| 00:03.46 | *** join/#asterisk joobie (~joobie@202-161-27-151.dyn.iinet.net.au) |
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| 00:33.05 | antiwire | hey anyone have a patch for this? trying to get it going on ARM http://pastebin.slackadelic.com/p/Ngf59w45.html |
| 00:34.14 | WIMPy | o.O |
| 00:34.36 | WIMPy | wonders why the host CPU type matters for software running on the card. |
| 00:34.55 | antiwire | I mention that because of possible ARCH differences |
| 00:35.36 | WIMPy | was referring to the output you got. |
| 00:35.58 | antiwire | you mean this part? /usr/src/USBFXO/dahdi-linux-complete-2.4.0+2.4.0/linux/drivers/dahdi/Kbuild:98: CPU Architecture 'arm' does not support VPMADT032 or HPEC. Skipping. |
| 00:36.03 | antiwire | right |
| 00:36.07 | WIMPy | yes |
| 00:37.30 | antiwire | there has to be a patch somewhere, I just can't find it |
| 00:37.33 | coppice | HPEC runs on the host, so that makes sense. I think the VPM module has some host code as well as the code in the card, and that code comes as binary from Octasic |
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| 00:38.00 | WIMPy | not cool |
| 00:38.32 | antiwire | coppice: what are VPM and HPEC for? |
| 00:38.37 | antiwire | I'm just using a USB FXO module |
| 00:38.44 | antiwire | the Sangoma thing |
| 00:38.53 | coppice | VPM is an EC hardware module. HPEC is software EC |
| 00:39.27 | coppice | what kind of ARM hardware are you using with PCI slots? |
| 00:40.04 | antiwire | this doesn't have PCI slots |
| 00:40.17 | antiwire | it's q USB FXO module I'm building for |
| 00:40.19 | antiwire | q/a |
| 00:40.24 | coppice | when what use is dadhi? |
| 00:40.35 | antiwire | apparently it needs dahdi |
| 00:40.36 | coppice | oh, OK |
| 00:41.33 | coppice | have you seen anything which indicates that USB gadget is supported on ARM machines? |
| 00:41.57 | WIMPy | Why should that make a difference? |
| 00:42.02 | antiwire | I have not seen that explicitly |
| 00:42.22 | antiwire | oh damn it. |
| 00:42.37 | antiwire | running the Sangoma driver without dadhi as a test....: Error: Unsuppored architecture type armv5tel! |
| 00:42.41 | antiwire | well well... |
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| 01:02.26 | *** join/#asterisk Khratos (~jespinal@66.128.60.148) |
| 01:02.47 | Khratos | Good afternoon |
| 01:03.41 | Khratos | Is there a way of, from asterisk source directory, compile some channel driver before the whole Asterisk source? |
| 01:05.10 | *** join/#asterisk MoreAllLess (~Justo@cpe-76-90-240-2.socal.res.rr.com) |
| 01:05.22 | WIMPy | Good morning. Sot sure, what you're trying to do. You need the whole thing. |
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| 01:09.50 | Khratos | Yes, I know, but when compiling the chan_h323 channel driver, it generates some files, and then stop asking to re-run make (in order to use those files it generated) |
| 01:10.41 | Khratos | I wanted to know if it is possible to try to compile chan_h323 before the rest of the whole Asterisk, so it doesn't stop there |
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| 01:12.38 | pabelanger | Khratos: not without changing the build process for 1.4 |
| 01:12.46 | pabelanger | 1.6.2+ it has changed |
| 01:15.21 | Khratos | uuuumm.. wow, that'll force me to make some weird tweaks then |
| 01:16.05 | Khratos | If I do a : if ( ! make ); then make; fi , it will re-issue the 'make' command, but I wanted to do it as last resource (I'm building from a script) |
| 01:18.16 | Khratos | Thanks pabelanger and WIMPy , I'll look a work-around then |
| 01:23.10 | *** join/#asterisk antiwire (~antiwire@unaffiliated/antiwire) |
