15:19.51 | *** join/#asterisk GameGamer43 (n=GameGame@69.129.142.83) |
| 15:20.17 | grharry | anybody ?? |
| 15:20.23 | grharry | :-( |
| 15:21.49 | *** join/#asterisk Defraz (n=Defraz@corp.fuzecore.com) |
| 15:22.33 | leif[mobile] | heyo |
| 15:27.52 | grharry | hi, I am builing asterisk with mISDN from mISDN.org .. Do I need to build install kernel modules etc for dahdi at all ??? thnx in advance ... |
| 15:30.47 | *** join/#asterisk raden (n=chatzill@66-168-15-19.dhcp.stpt.wi.charter.com) |
| 15:30.48 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
| 15:32.35 | *** join/#asterisk edwin_quijada (n=macaruch@200.26.172.50) |
| 15:33.59 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 15:35.32 | *** join/#asterisk alexshell (n=abc@unaffiliated/alexshell) |
| 15:43.19 | *** join/#asterisk RypPn (i=TuMbL@rosscom.co.uk) |
| 15:46.30 | *** join/#asterisk joshaidan (n=brianj@S0106001c1023e838.tb.shawcable.net) |
| 15:47.42 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 15:54.20 | grharry | thanks all !!!!! Much appreciated ... you all are such a gurus !!!!! |
| 15:54.27 | *** join/#asterisk Sajam (n=sajam@beta.intelligile.com) |
| 15:54.45 | *** part/#asterisk grharry (n=root@ppp-94-65-225-129.home.otenet.gr) |
| 15:58.20 | Sajam | Hello, i want to configure Qutecom to connect to asterisk, what should i enter in the account of Qutecom?? |
| 15:59.53 | *** join/#asterisk asterwiki (n=asterwik@69.77.169.14) |
| 16:00.41 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 16:00.43 | [TK]D-Fender | Sajam: user, pass, server. 3 little blanks to fill.. |
| 16:01.41 | Sajam | [TK]D-Fender, Thank you very much very your reply, actually i am having problems also after reading the bbok |
| 16:02.07 | Sajam | [TK]D-Fender, my question is user and pass shall be created on asterisk yeah?? |
| 16:02.33 | [TK]D-Fender | Sajam: Huh? |
| 16:02.37 | Sajam | like user= john pass=password ?? |
| 16:02.56 | [TK]D-Fender | Sajam: incorrect syntax, but the right idea |
| 16:03.01 | Sajam | okay, i installed astersik on Centos, so i need to create user and pass |
| 16:03.27 | Sajam | mmm, what is the right syntax |
| 16:03.33 | [TK]D-Fender | Sajam: Keep reading the book... |
| 16:03.53 | KavanS | http://www.voipusersconference.org/wp-content/uploads/2009/10/allison.jpg |
| 16:03.58 | KavanS | heh that is an interesting picture! |
| 16:04.03 | [TK]D-Fender | Sajam: And for a slightly old but still good example of a simple system : |
| 16:04.07 | [TK]D-Fender | ~jerjerguide |
| 16:04.08 | infobot | [~jerjerguide ] Jeremy McNamara's quick sample guide to setting up * : http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/ |
| 16:04.14 | KavanS | [TK]D-Fender, you at astricon? |
| 16:04.27 | *** join/#asterisk sircolin (n=sircolin@my83-216-68-241.mynow.co.uk) |
| 16:04.30 | [TK]D-Fender | KavanS: Never been... |
| 16:15.40 | Kobaz | okay.... sooooo |
| 16:15.48 | *** join/#asterisk blkry (n=chatzill@63-255-103-4.ip.mcleodusa.net) |
| 16:15.53 | *** join/#asterisk raden (n=chatzill@66-168-15-19.dhcp.stpt.wi.charter.com) |
| 16:16.16 | raden | Sandheaver: how goes it bro |
| 16:16.26 | Kobaz | i have an asteirsk box in front of an avaya... on t1... and then i have a t1 to the network (pstn).... i take a call from the avaya, and send it to the pstn over t1... it works fine |
| 16:16.46 | Kobaz | i dial out directly from a soft phone to the network, and the call is rejected |
| 16:17.06 | Kobaz | now i figure it's a callerid problem, but my callerid fields match exactly from what the avaya is sending... (i just have a passthrough dial) |
| 16:17.12 | Kobaz | so here's the debuggerades |
| 16:17.32 | Sajam | [TK]D-Fender, i just wanna ask if i can use cisco router 1760 connected to astersik on lan, instead of having tdm card |
| 16:17.54 | [TK]D-Fender | Sajam: Sure.. |
| 16:19.46 | Kobaz | http://pastebin.ca/1623864 <-- working call |
| 16:20.15 | Sajam | [TK]D-Fender, thank you very much, i will ask about the configuration with cisco later on :P |
| 16:20.17 | Kobaz | http://pastebin.ca/1623865 <-- not working call |
| 16:21.57 | Kobaz | all the callerid fields are the same |
| 16:22.19 | Katty | hmm. |
| 16:22.24 | Katty | i has been awoken. |
| 16:22.34 | Katty | mayhaps i should go to work today. |
| 16:22.43 | hardwire | I haz been amazed |
| 16:23.02 | [TK]D-Fender | <PROTECTED> |
| 16:23.12 | [TK]D-Fender | <PROTECTED> |
| 16:23.16 | Kobaz | it was a search and replace |
| 16:23.17 | [TK]D-Fender | ^^^^^^ |
| 16:23.24 | Kobaz | i didn't search and replace the other one |
| 16:23.30 | [TK]D-Fender | Kobaz: not the number... the pRESENTATION |
| 16:23.44 | *** join/#asterisk xpot-mobile (n=james@mx0.synergyconsultant.net) |
| 16:23.45 | [TK]D-Fender | Kobaz: SetCallerPres() |
| 16:23.48 | Kobaz | okay |
| 16:24.20 | [TK]D-Fender | Kobaz: Also : Good : > Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: 3.1kHz audio (16) |
| 16:24.33 | [TK]D-Fender | Kobaz: Also : Bad : Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) |
| 16:24.39 | [TK]D-Fender | Odd difference |
| 16:25.27 | Kobaz | wow |
| 16:25.50 | Kobaz | that worked |
| 16:25.50 | Kobaz | doing setcallerpres |
| 16:27.07 | Kobaz | so why would i have to set callerpres on dialing from an iax peer, and not when taking a call from another pri link |
| 16:28.15 | [TK]D-Fender | Kobaz: Privacy settings based on the channel of origin |
| 16:28.19 | Kobaz | ah |
| 16:29.04 | Kobaz | how long have you been monkeying with asterisk/telephony btw |
| 16:29.12 | Kobaz | there doesn't seem to be many things you can't fix |
| 16:29.12 | Kobaz | heh |
| 16:32.25 | [TK]D-Fender | Kobaz: Between 5-6 years with *, telephony goes back to the 300 baud modem days and phreaking with DTMF dialers and wire cutters at phone booths ;) |
| 16:32.33 | carrar | [TK]D-Fender sleeps on a bed made of Asterisk boxes |
| 16:32.37 | *** part/#asterisk sircolin (n=sircolin@my83-216-68-241.mynow.co.uk) |
| 16:32.48 | [TK]D-Fender | also has back issues... |
| 16:32.52 | Kobaz | aww |
| 16:35.56 | [TK]D-Fender | needs softer servers |
| 16:36.00 | Kobaz | i'm like htat with linux |
| 16:36.04 | Kobaz | *that |
| 16:36.13 | Kobaz | i've been using linux since 94 |
| 16:36.18 | Kobaz | there's not much I can't fix, heh |
| 16:38.15 | [TK]D-Fender | I only started using Linux around 2003/4 |
| 16:38.30 | [TK]D-Fender | And I'm far from "skilled" |
| 16:43.21 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 17:03.34 | *** join/#asterisk denon (i=denon@sassinak.net) |
| 17:03.34 | *** mode/#asterisk [+o denon] by ChanServ |
| 17:03.51 | cusco | help |
| 17:04.03 | cusco | http://paste.debian.net/49282/ |
| 17:04.11 | cusco | no span| |
| 17:05.16 | Kobaz | halp |
| 17:05.24 | Kobaz | help the computer |
| 17:05.29 | Kobaz | i am the computer |
| 17:05.37 | Kobaz | cusco: is dahdi loaded? |
| 17:07.47 | cusco | ok its solved. I will need help and I will explain, hold on a sec please |
| 17:08.17 | [TK]D-Fender | if its solved... why do you need "help"? |
| 17:08.54 | Kobaz | heh |
| 17:11.04 | *** join/#asterisk guaxinim (n=guaxinim@unaffiliated/guaxinim) |
| 17:12.12 | p3nguin | Didn't think that one through all the way, probably. |
| 17:13.18 | carrar | w00t PostgreSQL conf starts today here in Seattle! |
| 17:14.34 | Katty | :>>> |
| 17:16.59 | [TK]D-Fender | p3nguin: You had him ... at "didn't think" :p |
| 17:19.08 | KavanS | [TK]D-Fender, tried to use the AMD thing....I ended up just scripting an escape routing for the providers voicemail |
| 17:19.22 | KavanS | so far it is pretty reliable... |
| 17:20.23 | [TK]D-Fender | KavanS: "escape routing"? |
| 17:20.49 | KavanS | [TK]D-Fender, well...just sent DTMF to escape out of the voicemail...not really a defacto term |
| 17:20.59 | Kobaz | you always need a planned escape |
| 17:21.04 | KavanS | I got tired of the "press 1 to accept" being left on cell voicemail |
| 17:21.05 | Kobaz | never know when their gonna come to get you |
| 17:21.12 | KavanS | Kobaz, they are already here! |
| 17:21.21 | KavanS | no one had a solid solution to prevent this... |
| 17:21.45 | [TK]D-Fender | KavanS: Have them embed DTMF into their VM message so You can tell * to look for it :) |
| 17:21.55 | Kobaz | heh |
| 17:22.13 | [TK]D-Fender | it IS doable.... retarded.. but doable |
| 17:22.16 | KavanS | [TK]D-Fender, how can you get * to look for DTMF? |
| 17:22.26 | KavanS | ha, why would you call it retarded? |
| 17:22.36 | KavanS | my guys were getting literally 3-4 notifications that a call came in... |
| 17:22.46 | KavanS | add a cell voicemail to it, it's almost overwhelming |
| 17:22.56 | p3nguin | Couldn't you just set the timeout lower so that cell voicemail won't pick up? |
| 17:23.13 | KavanS | p3nguin, what happens when the phone is turned off? ;) |
| 17:23.18 | [TK]D-Fender | KavanS: use M() like you were before and fire off a read with timeout... |
| 17:23.29 | KavanS | p3nguin, this solution works no matter if you press reject, answer or have your phone off |
| 17:23.41 | p3nguin | kavans: not sure, that's why I threw out the suggestion. |