| 01:28.24 | Khratos | mmmm... just for the record (in this particular case), I could do: make menuelect.makeopts, set the env. variable GREP=/bin/grep, and after running build_tools/make_buildopts_h > include/asterisk/buildopts.h |
| 01:28.36 | Khratos | I can enter channels/h323 and do the make opt |
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| 01:54.34 | antiwire | well, it works on x86 |
| 01:55.04 | antiwire | I'll just use an x86 PC to terminate the ALS and send it to my main asterisk box over sip |
| 01:58.54 | *** join/#asterisk coppice (~chatzilla@m121-202-106-100.smartone-vodafone.com) |
| 02:15.38 | jaytee | considering installing fail2ban on my * box, anyone have suggestions or comments? |
| 02:22.13 | DoYouKnow | is g.729 the best codec? |
| 02:22.16 | DoYouKnow | (if available) |
| 02:22.26 | DoYouKnow | for low bandwidth |
| 02:23.17 | JerJer | LPC10 |
| 02:23.31 | JerJer | for the lowest bandwidth |
| 02:25.42 | Maliuta | DoYouKnow: depends on what you want g.792 is good quality audio over lower bandwidth |
| 02:26.17 | Maliuta | LPC10 is low b/w but quality may not be that great, you also need to consider what is supported at the other end of the link |
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| 04:49.54 | Siblor | Hello, does it possible to stream live conversation in channel to web application? |
| 04:55.37 | *** join/#asterisk corretico (~laguilar@201.201.44.82) |
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| 05:52.30 | CheBuzz_Home | !nat |
| 05:53.14 | CheBuzz_Home | Or is it... |
| 05:53.15 | drmessano | ~sipnat |
| 05:53.15 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, canreinvite, externhost or externip, and localnet. |
| 05:53.21 | CheBuzz_Home | That's it... |
| 05:53.51 | CheBuzz_Home | Ok, anybody know what bot software is powering infobot? |
| 05:54.18 | drmessano | ~about |
| 05:54.19 | infobot | http://www.infobot.org/ - /msg infobot help for a list of commands - http://www.infobot.org/guide-0.43.x.html for a guide |
| 05:54.50 | CheBuzz_Home | Thanks! |
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| 06:52.17 | ray_dominick | !hi |
| 06:53.37 | ray_dominick | Hi! I'm planning to do my academic project on VXML, and am downloading asterisk. I understand that I will need a VoIP based telephone. I was wondering if there is a Software replacement for it? I mean, Can I use any VoIP client software for development and testing? |
| 06:56.27 | Russ | ekiga |
| 06:56.31 | Russ | or just about anything |
| 06:57.17 | Russ | http://www.voip-info.org/wiki/view/Open+Source+VOIP+Software |
| 07:03.35 | ray_dominick | Thanks russ. I'm downloading it. It might take a couple hours (internet here sux). I'll try and configure it. |
| 07:04.01 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 07:06.31 | Siblor | Hello, does it possible to stream live conversation in channel to web application? |
| 07:07.45 | ray_dominick | Siblor, yes. try mibbit.com |
| 07:08.17 | Siblor | Thank you, I will check it out. |
| 07:09.02 | ray_dominick | Siblor, did you mean voice conversation? I'm sorry, that was for IRC. |
| 07:09.28 | ray_dominick | I was mistaken I was in #asterisk |
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| 07:18.33 | Siblor | Yes, I need to intercept and listen to the voices in specific channel. |
| 07:19.28 | *** part/#asterisk antiwire (~antiwire@unaffiliated/antiwire) |
| 07:20.00 | Siblor | ChanSpy can be use with the phone, but I also need web application to get the streaming audio file from Asterisk server as well. |
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| 08:38.28 | Weazel | I keep getting mails every hour with the error "mcelog: Cannot open /dev/mem for DMI decoding: No such file or directory" no idea what is that and why. can any1 help please ? |
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| 09:02.42 | helloritesh | Folks, has someone figured out how to detect DTMF inside Dial Macro "M()"? I am going nuts with it.. all commands execute properly but the "Read" just refuses to capture DTMF from the caller party |
| 09:03.08 | helloritesh | is there a issue /bug posted on this? |
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| 09:09.41 | helloritesh | http://pastebin.com/F3XmGgnE |
| 09:09.48 | helloritesh | That's a sample snapshot |
| 09:10.13 | helloritesh | looks really harmless but just refuses to work.. i have spent over 20 hours on these 5 lines and 100s of test calls.. no luck |
| 09:14.06 | *** join/#asterisk nicola_pav (~chatzilla@mail2.tikalnetworks.com) |
| 09:14.38 | nicola_pav | hello. i want to use a softphone installed on windows registered to Asterisk |
| 09:14.44 | nicola_pav | i want to use IPsec |
| 09:15.03 | nicola_pav | any open source IPsec clients on windows? |
| 09:17.08 | *** join/#asterisk pootietang (~pootietan@24-226-182-75.tr.cgocable.ca) |
| 09:19.24 | tamiel | nicola_pav: MS provides ipsec clients for windows (not opensource I think) |
| 09:19.55 | nicola_pav | tamiel: there are no free IPsec clients on windows? |
| 09:21.00 | kaldemar | doesn't windows have a built-in IPSec client nowadays? |
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| 09:23.00 | nicola_pav | windows xp? |
| 09:23.26 | *** join/#asterisk asterisk-Tester (~chatzilla@77.42.241.114) |
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| 09:40.05 | joachim_- | How can i hide callerid on outband calls? |
| 09:41.22 | Weazel | joachim_-: usually u can put "hidden" in the string but it may depend on ur provider |
| 09:41.57 | joachim_- | Weazel: thanks! will try that |
| 09:42.49 | Weazel | joachim_-: np |
| 09:42.58 | *** join/#asterisk ironm (~ironm@fwj00.e-fon.ch) |
| 09:43.34 | kaldemar | CALLERID function with datatypes num-pres and name-pres. values are probably the same as with the deprecated CALLERPRES function. |
| 09:44.19 | Weazel | kaldemar: thats true so |
| 09:44.23 | Weazel | joachim_-: u can put this > http://pastebin.com/UZ9aKvm9 |
| 09:45.17 | E-bola | Here in denmark it depends completely on ur provider |
| 09:45.29 | E-bola | asuming you want to call out to pstn |
| 09:46.18 | kaldemar | app SetCallerPres is also deprecated. use Set(CALLERID(num-pres)=...) or Set(CALLERID(name-pres)=...). |
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| 09:52.16 | Weazel | kaldemar: so hows the whole string looks like ? exten => _*43.,1,(CALLERID(num-pres)=(prohib_not_screened) ? |
| 09:53.54 | Weazel | or maybe Set(CALLERID(num-pres)=prohib) ? |
| 09:55.22 | kaldemar | the latter respects the syntax with braces. |
| 10:01.20 | Weazel | kaldemar: what do u mean by braces ? |
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| 10:09.21 | kaldemar | Weazel: (CALLERID(num-pres)=(prohib_not_screened) <-- this has one ( too many. and misses a "Set". |
| 10:10.43 | Weazel | kaldemar: ahh i see so its Set(CALLERID(num-pres)=prohib_not_screened) ? |
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| 10:14.22 | kaldemar | Weazel: yes. |
| 10:15.44 | kaldemar | app(func(datatype)=value) |
| 10:15.46 | Weazel | kaldemar: thx |
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| 10:47.02 | lera_zed | hello guys, I am experiencing strange problem with asterisk 1.6 - http://pastebin.com/BECCW6ft |
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| 11:44.58 | hrhrhr | 3. 2011-01-17 12:38:01DAHDI/1-1...501"501" <501>112ANSWERED00:24 |
| 11:45.07 | hrhrhr | how is that even possible?! |
| 11:59.24 | *** join/#asterisk schmidts (~schmidts@213.235.212.193) |
| 11:59.28 | schmidts | hello |