| 17:23.48 | KavanS | [TK]D-Fender, ahh good point....I'm using read right now, so you are suggesting to just record a "2" or something so it disconnects immediately |
| 17:23.53 | [TK]D-Fender | p3nguin: reject on getting the DTMF within 2s of answering, bridge otherwise |
| 17:23.58 | KavanS | [TK]D-Fender, also clever |
| 17:24.03 | [TK]D-Fender | <- SMRT |
| 17:24.07 | KavanS | lol |
| 17:26.49 | *** join/#asterisk CcRnp (n=shishir@208.179.165.18) |
| 17:28.54 | Katty | eppigy: :> |
| 17:31.06 | Katty | eppigy: HELLO DAVE |
| 17:31.12 | cusco | Kobaz: regarding that problem, for it to work I had to comment out the 2nd span in /etc/asterisk/dahdi-channels.conf - it now looks like: http://paste.debian.net/49290/ |
| 17:31.29 | cusco | how should I configure the 2nd span in /etc/asterisk/dahdi-channels.conf /etc/asterisk/dahdi-channels.conf |
| 17:31.33 | cusco | ? |
| 17:31.46 | Kobaz | the same way that you configure the first span |
| 17:31.50 | Kobaz | starting at the next channel |
| 17:32.04 | Kobaz | so if you end on channel 24... channel 25 is the first channel on span 2 |
| 17:32.06 | cusco | but that is already there! |
| 17:32.15 | cusco | it ended on 31, and it should go on on 32 |
| 17:32.21 | Kobaz | yeah |
| 17:32.22 | cusco | look at the pastebin! |
| 17:32.31 | Kobaz | i'm busy playing castle age |
| 17:32.37 | cusco | so I had no command pri in asterisk |
| 17:32.38 | Kobaz | need to stash my gold |
| 17:32.45 | cusco | until I commented out the 2nd span |
| 17:32.55 | cusco | I understand, I have a script playing travin for me atm |
| 17:32.56 | cusco | :p |
| 17:32.59 | cusco | travian |
| 17:33.00 | Kobaz | heh |
| 17:33.44 | Kobaz | okay |
| 17:33.47 | Kobaz | looks good |
| 17:33.54 | Kobaz | is dahdi configured with those channels |
| 17:33.58 | Kobaz | paste your dahdi/system.conf |
| 17:33.59 | cusco | err |
| 17:34.01 | cusco | hold |
| 17:34.43 | cusco | http://paste.debian.net/49291/ |
| 17:35.52 | Kobaz | and what's dahdi_cfg -vv say |
| 17:36.43 | cusco | http://paste.debian.net/49293/ |
| 17:36.49 | cusco | all calls went down now, when it reaches channel 31 |
| 17:37.01 | cusco | somebody said that channel 31 should be used internally for monitoring only |
| 17:37.09 | *** join/#asterisk DelphiWorld (n=Miranda@196.20.95.121) |
| 17:37.09 | DelphiWorld | hi |
| 17:37.13 | DelphiWorld | asterisk support the celt codec? |
| 17:37.15 | Kobaz | this is a live system? |
| 17:37.18 | cusco | yes |
| 17:37.20 | Kobaz | heh |
| 17:37.24 | cusco | we really need the 2nd span |
| 17:37.26 | Kobaz | yeah dahdi_cfg may reset your calls |
| 17:37.29 | [TK]D-Fender | DelphiWorld: G.722 |
| 17:37.35 | [TK]D-Fender | WILL |
| 17:37.39 | cusco | no, it was just before, Kobaz |
| 17:37.47 | cusco | :( |
| 17:37.57 | Kobaz | what's the permissions on /dev/dahdi |
| 17:37.57 | DelphiWorld | [TK]D-Fender: no, celt |
| 17:38.01 | Kobaz | and what's the error you're getting now |
| 17:38.26 | [TK]D-Fender | DelphiWorld: Do YOU see a codec for it in your install? |
| 17:38.30 | cusco | permissions are crw-rw---- 1 asterisk asterisk |
| 17:38.36 | Kobaz | for everything? |
| 17:38.36 | cusco | asterisk is running as user asterisk |
| 17:38.39 | cusco | yes |
| 17:38.40 | Kobaz | good |
| 17:38.49 | Kobaz | what error are you getting from dahdi trying to bring up the second span? |
| 17:38.52 | DelphiWorld | [TK]D-Fender: no, but mayb;) |
| 17:39.21 | *** join/#asterisk DrZeus (n=chatzill@201.226.170.106) |
| 17:39.24 | cusco | the error, i did not lookout for errors, I jsut commented the 2nd span becuase after unloading chandahdi.so |
| 17:39.37 | [TK]D-Fender | DelphiWorld: You are asking questions you already possess the answer to. You jsut aren't looking at it |
| 17:39.37 | cusco | the error, i did not lookout for errors, I jsut commented the 2nd span becuase after unloading chan_dahdi.so and loading it again |
| 17:39.42 | cusco | the command pri was not available ina sterisk |
| 17:39.56 | cusco | and then it worked |
| 17:40.05 | DrZeus | hi all. Question: im trying to run the asterisk command from other user than root, and tells me that it is unable to connect to remote asterisk |
| 17:40.05 | DelphiWorld | [TK]D-Fender: hehehe |
| 17:40.07 | Kobaz | we need logs/errors/etc |
| 17:40.10 | cusco | ok |
| 17:40.11 | DrZeus | how can I give this other user access? |
| 17:40.14 | CcRnp | hey guys do you have any idea about jiaxclient a iax library for java ? |
| 17:40.27 | cusco | Kobaz: errors only or warnings as well? |
| 17:40.30 | Kobaz | there's like 20 different things can can be wrong for a pri not to go up |
| 17:40.37 | Kobaz | paste everything |
| 17:40.52 | *** part/#asterisk DelphiWorld (n=Miranda@196.20.95.121) |
| 17:41.16 | [TK]D-Fender | DrZeus: Give it permissions to the binaries and the PID file that * looks for |
| 17:41.20 | cusco | we have other pending warningss not solved, not related |
| 17:41.26 | cusco | but hold, it is a big paste |
| 17:41.29 | [TK]D-Fender | ~asterisk-non-root |
| 17:41.30 | infobot | [~asterisk-non-root] Running * as non-root is covered in chapter 13 of Asterisk:The Future of Telephony 2nd Edition (~book) and an article written by Leif Madsen at http://www.taug.ca/node/115 , and for Debian : http://www.thinkdebian.org/archives/828 |
| 17:41.33 | [TK]D-Fender | ^^^^6 |
| 17:41.59 | DrZeus | [TK]D-Fender : thank you; i will check the book then |
| 17:42.20 | cusco | Kobaz: http://paste.debian.net/49294/ |
| 17:42.37 | DrZeus | this is mostly a unix question actually, but how can I run a script redirecting the root password as the stdin? |
| 17:42.46 | cusco | utils.c error is probably not that important right now |
| 17:42.50 | DrZeus | lets say, to automate an ssh connection\ |
| 17:43.16 | cusco | DrZeus: I automate ssh connection with digital signatures/keys |
| 17:43.24 | [TK]D-Fender | DrZeus: You don't. You create a certificate for automatic permission |
| 17:43.33 | *** join/#asterisk ghento (n=ghento@user146-171.wireless.utoronto.ca) |
| 17:43.37 | [TK]D-Fender | DrZeus: and this isn't a question for this channel.... |
| 17:43.50 | DrZeus | [TK]D-Fender: i know...just took 'the risk; |
| 17:43.51 | DrZeus | :) |
| 17:44.06 | Kobaz | cusco: and that's with the span 2 uncommented, right |
| 17:44.28 | cusco | Kobaz: it was commented in the middle of that log,like 30m ago, not its 18:44 |
| 17:44.54 | cusco | can't recall when |
| 17:45.10 | cusco | 1925124 -rw-r--r-- 1 root root 792 2009-10-16 18:08 /etc/asterisk/dahdi-channels.conf |
| 17:45.13 | Kobaz | [Oct 16 18:05:11] ERROR[21877] chan_dahdi.c: Unable to open channel 1: No such device or address |
| 17:45.15 | cusco | that was last modified date,r ight? |
| 17:45.22 | Kobaz | do you have a /dev/dahdi/1 and etc |
| 17:45.30 | Kobaz | yeah that's last modified |
| 17:45.34 | cusco | yes |
| 17:45.49 | cusco | until 62 |
| 17:45.59 | *** join/#asterisk yziquel (i=53acc979@gateway/web/freenode/x-pfruvklovkomcalv) |
| 17:45.59 | cusco | channel pseudo ctl timer |
| 17:46.19 | Kobaz | what's the output of dahdi_hardware |
| 17:46.24 | ghento | Hi all. I have a system setup utilizing the out files. I'm having a problem with CallerID however. In the out file I have: "CallerID: Name <5551112222>" on a line, however when it dials out, the callerid is something different, some random number ("1238756"). I think the syntax for setting the callerid is correct? |
| 17:46.30 | cusco | pci:0000:02:01.0 wct4xxp+ d161:0210 Wildcard TE210P (3rd Gen) |
| 17:46.48 | Kobaz | and: lsmod | grep dummy |
| 17:46.52 | yziquel | hi. I get a "channel.c:2781 ast_channel_make_compatible: No path to translate from" error. What does it mean? |
| 17:47.27 | cusco | by dummy you mean ct4xxp |
| 17:47.31 | Kobaz | nope |
| 17:47.32 | Kobaz | dummy |
| 17:47.33 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 17:47.36 | cusco | wct4xxp 259712 31 |
| 17:47.36 | cusco | dahdi 221136 120 dahdi_echocan_mg2,wct4xxp |
| 17:47.39 | cusco | no dummy |
| 17:47.40 | cusco | :/ |
| 17:47.43 | Kobaz | k, that's good |
| 17:47.51 | cusco | cool |
| 17:47.54 | Kobaz | dummy is loaded when it doesn't detect a card... and will throw off channel numbering |
| 17:48.01 | cusco | ah, k |
| 17:48.15 | cusco | its live and we are having calls right now |
| 17:48.21 | cusco | so card must be detected |
| 17:48.32 | Kobaz | yeah |
| 17:48.41 | Kobaz | well like |
| 17:48.41 | cusco | I don't understand what is wrong on that /etc/asterisk/dahdi-channels.conf |
| 17:48.50 | cusco | to activate the 2nd span |
| 17:48.51 | Kobaz | with sangoma cards... if dahdi is loaded first, and then the sangoma driver |
| 17:48.56 | Kobaz | dahdi dummy will be channel 1 |
| 17:49.02 | cusco | hm, I see |
| 17:49.24 | *** join/#asterisk sircolin (n=sircolin@my83-216-68-241.mynow.co.uk) |
| 17:49.26 | Kobaz | your config is a little weird |
| 17:49.31 | cusco | why? |
| 17:49.33 | Kobaz | but i don't see any errors |
| 17:49.45 | Kobaz | channel =>.. is the last thing you have once you set your configs |
| 17:49.52 | Kobaz | so you have ... channel => 1-15... |
| 17:49.58 | Kobaz | and then context=default and group=63 |
| 17:50.08 | Kobaz | that context and group... is being applied to the next channel def |