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| 12:07.37 | tuxx- | morninggg |
| 12:22.35 | *** join/#asterisk Faithful (~Faithful@202.189.73.144) |
| 12:26.03 | Faithful | Is there a way to get * to give you a tone when someone passes an assisted transfer to you, so that you know the other person is now on the line. |
| 12:27.07 | schmidts | Faithful there is an option AFAIK its called transferfinishtone |
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| 12:42.12 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:44.47 | chan_mobile | does anyone have found a way to correct the bluetooth bug 'hci0 SCO packet for unknown connection handle' ? |
| 12:45.14 | leifmadsen | chan_mobile: not a lot of people use chan_mobile in here afaik |
| 12:46.24 | chan_mobile | based on i've read this bug is not chan_mobile related, is kernel related. but i was unable to find the way to correct |
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| 13:33.58 | ChrisInSydney | hi there |
| 13:34.10 | ChrisInSydney | got a dependency question |
| 13:35.01 | ChrisInSydney | Installing 1.8 & I am assuming I will need resample to convert some SLN16 stuff to g722 wideband without losing quality |
| 13:35.36 | ChrisInSydney | will I need codec_resample ??? |
| 13:35.51 | ChrisInSydney | If so, its looking for resample(E) |
| 13:36.28 | ChrisInSydney | which I can not find in the CentOS respositories. Any clues. |
| 13:37.01 | ChrisInSydney | It might be like the butter in the fridge, I keep staring at it, but its next to the beer |
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| 14:17.40 | Katty | gooooooooooooooood morning!!! |
| 14:19.01 | WIMPy | Good afternoon |
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| 14:27.48 | visz | ehlo |
| 14:28.06 | pootietang | sup peeps |
| 14:28.18 | WIMPy | visz: Uset too much smtp by hand? |
| 14:28.25 | WIMPy | d |
| 14:28.30 | visz | =) |
| 14:28.44 | visz | or just enough |
| 14:29.18 | visz | any idea why rtp connects with delay when answering with a siemens gigaset c470 w/ latest firmware? |
| 14:29.56 | visz | i think i've had the same problem with another device, but can't remember the solution =/ |
| 14:30.26 | visz | there's about 10 seconds of silence before getting audio |
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| 14:45.55 | n3hxs[A] | Morning Katty |
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| 14:49.45 | *** mode/#asterisk [+o russellb] by ChanServ |
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| 14:55.06 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
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| 14:56.10 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:56.10 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:57.18 | *** join/#asterisk jkroon (~jkroon@dsl-241-245-183.telkomadsl.co.za) |
| 14:57.32 | jkroon | in SIP, the Call-ID value is only unique between two endpoints right? |
| 14:58.03 | jkroon | in other words, if I have three systems A, B and C with a call from A to C and B to C then in theory they could have the same Call-ID? |
| 14:58.36 | *** join/#asterisk cusco (~IRCLOIC@pcmedic.pt) |
| 14:58.40 | cusco | hi |
| 15:00.41 | cusco | We have two 4span digium cards |
| 15:00.43 | cusco | TE410P |
| 15:00.50 | *** join/#asterisk SeTTleR (~bernd@p5DDED2D1.dip.t-dialin.net) |
| 15:00.56 | cusco | one of the pri is like this: http://paste.debian.net/104863/ |
| 15:01.00 | cusco | what does this mean |
| 15:01.22 | russellb | jkroon: they could, but it should be incredibly unlikely |
| 15:01.53 | russellb | cusco: please contact Digium support - http://www.digium.com/en/supportcenter/ |
| 15:01.56 | cusco | also pri debug shows: http://paste.debian.net/104864/ |