| 17:50.21 | cusco | well it was autogenerated by dahdi_genconf |
| 17:50.47 | Kobaz | i mean if it works, that's fine... but some of those options are not being used... they are being set, and then being overridden |
| 17:50.58 | cusco | :/ |
| 17:51.01 | Kobaz | as far as why the channel can't come up... everything looks fine |
| 17:51.04 | Kobaz | pokes [TK]D-Fender |
| 17:52.08 | cusco | thank you Kobaz |
| 17:52.09 | Katty | pokes Kobaz |
| 17:52.12 | [TK]D-Fender | doesn't see * being stopped, dahdi_cfg --- run while down, * restarted, complete configs, or output at the point of failure |
| 17:52.32 | Kobaz | well you don't need to restart asterisk... just unload/load chan_dahdi |
| 17:52.44 | Kobaz | restarting asterisk would be a clean slate though |
| 17:53.05 | Kobaz | the configs are |
| 17:53.06 | cusco | I did thant , and calls would not come in, then I restarted asterisk and I noticed I had no "pri" command |
| 17:53.10 | [TK]D-Fender | Kobaz: NO. No trust. Whatsoever. No half-way measures. |
| 17:53.18 | Kobaz | http://paste.debian.net/49294/ http://paste.debian.net/49291/ http://paste.debian.net/49293/ http://paste.debian.net/49290/ |
| 17:53.33 | Kobaz | cusco: yeah, the pri command won't be available if chan_dahdi fails to load |
| 17:53.33 | cusco | then I commented that 2nd span, and restarted asterisk and i had pri again |
| 17:53.41 | cusco | ah I had an error |
| 17:53.41 | cusco | sorry |
| 17:53.45 | cusco | I remember let me look |
| 17:53.55 | cusco | tham its gone |
| 17:54.03 | cusco | there was an error while trying to laod chan_dadhi |
| 17:55.00 | Kobaz | [Oct 16 18:05:11] ERROR[21877] codec_dahdi.c: Failed to open /dev/dahdi/transcode: No such file or directory |
| 17:55.03 | Kobaz | i've never seen that before |
| 17:55.06 | cusco | by the way, while Im looking for it, this error comes when it reaches chan 31: [Oct 16 18:17:31] ERROR[22660] chan_dahdi.c: !! Got reject for frame 108, but we have nothing -- resetting! |
| 17:55.14 | cusco | and all calls go down |
| 17:55.40 | Kobaz | a call comes in on 31? or dahdi brings up 31 |
| 17:55.48 | cusco | err... |
| 17:56.13 | cusco | good question, what I did to monitor was: asterisk -r "core show channels" |grep DAHDI|wc -l |
| 17:56.43 | *** join/#asterisk errotan (n=errotan@5403E5A1.catv.pool.telekom.hu) |
| 17:56.54 | cusco | asterisk -rx that was |
| 17:57.05 | Kobaz | so just for sanity |
| 17:57.09 | cusco | and it reached 31 |
| 17:57.10 | cusco | :/ |
| 17:57.19 | Kobaz | the output of dahdi_cfg... is that from when asterisk was not running |
| 17:57.27 | ghento | Is there a reason why, in asterisk the ${CALLERID()} is properly set, but when dialed, on my phone, the number does not appear (instead a random 7 digit number shows up) |
| 17:58.01 | DavidR2008 | ghento, does this use POTS at any point? |
| 17:58.05 | Kobaz | Set(CALLERID()=1234..) |
| 17:58.20 | ghento | DavidR2008: SIP -> mobile |
| 17:58.39 | Kobaz | er |
| 17:58.39 | cusco | Kobaz: er, no! asterisk was running |
| 17:58.39 | Kobaz | Set(CALLERID(num)=1234..) |
| 17:58.53 | Katty | is your outbound callerid number set to 10 digit? |
| 17:58.58 | cusco | I just got that [Oct 16 18:17:31] ERROR[22660] chan_dahdi.c: !! Got reject for frame 108, but we have nothing -- resetting! |
| 17:58.59 | Kobaz | cusco: all the usual stuff, isn't working.. so... last resort is trying from a 'clean' state |
| 17:59.03 | Katty | i know several cell phone carries that won't pass anything unless it's 10 digit. |
| 17:59.08 | Kobaz | Katty: yeap |
| 17:59.09 | ghento | Katty: yep it is |
| 17:59.22 | cusco | Kobaz: ok, its a live system, I have to wait until the volume of calls go down |
| 17:59.24 | Kobaz | ghento: your provider is probably overriding your callerid |
| 17:59.30 | Kobaz | cusco: yeah.. i know how that goes |
| 17:59.41 | cusco | so I need to run dahdi_cfg again when asterisk is not running |
| 17:59.52 | ghento | Kobaz: ah true, good point. They are an evil company so I wouldn't be suprised :) |
| 17:59.54 | Kobaz | basically... start from scratch |
| 17:59.56 | Katty | who is your sip providor? |
| 17:59.58 | *** join/#asterisk copantl (n=copantl@190.92.29.37) |
| 18:00.01 | copantl | hi guys |
| 18:00.02 | Kobaz | unload all the drivers |
| 18:00.06 | Kobaz | stop asterisk |
| 18:00.11 | cusco | and try uncoment the 2nd span, log the error while trying to load chan_dadhi.so |
| 18:00.12 | Kobaz | well.. in the opposite order |
| 18:00.15 | Kobaz | yeah |
| 18:00.17 | Katty | if you can get them on the phone, they can tell you what callerid is being sent to them, and then what is sent to the other carrier out |
| 18:00.19 | copantl | i like to know if dahdi supports R2 protocol? |
| 18:00.25 | Kobaz | get exact details of what you did... when... and the logs while doing it |
| 18:00.38 | cusco | ok Kobaz |
| 18:00.44 | Kobaz | and use dahdi_cfg -vv... dmesg... and the asterisk console for debug into |
| 18:00.46 | ghento | Katty: Thanks, thats good advice. I will contact them. |
| 18:01.01 | copantl | i got a te110p but i like to know if i can connect this card to a R2 circuit? |
| 18:01.07 | cusco | Kobaz: are you going to be there much longer? |
| 18:01.12 | Kobaz | i'm around |
| 18:01.18 | Kobaz | tk-fender is also around |
| 18:01.20 | cusco | thanks a lot |
| 18:01.28 | Kobaz | i'll be on till about 7pm est |
| 18:01.40 | cusco | 1 hour time you will leave |
| 18:01.47 | cusco | I will try to acomplish that before then |
| 18:01.50 | Kobaz | it's 2pm right now |
| 18:01.54 | cusco | ah! sorry |
| 18:02.15 | Kobaz | so you got 5 hours |
| 18:02.17 | Kobaz | mush mush |
| 18:02.17 | Kobaz | heh |
| 18:02.27 | copantl | guys anybody know what happen with libr2? |
| 18:05.50 | copantl | does any body know how to configure R2 in asterisk and dahdi? |
| 18:09.09 | DrZeus | question: what does this error mean? No data provided after channel type! |
| 18:09.30 | DrZeus | im trying to give the sip extensions as arguments for a script to originate the call, but gives me that error |
| 18:09.35 | DrZeus | don't understand what it means |
| 18:09.55 | [TK]D-Fender | DrZeus: PASTEBIN <- |
| 18:09.57 | [TK]D-Fender | ~pb |
| 18:09.57 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 18:09.59 | [TK]D-Fender | ^^^^^^^^ |
| 18:10.26 | DrZeus | nods...but is opening pastebin. |
| 18:10.44 | moy | copantl: as of asterisk 1.6.2 you can use r2 with openr2 |
| 18:10.55 | moy | previous versions need to be patched to use it |
| 18:11.15 | DrZeus | http://pastebin.com/d3795c7e4 |
| 18:11.25 | moy | and regarding, libr2, it became libmfcr2 module of unicall |
| 18:11.45 | moy | which then was used for chan_unicall |
| 18:12.06 | Katty | does anyone have a recipe for an awesome fruit salad? |
| 18:13.08 | Qwell | Katty: take fruit. mix liberally |
| 18:13.37 | Katty | Qwell: no special sauce? |
| 18:13.42 | Qwell | fruit sauce |
| 18:13.49 | Katty | k |
| 18:13.50 | DrZeus | [TK]D-Fender: so, what do you think? http://pastebin.com/d3795c7e4 |
| 18:14.27 | Qwell | moy: libr2 or libopenr2? O.o |
| 18:14.35 | Qwell | are they different? I'm confused |
| 18:14.38 | [TK]D-Fender | DrZeus: I don't see the attempt and the failure |
| 18:14.39 | Katty | Qwell: what's your favorite kinds of fruit Qwell? |
| 18:14.46 | Qwell | Katty: the fruity kind |
| 18:14.51 | Katty | >.< |
| 18:14.51 | DrZeus | Katty: http://www.nibbledish.com/ , maybe it can give you ideas |
| 18:15.04 | Katty | looks |
| 18:15.43 | moy | Qwell: libr2, Steve had done libr2 before doing Unicall AFAIK, Asterisk 1.2 still has some ifdefs in chan_zap.c mentioning libr2, then libmfcr2 was born as module to Unicall and later libopenr2 came in |
| 18:15.49 | DrZeus | [TK]D-Fender: http://pastebin.com/d287fee6 |
| 18:15.53 | Qwell | moy: ahh, k |
| 18:15.58 | Katty | eesha. |
| 18:16.01 | Katty | this stuff looks fancy |
| 18:16.13 | Katty | i'm from southern missouri |
| 18:16.15 | DrZeus | i know the second error "There are two ways to use this command...", is related to the misuse of originate\ |
| 18:16.15 | Katty | we don't do much fancy here |
| 18:16.52 | DrZeus | Katty: maybe you can be the new sensation with a fancy fruit salad! |
| 18:16.54 | [TK]D-Fender | DrZeus: Why don't I see you calling it from CLI? echo $CALL <- this should show what you're doing |
| 18:16.59 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:17.08 | Katty | DrZeus: yeah. i'll put bacon in it and submit it to reddit |
| 18:18.31 | Katty | cantelope sounds good right now. |
| 18:18.34 | Katty | very, very good. |
| 18:18.39 | copantl | moy: do you know where i can found how to configurate R2? |
| 18:19.13 | DrZeus | Katty: cantaloupe and bacon...Atkins diet maybe? |
| 18:19.49 | Katty | no just cantaloupe |
| 18:20.20 | Katty | haha there's a recipe on this website called open source pancakes |
| 18:20.22 | Katty | that's hilarious!!! |
| 18:20.43 | [TK]D-Fender | DrZeus: Cantaloupe is very Antt-Atkins |
| 18:20.58 | [TK]D-Fender | drmessano: and Dr. Atkins KILLED HIMSELF with his "diet" |
| 18:21.07 | [TK]D-Fender | DrZeus: and Dr. Atkins KILLED HIMSELF with his "diet" |