| 15:02.07 | cusco | I thought perhaps what that meant |
| 15:03.13 | jkroon | russellb, cool. basically I'm trying to write something that sniffs my ethernet interface looking for RTP timing issues and then reporting the associated call. |
| 15:03.20 | jkroon | so if it can happen I should probably deal with it. |
| 15:03.52 | cusco | russellb: the card is working fin on two other PRI connections, this one is from a different telco |
| 15:03.55 | cusco | it has been working before |
| 15:04.01 | cusco | now it shows like that |
| 15:04.10 | russellb | jkroon: yeah |
| 15:04.26 | jkroon | thanks russellb - much appreciated. |
| 15:04.43 | russellb | np |
| 15:04.56 | jkroon | russellb, and I take it within the same conversation I can use the CSeq values to detect retransmits? |
| 15:05.39 | russellb | yeah, I think so |
| 15:05.52 | jkroon | (Damn D-Link routers drops certain 200 OK packets and I'd like to pro-actively look for this rather than getting a client "outbound calls drops after exactly 30 seconds") |
| 15:06.10 | jkroon | thanks russellb you've been very helpful. |
| 15:06.39 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 15:11.56 | pootietang | Im having problems with D-Link as well |
| 15:15.11 | *** join/#asterisk wesphillips (~wphillips@adsl-75-53-139-49.dsl.hstntx.sbcglobal.net) |
| 15:19.52 | Katty | hugs n3hxs |
| 15:21.26 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 15:24.37 | n3hxs | hugs Katty back. |
| 15:24.40 | n3hxs | and front. |
| 15:27.42 | *** join/#asterisk JustinCampbell (~JustinCam@dsl092-226-147.phl1.dsl.speakeasy.net) |
| 15:29.36 | Katty | n3hxs: :> |
| 15:31.55 | *** join/#asterisk russellb (~russell@asterisk/digium-open-source-team-lead/russellb) |
| 15:31.55 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:41.23 | *** join/#asterisk DaveCanoe (~Dave@2001:1928:0:ffff::2) |
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| 15:44.00 | cusco | hi |
| 15:44.07 | cusco | I have another question |
| 15:44.32 | cusco | using a .call file .. |
| 15:45.10 | cusco | channel is Local/90210332499@incoming-pri (it simulates all dialplan from a real PRI call, sets vars logs stuff etc) |
| 15:45.11 | *** join/#asterisk maxagaz (7b76504c@gateway/web/freenode/ip.123.118.80.76) |
| 15:45.18 | maxagaz | Hi |
| 15:46.41 | Katty | hai |
| 15:50.18 | *** join/#asterisk i_heart_asterisk (~chatzilla@firewall-a.buf.ny.i-evolve.net) |
| 15:50.22 | cusco | sorry, carrying on |
| 15:50.31 | *** join/#asterisk Circlefusion (~circlefus@74-138-194-204.dhcp.insightbb.com) |
| 15:50.57 | cusco | based on that number it goes directly to a specific ael.. at that moment that ael queues() the call and some one pics it up |
| 15:51.10 | cusco | but because there is no other party, it is automatically hangged up |
| 15:51.46 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 15:52.26 | cusco | now... I would like that the agent that pics up the call, to run some ael in the same channel re-using the channel variables, and that ael would put him in a conference call |
| 15:52.43 | cusco | Im having troubles with this bit |
| 15:54.44 | i_heart_asterisk | is there a way to disable VAD in asterisk? |
| 15:55.00 | i_heart_asterisk | im getting: |
| 15:55.06 | WIMPy | There is no way to enable it. |
| 15:55.07 | i_heart_asterisk | Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: from another asterisk box |
| 15:55.35 | WIMPy | did you read the part that says "on client"? |
| 15:55.56 | i_heart_asterisk | Client IP: = asterisk box |
| 15:56.23 | WIMPy | client = the other end |
| 15:56.37 | WIMPy | Err what? |
| 15:56.49 | WIMPy | From Asterisk to Asterisk? |
| 15:56.52 | i_heart_asterisk | yeah |