| 18:21.11 | [TK]D-Fender | dang auto-complete |
| 18:21.12 | Qwell | So... |
| 18:21.16 | Qwell | it sounds like a great idea. |
| 18:21.18 | Katty | a needle pulling thread! |
| 18:21.27 | [TK]D-Fender | Qwell: I have a few candidates lined up already! |
| 18:21.39 | Katty | open source Bacon pancakes |
| 18:21.54 | [TK]D-Fender | Katty: NO SOUND OF MUSIC! |
| 18:21.55 | Katty | mwuahahaha, i bet that'd be good. |
| 18:22.03 | Katty | [TK]D-Fender: :< |
| 18:22.11 | Katty | [TK]D-Fender: THE HILLLLLSSS ARE ALIVE |
| 18:22.15 | Katty | [TK]D-Fender: WITH THE SOUND OF MUSICCCC |
| 18:22.18 | Katty | [TK]D-Fender: La la la la! |
| 18:22.21 | [TK]D-Fender | bombs the hills |
| 18:22.27 | DrZeus | [TK]D-Fender: oh, didn't knew that |
| 18:22.33 | [TK]D-Fender | loves the smell of napalm in the morning |
| 18:22.48 | DrZeus | well, he ate a lot of bacon, that's for sure! |
| 18:23.18 | [TK]D-Fender | DrZeus: where's my new backup? |
| 18:23.24 | Katty | http://farm1.static.flickr.com/48/120944972_7a46a1e32d.jpg <- Dinner. |
| 18:24.07 | DrZeus | On April 8, 2003, at age 72, a day after a major snowstorm in New York, Dr. Atkins slipped on the ice while walking to work, hitting his head and causing bleeding around his brain. |
| 18:24.24 | Katty | that's most unfortunate. |
| 18:24.25 | DrZeus | His death certificate states that the cause of death was "blunt impact injury of head with epidural hematoma" |
| 18:24.30 | Qwell | DrZeus: he wouldn't have tripped if he was chubby |
| 18:25.20 | DrZeus | Qwell: omG... |
| 18:25.30 | Katty | oh emmm gee! |
| 18:25.31 | Qwell | DrZeus: it all makes sense now, doesn't it? |
| 18:25.41 | DrZeus | just can't believe it. |
| 18:25.55 | DrZeus | [TK]D-Fender: there is the final string of the command http://pastebin.com/d45414ad1 |
| 18:26.13 | DrZeus | that's what originates from the script |
| 18:27.54 | moy | copantl: there is a full guide in openr2 google code site |
| 18:27.55 | DrZeus | this one's better: http://pastebin.com/d43449d6e it has the total output |
| 18:28.14 | *** join/#asterisk Firass-z0r (n=asadf@c-67-201-205-34.reshall.wwu.edu) |
| 18:28.39 | copantl | moy : thanks |
| 18:28.41 | [TK]D-Fender | DrZeus: I want to se it being run |
| 18:28.57 | *** part/#asterisk fnord0 (n=fnord0@unaffiliated/fnord0) |
| 18:28.59 | *** join/#asterisk d00gster (n=doughant@94.99.57.187) |
| 18:29.08 | DrZeus | http://pastebin.com/d43449d6e , this one has the whole shell running the command |
| 18:30.34 | ChannelZ | Ok here's one. I've got an extension in my dialplan so you can pick up a different ringing extension using Pickup() - Is it possible to get the picking-up extension to 'suck up' the CallerID from the actual called extension? |
| 18:31.07 | *** join/#asterisk bluOxigen (n=asad@static-host119-73-64-76.link.net.pk) |
| 18:31.37 | [TK]D-Fender | drdrun asterisk -rx 'originate sip/3000 extension 3001@from-internal' yourself |
| 18:32.56 | DrZeus | [TK]D-Fender: drdrun? |
| 18:33.15 | [TK]D-Fender | DrZeus: autocomplete fail |
| 18:33.29 | DrZeus | oh ok; yes, when I run that line the call gets connected |
| 18:34.38 | DrZeus | I think this error: No data provided after channel type! is what needs to be addressed; i don't know what is it |
| 18:35.54 | [TK]D-Fender | DrZeus: probably an escaping issue in your script |
| 18:36.50 | *** join/#asterisk mchou (n=quassel@unaffiliated/mchou) |
| 18:36.54 | DrZeus | im trying to check if there is any whitespace or something |
| 18:37.19 | [TK]D-Fender | drthere was a double in your script before |
| 18:40.35 | *** join/#asterisk ManxPower-work (n=EWieling@24.42.221.26) |
| 18:40.37 | DrZeus | [TK]D-Fender: double? |
| 18:41.03 | ManxPower-work | How can I tell if the HWEC is active? All I see in the logs is "dahdi_echocan_mg2: Registered echo canceler 'MG2" |
| 18:41.35 | ManxPower-work | Sorry, I also get this message when the driver loads "VPMADT032: Present and operational (Firmware version 117)" |
| 18:41.47 | [TK]D-Fender | DrZeus: CALL=`asterisk -rx 'originate $ORIGEN extension $DEST'` |
| 18:43.09 | DrZeus | what do you mean by double? |
| 18:43.13 | ManxPower-work | Users are complaining of massive echo. I'm trying to figure out if the crappy software EC is being used or the almost as crappy HWEC is being used. |
| 18:47.05 | ManxPower-work | Personally I'm recommending we swap out the Digum card for a Sangoma. |
| 18:49.31 | asterwiki | ManxPower-work: Sangoma I find is excellent, using it on over 12 boxes across spanning 7 different countries with no problem (using various strains of 1.2, 1.4, 1.6); |
| 18:49.35 | *** join/#asterisk sfr33man (n=sfreeman@64.183.147.98) |
| 18:50.09 | [TK]D-Fender | DrZeus: Count your spaces |
| 18:50.11 | *** join/#asterisk voipmonk (n=voipmonk@74-132-202-71.dhcp.insightbb.com) |
| 18:50.57 | ManxPower-work | asterwiki: I'm just trying to convince my boss to dump the Digium analog cards for Sangoma. |
| 18:50.59 | DrZeus | I tried this alone(with a shell function, to keep the variables values) asterisk -rx 'originate $ORIGEN extension $DEST' , and gave me the error |
| 18:53.16 | ManxPower-work | asterwiki: I can't even figure out of HWEC or SW EC is happening on the box. |
| 18:54.16 | ManxPower-work | It looks the best way to just pray |
| 18:55.29 | Qwell | So, you haven't even determined if it's the hardware, and you want to replace it? |
| 18:55.54 | asterwiki | ManxPower-work: let me see your wanpipe config file |
| 18:56.22 | ManxPower-work | Qwell: I've never had problems with Sangome. |
| 18:56.32 | Qwell | You don't even know if it's the hardware. |
| 18:56.41 | ManxPower-work | But I could start troubleshooting this Digium card if I knew how to find out if it's using HWEC or not. |
| 18:56.50 | ManxPower-work | Qwell: We have at least 5 systems with chronic echo issues. |
| 18:57.10 | ManxPower-work | Qwell: Do you know the answer to that? |
| 18:57.41 | Qwell | No, but support would. |
| 18:58.02 | Joel | I'm a big fan of just getting rid of analog lines |
| 18:58.05 | Joel | it's always the best option |
| 18:58.09 | ManxPower-work | I've never had support solve any issue I've had. But I'll give them another call. |
| 18:58.53 | ManxPower-work | Joel: Until my new job I refused to deal with analog lines at all. At me new job that's not an option |
| 19:01.12 | Joel | sounds like it could be time to make new->old |
| 19:01.12 | Joel | :D |
| 19:01.28 | eppigy | ALLO KATTY |
| 19:01.41 | ManxPower-work | Joel: My old job had other issues like a new, stupid, IT manager |
| 19:03.30 | TJNII | screams at the undocumented, un-intuituve, non-user friendly program he as to use for work |
| 19:04.12 | *** join/#asterisk umay (n=chris@174-16-31-61.hlrn.qwest.net) |
| 19:04.18 | jaytee | ManxPower-work, sounds like my present job |
| 19:04.27 | [TK]D-Fender | screams at the un-intuituve, non-friendly users at his work |
| 19:04.28 | ManxPower-work | I figure with as much time as we'd save with Sangoma, we woudl more than save enough money to pay for itself. |
| 19:04.43 | ManxPower-work | If they were T-1 interfaces I'd just throw a tellabs on it. |
| 19:05.46 | ManxPower-work | Qwell: Just e-mailed support. I'll let you know if they have any useful informatio |
| 19:07.14 | eppigy | [TK]D-Fender: sir please calm down |
| 19:08.06 | DrZeus | [TK]D-Fender: still nothing. |
| 19:09.36 | [TK]D-Fender | DrZeus: Well your scripting seems to have an error, and I'm not really qualified to debug it. |
| 19:10.59 | Kobaz | it's teh bork |
| 19:10.59 | DrZeus | [TK]D-Fender: that's what it seems then...thank you or your time with this |
| 19:15.31 | *** join/#asterisk Naikrovek (n=jjohnson@unaffiliated/naikrovek) |
| 19:17.10 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:17.54 | CcRnp | can anyone helpme who to use CEL ? |
| 19:18.53 | *** join/#asterisk Gugge (n=gugge@91.208.16.1) |
| 19:20.35 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 19:20.44 | *** join/#asterisk Ad-Hoc (n=nimbus@ppp169-251.adsl.forthnet.gr) |
| 19:23.49 | CcRnp | Please help me for CEL logging, how can i use it ? |
| 19:24.45 | ManxPower-work | Yay! My boss said all new systems will be Sangoma |
| 19:26.17 | ManxPower-work | CcRnp: if we knew what CEL is we might be able to? |
| 19:27.13 | CcRnp | its Channel Event Logging |
| 19:29.07 | ManxPower-work | Ah. I didn't know that was a feature of Asterisk |
| 19:30.34 | *** join/#asterisk kazaa_lite (n=msaleem@cpc1-lamb4-0-0-cust590.bmly.cable.ntl.com) |
| 19:30.54 | [TK]D-Fender | CcRnp: And where do we see that this has been merged with production * branches? |
| 19:33.37 | CcRnp | http://www.asterisk.org/node/48358 |
| 19:35.07 | [TK]D-Fender | CcRnp: and what does this show us? |
| 19:35.31 | DrZeus | im Done. |
| 19:35.38 | DrZeus | don't know what is happening |
| 19:36.08 | [TK]D-Fender | CcRnp: http://www.venturevoip.com/news.php?rssid=2011 <- this is NEWER than the article you just linked |
| 19:36.17 | CcRnp | recording the call events for transfers using CEL |
| 19:36.57 | CcRnp | i went throught this article but there is no http://svn.digium.com/svn/asterisk/team/group/newcdr |
| 19:36.58 | *** join/#asterisk kenwiesner (n=kenwiesn@173-24-52-119.client.mchsi.com) |
| 19:38.03 | ManxPower-work | CcRnp: all cdr options are in the /etc/asterisk/cdr*.conf files. |
| 19:38.07 | [TK]D-Fender | CcRnp: that clearly isn't the good path. go cruise the base of SVN to find the proper path |
| 19:38.28 | CcRnp | alrite ! let me check it out |
| 19:38.58 | DrZeus | im leaving folks; have a great day/night/etc. |
| 19:39.02 | DrZeus | bye\ |
| 19:45.38 | *** join/#asterisk garymc (n=garymc@host86-159-106-40.range86-159.btcentralplus.com) |
| 19:46.16 | *** part/#asterisk kenwiesner (n=kenwiesn@173-24-52-119.client.mchsi.com) |
| 19:48.09 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 19:48.23 | TJNII | just found a "$10 off urgent care cupon" |
| 19:48.26 | garymc | Hi guys. Boss just asked me. Can we plug a Fax into this asterisk system? Im not sure |
| 19:48.37 | TJNII | There is something just not right about that.... |
| 19:48.38 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.157) |
| 19:48.41 | garymc | through an rj45 or somthing? |
| 19:48.52 | garymc | I dont know why he wants a fax? |
| 19:50.27 | [TK]D-Fender | garymc: You have a digital interface card. No, you cannot plug a fax machine into it |
| 19:50.40 | Gugge | garymc: some ata with T.38 could maybe work ... |
| 19:50.43 | garymc | Ok thanks. what would he use for fax then? |
| 19:50.52 | asterwiki | garymc: Get an ATA (Cisco186 or LinkSys SPA2102 / PAP2),,, |
| 19:51.22 | TJNII | gandhijee: Depends. Does your ITSP support faxing or do you have physical lines? |
| 19:51.42 | TJNII | s/gandhijee/garymc/ |
| 19:52.23 | [TK]D-Fender | garymc: get yourself an A200 and the sync cable for the 2 cards |
| 19:52.32 | *** join/#asterisk trifon (n=chatzill@tmo-100-84.customers.d1-online.com) |
| 19:52.56 | garymc | Cant he just use the internet? |
| 19:53.16 | garymc | to send faxes? |
| 19:53.42 | garymc | ill just get him to use a anolougue line hes got |
| 19:54.01 | *** join/#asterisk |R (n=bob@66.49.231.84) |
| 19:54.10 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 19:55.00 | |R | anyone had a major borkage on debian going to asterisk 1.6.2.0~dfsg~rc1-1 with their voicemail? |
| 19:55.24 | asterwiki | garymc: unless you are going to setup Hylafax which can both send/receive faxes |
| 19:55.42 | [TK]D-Fender | garymc: "just use the internet"? How do you take a physical fax you have and "just use the internet"? |
| 19:56.08 | garymc | yeah i reckon internet is best, but hes proper old school |
| 19:56.16 | TJNII | [TK]D-Fender: methinks he hasn't thought his cunning plan all the way through. |
| 19:56.18 | garymc | so are some of his customers |
| 19:56.46 | garymc | well possibly not as i forgot all about the fax |
| 19:56.58 | [TK]D-Fender | |R: 1.6.2 is in RC status, not even full release. It should not be used in production. |
| 19:57.00 | Kobaz | the interwebs is the solution for everything |
| 19:57.13 | Kobaz | *EVERYTHING* |
| 19:57.50 | |R | [TK]D-Fender : damn me for running unstable on my gateway... but always worked under 1.4, i was wondering if it was a known issue with a simple fix :) |
| 19:57.56 | garymc | I can think of things the internet is no good for |
| 19:58.13 | |R | [TK]D-Fender: i'm actually wondering why is debian packaging RC1 versions... |
| 19:58.22 | Kobaz | garymc: yeah... sex over the internet doesn't work as well |
| 19:58.28 | Kobaz | garymc: okay... so... mostly everything |
| 19:58.44 | garymc | I didnt even think of sex |
| 19:58.51 | garymc | :P |
| 19:59.07 | *** join/#asterisk IBC_jkenney (n=jkenney@ip65-44-169-66.z169-44-65.customer.algx.net) |
| 19:59.29 | IBC_jkenney | anyone here have any experience with IAXMODEm |
| 20:00.27 | IBC_jkenney | I am having a problem where its not running right and i don't quite know why |
| 20:00.42 | Kobaz | IBC_jkenney: logs, errors, configs... the usual |
| 20:01.24 | *** join/#asterisk andres833 (n=andres83@201.244.125.6) |
| 20:02.53 | |R | Thanks :) |
| 20:02.55 | *** part/#asterisk |R (n=bob@66.49.231.84) |
| 20:03.52 | garymc | would you say faxing is getting phased out? |
| 20:04.05 | garymc | its like proper old tech now |
| 20:04.13 | garymc | im gonna talk him out of it |
| 20:04.15 | garymc | :) |
| 20:05.12 | [TK]D-Fender | garymc: Sure you are.... and when you're done I'm sure there are some eskimos that just haven't realized how much they want to buy a refridgerator from you ;) |
| 20:06.05 | Kobaz | hey sometimes it gets warm up there |
| 20:06.31 | garymc | infact eskimos need refridgerators otherwise all their goods would freeze!! ;) |
| 20:06.52 | netpro25_ | So I am still fooling around with my NAT issues. My server is a public IP on the web, and my client Ekiga is behind a NAT. I cannot get Ekiga to connect. When I look at the packets in asterisk I see that the client is not receiving any response packets. |
| 20:07.44 | [TK]D-Fender | netpro25_: Do you know what we see? |
| 20:08.10 | netpro25_ | [TK]D-Fender: nope |
| 20:08.20 | [TK]D-Fender | netpro25_: Nothing :0 |
| 20:08.40 | [TK]D-Fender | netpro25_: And the reason you don't have SIP debug up in a pastebin for us to look at is...? :) |
| 20:08.41 | netpro25_ | [TK]D-Fender: I will pastebin some info |
| 20:09.24 | *** join/#asterisk mahlon (i=mahlon@martini.nu) |
| 20:09.25 | garymc | netpro25_ good lad thats the spirit ;) |
| 20:10.36 | netpro25_ | lol |
| 20:10.38 | netpro25_ | so here it is |
| 20:10.39 | netpro25_ | http://pastebin.com/m74392456 |
| 20:10.55 | *** join/#asterisk chodorenko (n=chodoren@ext.one.by) |
| 20:10.55 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 20:11.06 | netpro25_ | Client is saying "Could not register" |
| 20:11.18 | netpro25_ | (Timeout) |
| 20:11.48 | [TK]D-Fender | SIP/2.0 401 Unauthorized <------- |
| 20:12.09 | [TK]D-Fender | netpro25_: What router is it behind? |
| 20:12.17 | netpro25_ | Iptables |
| 20:13.13 | [TK]D-Fender | netpro25_: It should get the unauthorized error we see there... something is fishy with your routing. |
| 20:13.30 | [TK]D-Fender | netpro25_: Go check yo firewalls out, etc |
| 20:14.17 | netpro25_ | k |
| 20:14.43 | *** join/#asterisk denon (i=denon@sassinak.net) |
| 20:14.43 | *** mode/#asterisk [+o denon] by ChanServ |
| 20:15.17 | *** join/#asterisk flynux (n=flynux@2a01:38:0:0:0:0:0:0) |
| 20:15.20 | *** join/#asterisk jshriver (n=jshriver@cblmdm24-53-165-86.buckeyecom.net) |
| 20:15.35 | jshriver | greeting I'm trying to create a test demo account and I keep getting this error |
| 20:15.38 | jshriver | request '500@default' does not exist |
| 20:15.49 | jshriver | though 500 is listed in voicemail.conf, sip.conf and extension.conf |
| 20:16.09 | [TK]D-Fender | jshriver: pastebin the entire call where that is generated |
| 20:16.17 | [TK]D-Fender | jshiAlone it provides no sense of context. |
| 20:16.37 | jshriver | ok one second |
| 20:18.11 | jshriver | ok |
| 20:18.17 | jshriver | http://pastebin.com/m62cb9d8 |
| 20:18.29 | jshriver | just changed IP and server name |
| 20:19.07 | *** join/#asterisk shido6 (n=shido6@74-132-202-71.dhcp.insightbb.com) |
| 20:19.08 | jshriver | on the target system I get this: |
| 20:19.09 | jshriver | Oct 16 16:17:03 NOTICE[1908]: chan_iax2.c:7398 socket_read: Rejected connect attempt from 72.240.39.37, request '500@default' does not exist |
| 20:19.27 | [TK]D-Fender | jshriver: Where is the backup for it? |
| 20:19.33 | jshriver | ? |
| 20:19.44 | [TK]D-Fender | jshriver: the DIALPLAN? |
| 20:19.44 | jshriver | backup? |
| 20:19.53 | jshriver | ok let me get that one moment |
| 20:20.40 | jshriver | http://pastebin.com/m572dbb0c |
| 20:20.47 | *** join/#asterisk ManxPower-work (n=EWieling@24.42.221.26) |
| 20:21.13 | jshriver | Basically want to create an extension that plays some demo voice. So I can test the lines. |
| 20:21.41 | [TK]D-Fender | jshriver: Not some tiny portion of it <-------- |
| 20:22.06 | jshriver | so you want the whole extensions.conf? |
| 20:22.10 | chodorenko | jshriver: on remote mashine show iax2 peer |
| 20:22.18 | [TK]D-Fender | jshriver: and for the obvious... |
| 20:22.27 | [TK]D-Fender | jshriver: extern => 500, 1, Wait, 1 <----- Spelling FAIL |
| 20:22.44 | [TK]D-Fender | jshriver: Wash. Rinse. Repeat |
| 20:22.47 | chodorenko | may be context not true |
| 20:22.50 | jshriver | what is wrong with that line |
| 20:22.57 | [TK]D-Fender | jshriver: extern => 500, 1, Wait, 1 <----- Spelling FAIL |
| 20:23.03 | [TK]D-Fender | jshriver: Seriously... LOOK at it |
| 20:23.21 | *** join/#asterisk JKac3BEq (n=JKac3BEq@173-20-71-143.client.mchsi.com) |
| 20:23.38 | jshriver | extern and Wait are spelled correctly. |
| 20:23.40 | [TK]D-Fender | jshriver: And then remove the extra white-space, and us () for your app data |
| 20:23.48 | *** join/#asterisk hugorebelo (n=hugorebe@200.171.132.124) |
| 20:23.59 | [TK]D-Fender | [16:23]<jshriver>extern and Wait are spelled correctly. <- FAIL |
| 20:24.05 | jshriver | oh no r |
| 20:24.14 | [TK]D-Fender | reaches for his ClueBat (tm) |
| 20:24.25 | ManxPower-work | now you know why we want to SEE IT. |
| 20:24.55 | jshriver | that was it |
| 20:24.56 | ManxPower-work | jshriver: you might want to re-read the Asterisk book |
| 20:25.02 | TJNII | EXTENsion not EXTERNal |
| 20:25.10 | jshriver | recommend one? I can't stand working with phones but it's part of my job now |
| 20:25.18 | ManxPower-work | ~book |
| 20:25.19 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 20:25.22 | *** join/#asterisk bn-7bc (n=bjarne-i@97.84-49-72.nextgentel.com) |
| 20:25.37 | [TK]D-Fender | Ok, checkout time, BBL |
| 20:26.05 | jshriver | Heard ther is an "echo" demo.. anyone know how to set that up? I'm having random dropped calls but it's not bw related |
| 20:26.38 | Qwell | jshriver: call the Echo application.. |
| 20:26.51 | jshriver | ok googling now |
| 20:26.56 | TJNII | jshriver: I like working with phones. Why don't we switch jobs? You can deal with this unducumented mess of Visual Basic that my boss expects me to become an "expert" on. |
| 20:27.11 | jshriver | my extent of* is editing extensions.conf and copy/paste and changing #'s. |
| 20:27.44 | jshriver | TJNII: :) not a fan of Visual Studio or it's languages, but would trade that over this :) wish I could |
| 20:27.51 | jshriver | VB ouch :( |
| 20:28.03 | ManxPower-work | You need to drink the Microsoft Kool-Aid |
| 20:28.27 | jshriver | heh |
| 20:29.22 | TJNII | jshriver: I don't even have the code. I'm just trying to figure out how to _use_ it. Though tech support is willing to program it to do whatever I want at very reasonable rates.... |
| 20:29.25 | ManxPower-work | jshriver: find yourself a consultant. |
| 20:29.47 | TJNII | (And they let my boss know about said rates whenever I call to ask a "How do I" question) |
| 20:30.09 | *** join/#asterisk Pazzo (n=ugelt@195.254.246.59) |
| 20:30.21 | IBC_jkenney | i fixed it |
| 20:30.34 | IBC_jkenney | sorry i started putting stuff in paste bin and saw it |
| 20:32.41 | jshriver | ManxPower-work: ty at this point I woul pay out of pocket for someone to come in and clean this crap up and let me observe to understand how it works. |
| 20:33.38 | jshriver | Asterisk as a software platform seems pretty solid, I've had a heck of a time with the zaptel cards though, seems flaky |
| 20:33.59 | jshriver | those little red things burn out easy, even with surge protectors on each line. |
| 20:34.38 | Kobaz | yeah, that's digium cards |
| 20:34.43 | Kobaz | i've never had a sangoma fry |
| 20:35.03 | jshriver | will look them up.. went with digium since they support *, but the hardware sucks |
| 20:36.20 | *** part/#asterisk jshriver (n=jshriver@cblmdm24-53-165-86.buckeyecom.net) |
| 20:37.56 | *** join/#asterisk netpro25_ (n=mmanning@64-238-176-104.ksg.apt.gru.net) |
| 20:41.17 | Naikrovek | one would think that because hardware is (I imagine) Digium's main source of income that they would put some effort into improving the quality |
| 20:41.54 | Chainsaw | My TDM400 keeps on trucking. |
| 20:41.56 | *** join/#asterisk [TK]D-Fender (n=joeblow@161.216.159.31) |
| 20:42.06 | Chainsaw | It does hate the BT line test, but I believe there's a patch in the tracker somewhere. |
| 20:42.27 | Chainsaw | (Which, traditionally, hasn't been applied yet and is in limbo. But it's better then nothing.) |
| 20:42.42 | *** join/#asterisk neurosys (n=vinix@c-71-196-19-254.hsd1.fl.comcast.net) |
| 20:43.25 | p3nguin | What's better than nothing? |
| 20:43.29 | ManxPower-work | Chainsaw: how many calls a day does the card process? |
| 20:43.45 | Chainsaw | ManxPower-work: If it receives 5 faxes on a single day, it is *extremely* busy. |
| 20:44.18 | ManxPower-work | Chainsaw: try production volume sometime. |
| 20:44.49 | Chainsaw | ManxPower-work: I wouldn't want production volume on an analog line. My calls are coming/going on ISDN BRI. |
| 20:45.54 | *** join/#asterisk geneticx (n=chatzill@host-208-88-126-198.biznesshosting.net) |
| 20:47.00 | geneticx | hello, I would like to know how I can set up my dialplan so the next available trunk is used if the first one is congested or busy with another call..can anyone point me in the right direction? |
| 20:47.35 | ManxPower-work | geneticx: use g1 instead of 1 in your Zap Dial line. i.e. Dial(Zap/g1/5551212). set the group number in zapata.conf |
| 20:49.45 | *** join/#asterisk Toerkeium (i=Toerkeiu@201.216.206.221) |
| 20:51.03 | [TK]D-Fender | geneticx: dial them back to back |
| 20:53.03 | *** join/#asterisk louben (n=lou@212-70-216-131.ath.static.tee.gr) |
| 20:55.06 | geneticx | [TK]D-Fender: can a group be created like suggested above? |
| 20:55.42 | ManxPower-work | geneticx: for Zap channels sure |
| 20:56.01 | *** join/#asterisk friartuck (n=pmccary@66.162.90.57) |
| 20:56.18 | [TK]D-Fender | ONLY zap/dahdi |
| 20:56.36 | geneticx | Ok |
| 20:57.07 | geneticx | Thank you both |
| 20:59.13 | *** join/#asterisk sircolin (n=sircolin@my83-216-68-241.mynow.co.uk) |
| 20:59.17 | *** join/#asterisk Mango (n=Mango@d154-20-72-219.bchsia.telus.net) |
| 21:00.20 | Mango | I'm in Vancouver connecting to a SIP server in Los Angeles. I just switched ISPs and my latency has jumped from 35ms to 120ms because they route traffic to Los Angeles through Washington DC. |
| 21:00.29 | Mango | Do I have any hope of having them fix that? |
| 21:03.06 | *** join/#asterisk luminblade (n=luminbla@cpe-24-28-78-39.austin.res.rr.com) |
| 21:03.38 | *** part/#asterisk luminblade (n=luminbla@cpe-24-28-78-39.austin.res.rr.com) |
| 21:03.48 | *** join/#asterisk luminblade (n=luminbla@cpe-24-28-78-39.austin.res.rr.com) |
| 21:05.12 | ManxPower-work | Mango: Are you asking us if we can redesign your ISP's network? |
| 21:05.37 | Mango | No. I'm asking if I have any hope of convincing them to do it. |
| 21:05.51 | p3nguin | It's not looking good at this point. |
| 21:05.59 | Mango | Heh. |
| 21:06.10 | p3nguin | Maybe you can choose another server closer to DC. |
| 21:06.14 | netpro25_ | [TK]D-Fender: so I found something similar to my issue online |
| 21:06.21 | netpro25_ | http://code.google.com/p/sipdroid/issues/detail?id=15 |
| 21:06.31 | Mango | next closest is 80ms :( |
| 21:06.39 | p3nguin | It's better. |
| 21:06.42 | Mango | True. |
| 21:06.50 | netpro25_ | [TK]D-Fender: basically says you have to use this format instead of what I was using auth=31337@my-super-secret-password@domain.com |
| 21:07.37 | luminblade | i'm trying to do T.38 pass-thru from one SIP peer to another, but Asterisk is crashing when T.38 is negotiated... I have t38pt_udptl=yes set on both peers (and in general). I have also recomipled with "global_t38_capability = T38FAX_VERSION_1..." in chan_sip.c... is there anything else I need to d o? |
| 21:09.20 | *** join/#asterisk superbeef (n=IMP-IT@74.84.194.4) |
| 21:09.28 | ManxPower-work | luminblade: if it's crashing make sure you are using the latest Asterisk and if you are, file a bug report. Asterisk should not crash |
| 21:09.45 | Mango | "David: I understand that you are concerned about the speed of your connection." |
| 21:09.47 | Mango | That ain't good. |
| 21:11.32 | *** join/#asterisk Ad-Hoc (n=nimbus@ppp169-251.adsl.forthnet.gr) |
| 21:15.44 | netpro25_ | Can someone take a look at my pastebin? http://pastebin.com/m2c5760b0 |
| 21:16.16 | netpro25_ | I was getting unauthorized errors now it just keeps trying to resend packets and they never get to the device. |
| 21:16.51 | *** part/#asterisk came0 (n=came0@rrcs-71-42-25-233.se.biz.rr.com) |
| 21:18.14 | *** join/#asterisk toddejohnson (n=toddejoh@70.226.215.44) |
| 21:19.01 | luminblade | apart from asterisk should not crash, is there anything else that should be required for peer to peer pass-through of T.38? does it work? |
| 21:19.49 | luminblade | ...asterisk in this case is not talking to ata's or any other device, just a SIP peer (a carrier on both sides). |
| 21:20.15 | Chainsaw | luminblade: It starts out on ulaw/alaw until the fax tone is heard. Don't try to block everything except T.38 |
| 21:20.47 | Chainsaw | luminblade: (And you'll want Asterisk 1.6 for proper T.38 operation) |
| 21:21.20 | luminblade | i have ulaw and g729 enabled. regular ulaw and g729 calls work on the peers... i'm on 1.4.26.2, i was afraid of the 1.6 answer :) |
| 21:22.46 | CrazyTux[w] | [TK]D-Fender: needing a bit of help.... remember that solution you guys helped me with the other day for Local -> and I'm passing like a simultaneous dial out? |
| 21:23.02 | *** join/#asterisk came0 (n=came0@rrcs-71-42-25-233.se.biz.rr.com) |
| 21:23.14 | CrazyTux[w] | [TK]D-Fender: i.e. i'm passing a header i.e. Local/FOOA@internal&LOCAL/FOOB@internal to a Dial() command, then splitting that string and dialing out |
| 21:23.25 | CrazyTux[w] | [TK]D-Fender: ring any bells? |
| 21:23.38 | *** part/#asterisk came0 (n=came0@rrcs-71-42-25-233.se.biz.rr.com) |
| 21:26.25 | CrazyTux[w] | [TK]D-Fender: in any regard, I'm having a problem where in [context1] Dial(LOCAL.....,g) (so upon CALLED partying hanging up it continues in dial plan...... and also have it inside of the LOCAL/Dial...... but the problem I'm having is when a destination is called and they hang up it doesnt keep dialing the other contacts |
| 21:30.47 | [TK]D-Fender | brb |
| 21:31.13 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 21:31.57 | [TK]D-Fender | CrazyTux[w]: link & Q again... |
| 21:35.13 | CrazyTux[w] | [TK]D-Fender: it was you and the other guy on the save wave link that day :) |
| 21:36.54 | CrazyTux[w] | [TK]D-Fender: here is the problem: [contextA] exten => s,1,Dial(Local/A@internal&Local/B@internal) [internal] exten => X.,1,Dial(SIP/${EXTEN}@GW,g) ............ upon hangup at this "internal" Local -> Dial (lets say SIP/A was picked up and hung up) I still want SIP/B to ring |
| 21:37.14 | CrazyTux[w] | [TK]D-Fender: which I think I effectively see why this is not possible like this as it is "one call" essentially? because of the parent "Dial" |
| 21:37.37 | CrazyTux[w] | [TK]D-Fender: To be completely clear, in this scenario I want to be able to hangup on SIP/A and have SIP/B still ringing |
| 21:37.47 | *** join/#asterisk Ritzerisk (n=Ritztech@nv-65-40-156-46.sta.embarqhsd.net) |
| 21:38.46 | Ritzerisk | would anyone know if they ever hear (only when calling a autoattendant or a system that picks up like vmail i get slow audio coming back its like they took the recording and slowed it down |
| 21:38.49 | Ritzerisk | verry odd |
| 21:38.49 | Ritzerisk | haha |
| 21:39.07 | *** join/#asterisk bn-7bc (n=bjarne-i@83.170.109.88) |
| 21:43.00 | cusco | Kobaz: you there? |
| 21:45.40 | [TK]D-Fender | crazyYou can as long as A isn't answered <- |
| 21:45.59 | [TK]D-Fender | CrazyTux[w]: And you were supposed to use M() to validate the answer, NOT "g" |
| 21:46.36 | [TK]D-Fender | Ritzerisk: ---> |
| 21:46.38 | [TK]D-Fender | ~gsmbug |
| 21:46.38 | infobot | [~gsmbug] there is a bug compiling Asterisk with GCC 4.2 where optimization errors cause GSM transcoding to distort heavily. Compile with GCC 4.1 or lower, or follow this patch : http://bugs.digium.com/view.php?id=11243 Also please read : http://forums.digium.com/viewtopic.php?p=116731&sid=3c65e49ec840380ed20f1c8426382b39 |
| 21:46.40 | [TK]D-Fender | ^^^^^^^ |
| 21:46.52 | [TK]D-Fender | Ritzerisk: Onlyt hing like it I've ever heard |
| 21:47.06 | [TK]D-Fender | Ritzerisk: Unless you're running VM's |
| 21:47.10 | CrazyTux[w] | [TK]D-Fender: I do have M in there, g was just stating the g option is in effect too |
| 21:47.16 | CrazyTux[w] | [TK]D-Fender: rgM() |
| 21:47.27 | [TK]D-Fender | CrazyTux[w]: r = EVIL!!!!!!!!! |
| 21:47.30 | CrazyTux[w] | [TK]D-Fender: the validation is coming through..... |
| 21:47.48 | [TK]D-Fender | CrazyTux[w]: Show me what its doing and we'll see |
| 21:47.51 | CrazyTux[w] | [TK]D-Fender: but the problem is, I answer on SIP/A, SIP/B is still ringing, but if i hang up on SIP/A, SIP/B stops ringing and hangs up too... which is not the behavior that I want |
| 21:48.12 | [TK]D-Fender | CrazyTux[w]: Show me |
| 21:48.16 | Katty | peeks in |
| 21:48.29 | Katty | has honeydew :> |
| 21:51.01 | hardwire | and bacon? |
| 21:51.57 | Katty | no |
| 21:51.59 | Katty | just honeydew melon |
| 21:52.09 | *** join/#asterisk cuco (n=cuco@bzq-82-81-32-128.red.bezeqint.net) |
| 21:55.18 | *** part/#asterisk robl^laptop (n=robl@m4c5336d0.tmodns.net) |
| 21:56.37 | Nugget | yum |
| 21:56.46 | Katty | yesh |
| 21:56.55 | *** part/#asterisk sircolin (n=sircolin@my83-216-68-241.mynow.co.uk) |
| 21:59.00 | *** join/#asterisk manxpower (n=ewieling@24.42.221.26) |
| 22:02.59 | Ritzerisk | even sooo gsmbug but when i call a cell phone or a LIVE caller i have no problems only when i call like a recording VM or autoattendant |
| 22:08.04 | [TK]D-Fender | Ritzerisk: Transcoding <- |
| 22:08.18 | [TK]D-Fender | Ritzerisk: I doubt GSM is involved in your voice call... |
| 22:08.52 | [TK]D-Fender | Ritzerisk: But the default install options include only GSM prompts and VM recordings are that way as well per samples (where most people start from) |
| 22:11.13 | *** join/#asterisk cuco (n=cuco@bzq-82-81-32-128.red.bezeqint.net) |
| 22:19.33 | *** join/#asterisk DelphiWorld (n=Miranda@196.20.95.121) |
| 22:19.40 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 22:28.47 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
| 22:29.31 | DelphiWorld | how use a authentication user name to register to a sip provider? |
| 22:29.44 | DelphiWorld | this provider require a user name and a auth_username |
| 22:33.23 | [TK]D-Fender | authuser= |
| 22:33.31 | [TK]D-Fender | username= |
| 22:33.32 | [TK]D-Fender | ^^^^^^ |
| 22:35.08 | DelphiWorld | [TK]D-Fender: thx |
| 22:35.14 | *** join/#asterisk manxpower (n=ewieling@24.42.221.26) |
| 22:36.37 | *** part/#asterisk luminblade (n=luminbla@cpe-24-28-78-39.austin.res.rr.com) |
| 22:40.05 | DelphiWorld | [TK]D-Fender: asterisk can register to a provider or just route call without register? |
| 22:40.19 | DelphiWorld | [TK]D-Fender: i'm giving a trunk to a friend that there asterisk call me but is not registered |
| 22:44.00 | *** join/#asterisk seanmh (n=johndoe@c-69-254-131-168.hsd1.nm.comcast.net) |
| 22:47.01 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 22:47.15 | *** join/#asterisk Micc (n=Micc@c-71-231-123-28.hsd1.wa.comcast.net) |
| 22:47.20 | *** join/#asterisk SomethingISODD (n=dan@69.168.155.251) |
| 22:47.55 | p3nguin | If a two-line Polycom can support 10 calls (5 calls on each of the two line registries), would a Cisco multi-line phone have the same capabilities? |
| 22:48.07 | [TK]D-Fender | DelphiWorld: * is NOT a "SIP router". |
| 22:48.18 | [TK]D-Fender | DelphiWorld: And registration has nothing to do with placing calls |
| 22:48.19 | SomethingISODD | hello Can anyone tell me is there way to see how many calls are connected and with how much duration? |
| 22:48.33 | [TK]D-Fender | SomethingISODD: "core show channels concise" |
| 22:48.43 | [TK]D-Fender | p3nguin: No. |
| 22:48.47 | SomethingISODD | thank you [TK]D-Fender |
| 22:49.02 | [TK]D-Fender | p3nguin: Because they don't dedicate any effort to their firmware |
| 22:49.25 | SomethingISODD | [TK]D-Fender one more question is there anyway to cut a call through the manage interface? |
| 22:49.37 | [TK]D-Fender | SomethingISODD: Yes |
| 22:50.02 | p3nguin | I've seen some suggestions to register both lines (on Cisco two-line phones) with the same user/secret to double the call capacity per "extension." Does that seem silly? |
| 22:50.09 | SomethingISODD | ok thanks. i will look it up just wanted to get a quick answer before i started looking |
| 22:51.12 | manxpower | p3nguin: it depends on what you want to do. |
| 22:51.22 | DelphiWorld | [TK]D-Fender: is registered;) |
| 22:52.06 | p3nguin | Basically I just want to be able to place one call on hold and dial a second call. With a two-line phone, pressing line2's button allows that if line2 has been configured. |
| 22:52.12 | manxpower | If you just want to toss calls at a phone and hope the phone does something logical with the call, then most phone support multiple lines with one registrations. But if you want total control over the call, including what line appearance the call shows up on, how to roll between lines, etc, then you want one registration per line. |
| 22:52.25 | [TK]D-Fender | p3nguin: they usually don't actually double-reg (which isn't sane). Normally it'd see that its the same target and logistcally span across it |
| 22:53.16 | p3nguin | If that's the case, then it seems perfectly reasonable to set both lines with the same credentials. |
| 22:53.21 | p3nguin | in my opinion. |
| 22:53.25 | manxpower | In my phone I have two line appearances (one registration) for my extension, one line / registration for my work DID and one line/registration for my personal DID |
| 22:56.22 | p3nguin | For ease of explanation, Asterisk will handle all DIDs and direct callers according to IVR responses (dialing a person's extension when prompted), and each user has a two-line Cisco phone and only needs one "extension." |
| 22:56.57 | p3nguin | So if you call the DID number, prompted for an extension, you dial 3001, Mary's phone rings. |
| 22:58.13 | manxpower | p3nguin: You'd be surprised at how users will insist on the most bizarre call routing you can imagine. |
| 22:58.15 | p3nguin | Mary decides she needs to call someone else for a second, so she presses the line2 button on the phone. That places the other call on hold and gives Mary a dial tone, where she can successfully complete an outgoing call. |
| 22:59.10 | p3nguin | The only way I can see to achieve that is A) give the phone a user/secret for each line, or B) use the same user/secret on both. |
| 22:59.30 | p3nguin | If what [tk]d-fender said was true, scenario B seems like a good idea. |
| 22:59.48 | *** part/#asterisk DelphiWorld (n=Miranda@196.20.95.121) |
| 23:00.04 | p3nguin | I don't really want to manage two peers for each phone, so I'll go with B. |
| 23:00.25 | p3nguin | Unless, of course, there is a good reason to NOT do it that way. |
| 23:02.10 | p3nguin | Now if I could get line1 and line2 to have distinctive rings, I might go with scenario A for a couple important phones. |
| 23:03.08 | *** join/#asterisk superbeef (n=IMP-IT@74.84.194.4) |
| 23:03.13 | manxpower | p3nguin: Ok. What do you want to have happen when Mary gets another call while she's already on with someone else? |
| 23:03.43 | *** join/#asterisk sahafeez (n=sahafeez@65-119-47-100.dia.static.qwest.net) |
| 23:04.28 | manxpower | BTW, Mary wants her phone to roll over to Jenny when she's not at her desk. But Mary wants the call to land in her voicemail if Jenny doesn't answer a rolled call. |
| 23:04.49 | p3nguin | manxpower: I think it depends on whether or not call waiting is turned on or off on the phone. If call waiting is on, line1's call waiting beep will sound in her ear. I am guessing if it is off, line2 would ring. |
| 23:05.02 | manxpower | Mark's boss thinks he's a slacker and so wants Mark's phone to roll to Mark's boss if Mark doesn't answer. |
| 23:05.49 | manxpower | p3nguin: starting out with one reg per phone is not a bad thing, just remember that's not the only way to do it and sometimes you'll have a need where it won't work. |
| 23:06.21 | p3nguin | I'm trying to explore the possibilities. |
| 23:07.05 | *** join/#asterisk Triplef_911 (n=Triplef_@70.82.147.168) |
| 23:08.21 | *** join/#asterisk slinksh0t (n=slinksh0@c-76-108-31-194.hsd1.fl.comcast.net) |
| 23:08.38 | Triplef_911 | anyway to disable the # for trx'es but still let the user transfer ? i dont want the incall # to messup data entrry |
| 23:08.54 | p3nguin | If I do not activate line2 with either a duplicated user/secret or with its own peer, is there another way to make a second outbound call? |
| 23:12.25 | manxpower | ~trixbox |
| 23:12.26 | infobot | rumour has it, trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org/. We do not recommend using it. |
| 23:12.53 | manxpower | Triplef_911: spell your words out. Yes, # transfer is disabled by default. |
| 23:13.29 | manxpower | The T/t/W/w options on the Dial line control DTMF based transfers and other DTMF features. |
| 23:13.43 | Triplef_911 | doesnt seem like it ;parkcall => **700 |
| 23:14.12 | Micc | can I put a hint on an IAX trunk call to a DID? Or does it have to be a single device to do a hint? |
| 23:14.13 | Triplef_911 | in features.conf... a fresh stop now and restart make no changes... # still does it.. i did include features in the context... for the trx to 700 etc |
| 23:14.13 | manxpower | Triplef_911: you are using the sample configs then. That's different. |
| 23:14.37 | manxpower | Triplef_911: The T/t/W/w options on the Dial line control DTMF based transfers and other DTMF features. |
| 23:14.44 | manxpower | Remove those options from your Dial lines. |
| 23:15.04 | Triplef_911 | ok then i cant park ? |
| 23:15.12 | manxpower | I did not say that. |
| 23:15.32 | manxpower | If you read the "core show application dial" you will see what specific letter what what specific thing. |
| 23:15.35 | Triplef_911 | ioh it's t,t |
| 23:15.37 | Triplef_911 | tT |
| 23:15.46 | manxpower | You can configure W/w to do transfers too. |
| 23:16.15 | manxpower | You had both t and T. You realise that means people calling into the system could transfer THEMSELVES, right? |
| 23:16.31 | Triplef_911 | t is blind trasnfer K is parking |
| 23:16.35 | manxpower | And, depending on the contexts, even transfer themselves outside the system. |
| 23:16.38 | Triplef_911 | yes |
| 23:16.53 | [TK]D-Fender | Triplef_911: What phones? |
| 23:16.57 | Triplef_911 | its only in the locals context.. not from inbound context |
| 23:17.04 | Triplef_911 | linksys sap942 and granstreams |
| 23:17.21 | [TK]D-Fender | Triplef_911: SCREW DTMF TRANSFERS. You don't need them |
| 23:17.35 | [TK]D-Fender | Triplef_911: You are asking for the proper wording for the wrong request |
| 23:17.36 | manxpower | Why don't you like the transfer features of those phones? |
| 23:17.51 | [TK]D-Fender | Triplef_911: these phones have their own NATIVE transfer features. DTMF = BS |
| 23:18.18 | [TK]D-Fender | Triplef_911: You do not need "#" for transfers or parking. |
| 23:18.24 | Triplef_911 | try to put tyour credit card number |
| 23:18.24 | manxpower | [TK]D-Fender: But all the cool kids do DTMF transfers! |
| 23:18.33 | Triplef_911 | or any ivr that needs a # at the end |
| 23:18.51 | manxpower | Triplef_911: I've never ever had Asterisk have a problem with # |
| 23:18.54 | [TK]D-Fender | manxpower: So long as I get to dump them into the vat of liquid nitrogen :D |
| 23:19.00 | Triplef_911 | i know... but users ..dial mastercard.. it asks for card plus #.. the # does a trx lol |
| 23:19.09 | [TK]D-Fender | Triplef_911: You should not be using "#" for transfers period |
| 23:19.18 | Triplef_911 | i know i want to disable |
| 23:19.20 | manxpower | Yes, but I use the transfer feature of my phone, not some silly DTMF transfer hack. |
| 23:19.22 | Triplef_911 | i removed t and k |
| 23:19.28 | [TK]D-Fender | Triplef_911: so stop doing Tt in your Dial() |
| 23:19.56 | Triplef_911 | hmm i can still trasnfer to other extensions ? |
| 23:20.06 | [TK]D-Fender | triYes |
| 23:20.10 | Triplef_911 | ah its for the features part only ok |
| 23:20.44 | [TK]D-Fender | Triplef_911: Only "features" you need to DTMF are recording, and triggered applicationmap |
| 23:22.09 | Triplef_911 | testing , thanks |
| 23:22.57 | p3nguin | How do you do parking without a DTMF transfer? If I blind xfer into 700, I don't hear which parking slot it is using. |
| 23:23.11 | Triplef_911 | use attended ttrx ? |
| 23:23.21 | Triplef_911 | no idea |
| 23:23.26 | p3nguin | I don't even know what a ttrx is. |
| 23:23.36 | [TK]D-Fender | Yes |
| 23:23.44 | [TK]D-Fender | Attended Transfers <- for parking |
| 23:23.56 | Triplef_911 | yeah its not working for my xlite at least |
| 23:23.57 | p3nguin | Let me try it. I thought it wouldn't work. |
| 23:24.41 | *** join/#asterisk sahafeez (n=sahafeez@65-119-47-100.dia.static.qwest.net) |
| 23:24.44 | p3nguin | well I'll be damned. |
| 23:25.06 | p3nguin | That worked far better than I expected it to. |
| 23:25.31 | [TK]D-Fender | Ok, stage time... later all |
| 23:26.25 | Triplef_911 | wow didnt work at all here |
| 23:27.39 | Triplef_911 | http://pastebin.ca/1624367 |
| 23:27.51 | Triplef_911 | weird.. it told me exten , then i hear music, then came back |
| 23:27.54 | Triplef_911 | and a zombie |
| 23:27.55 | p3nguin | When I hit transfer, it gave me a dial tone. I dialed 700, it said the slot number and gave me hold music. I pressed the transfer button again, and it disconnected. I was then able to retrieve the call by dialing the slot number that I was given. That's exactly how it works for a DTMF transfer, but without any DTMF. |
| 23:31.10 | Katty | has chinese foods. |
| 23:33.41 | *** join/#asterisk kaptengu (n=kaptengu@unaffiliated/kaptengu) |
| 23:34.05 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 23:34.09 | Triplef_911 | works #1 |
| 23:34.10 | Triplef_911 | hmm |
| 23:34.26 | Triplef_911 | p3nguin same scenario perfect all fixed |
| 23:34.40 | Triplef_911 | now i need a 8 horu drive to go show a chick how to do a transfer |
| 23:34.46 | Triplef_911 | s/horu/hour |
| 23:34.51 | p3nguin | sits by katty |
| 23:35.13 | Katty | gives p3nguin a fork, and shares. |
| 23:35.39 | p3nguin | Hope it's sweet and sour pork. That's my favorite. |
| 23:36.00 | Katty | general tso and honey chicken, actually. |
| 23:36.15 | Katty | they have this place here called Mongolian Grill Buffet |
| 23:36.27 | Katty | they offer cartons to go, for super cheap. 4 bucks. |
| 23:36.35 | Katty | all you can cram in there (= |
| 23:36.50 | p3nguin | I like the General's chicken, as long as it is made correctly. |
| 23:37.06 | p3nguin | I had some the other day that was just crap. |
| 23:37.16 | Triplef_911 | thanks all youv been helpfull |
| 23:38.56 | *** join/#asterisk Ad-Hoc (n=nimbus@ppp169-251.adsl.forthnet.gr) |
| 23:39.07 | p3nguin | We have a Mongolian Grill in the Rice 'n' Fries buffet. I wasn't pleased the last time I had them cook stuff for me. |
| 23:39.15 | *** part/#asterisk Triplef_911 (n=Triplef_@70.82.147.168) |
| 23:40.37 | p3nguin | I think I would rather have some fried squid or something. |
| 23:42.10 | Katty | :< |
| 23:42.14 | Katty | that sounds icky. |
| 23:42.40 | p3nguin | Oh no...it's good! |
| 23:44.05 | Katty | doesn't sound it |
| 23:47.37 | p3nguin | I can understand that. You hear squid, you think squishy sea creature with bad social habits. But when you take the meat and fry it, it's quite yummy. |
| 23:48.09 | *** join/#asterisk alexshell (n=abc@unaffiliated/alexshell) |
| 23:48.42 | *** join/#asterisk digilink (n=digilink@76.123.245.221) |
| 23:50.28 | jaytee | I love calamari with aioli dipping sauce |
| 23:56.58 | alexshell | hi all |
| 23:57.51 | alexshell | I'm just testing phpagi with the following example: Get DTMF tones from the user and say the digit |
| 23:58.03 | alexshell | code http://phpagi.sourceforge.net/phpagi2/docs/__examplesource/exsource__root_phpagi-2.14_examples_dtmf.php_9f0d08538805cb50bb0f290606fe78d3.html |
| 23:58.22 | *** join/#asterisk denon (i=denon@sassinak.net) |
| 23:58.22 | *** mode/#asterisk [+o denon] by ChanServ |
| 23:58.32 | alexshell | but I just receive the beeps |
| 23:59.01 | alexshell | I've already installed festival |
| 23:59.15 | alexshell | does someone have phpagi experience? |