| 15:57.22 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 15:57.28 | WIMPy | That other Asterisk isn't behind the same IP as some of its clients is it? |
| 15:58.06 | i_heart_asterisk | WIMPy, here is call flow: lucent TNT -> asterisk-1 -> asterisk-2 |
| 15:58.16 | WIMPy | I would suspect that you're not talking to that other Asterisk but the media stream was redirected. |
| 15:58.27 | i_heart_asterisk | WIMPy, yes |
| 15:58.41 | i_heart_asterisk | asterisk-2 is getting the Comfort noise message |
| 15:58.47 | WIMPy | How is the Lucent connected to *1? |
| 15:58.58 | i_heart_asterisk | it says it's from the asterisk-1 IP |
| 15:59.16 | i_heart_asterisk | TNT sends calls to asterisk via fiber |
| 15:59.56 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 16:00.08 | WIMPy | What protocoll? |
| 16:01.16 | WIMPy | And what is the network situation like? |
| 16:03.00 | i_heart_asterisk | SIP |
| 16:03.41 | i_heart_asterisk | the TNT is practically right next asterisk-1 |
| 16:04.21 | WIMPy | still thinks *1 got out of the way. |
| 16:04.52 | i_heart_asterisk | WIMPy, what do you mean? got out of the way? |
| 16:05.23 | WIMPy | The good thing about SIP is that it can optimize the media flow. |
| 16:05.49 | leifmadsen | Asterisk has no VAD |
| 16:05.56 | leifmadsen | turn it off on the TNT |
| 16:06.04 | leifmadsen | otherwise, ignore the message |
| 16:06.13 | WIMPy | So that when *1 sends a call from the TNT to *2 the audio would go directly from the TNT to *2. |
| 16:08.25 | WIMPy | is just curious if there is NAT involved as well or if Asterisk gives a wrong message. |
| 16:10.59 | i_heart_asterisk | WIMPy, i suspected the VAD was coming from TNT but we have it disabled |
| 16:13.43 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
| 16:14.49 | WIMPy | Where does that call from the TNT come from? Some line? A local phone? |
| 16:15.03 | *** join/#asterisk Deeewayne (~dwayne@75-150-14-49-Atlanta.hfc.comcastbusiness.net) |
| 16:15.03 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 16:15.23 | *** part/#asterisk UQlev (~yuriy@212.50.99.8) |
| 16:15.35 | i_heart_asterisk | WIMPy a cell phone |
| 16:15.53 | i_heart_asterisk | TDM -> PRI |
| 16:15.54 | WIMPy | Via PSTN? |
| 16:16.08 | WIMPy | Ok. |
| 16:17.06 | WIMPy | So if you have VAD disables on the TNT, you should have done, what's needed. |
| 16:17.11 | WIMPy | NFI what's going on there. |
| 16:17.20 | *** join/#asterisk Tim_Toady (~moi@178.128.134.192.dsl.dyn.forthnet.gr) |
| 16:19.11 | i_heart_asterisk | yeah...effin TNT has so many options and so little support |
| 16:21.15 | *** join/#asterisk ketema (~ketema@kjhmacpro.ketema.net) |
| 16:26.03 | *** join/#asterisk babilen (~babilen@unaffiliated/babilen) |
| 16:26.04 | *** part/#asterisk babilen (~babilen@unaffiliated/babilen) |
| 16:26.48 | *** join/#asterisk ray_dominick (~dominick@59.164.98.180) |
| 16:30.19 | ray_dominick | Hi! I am trying to develop a VXML system. Can somebody tell me if asterisk can take user's voice as input? |
| 16:31.27 | *** join/#asterisk babilen (~babilen@unaffiliated/babilen) |
| 16:32.52 | *** join/#asterisk vinhdizzo (~vinh@pool-173-51-129-161.lsanca.fios.verizon.net) |
| 16:33.15 | russellb | ray_dominick: http://www.digium.com/en/products/software/lumenvox.php |
| 16:33.49 | russellb | or http://www.digium.com/en/products/software/vestec.php |
| 16:39.31 | *** join/#asterisk viraptor (~viraptor@212.11.65.66) |
| 16:40.18 | ray_dominick | russellb, is there a free version? GPL probably? |
| 16:40.26 | russellb | nope |
| 16:40.35 | russellb | though you can try your luck with Sphinx, |
| 16:40.51 | Katty | hugs russellb |
| 16:41.26 | russellb | hi Katty! |
| 16:41.31 | raden | hugs Katty |
| 16:41.31 | coppice | pocket sphinx isn't great, but then neither are the others |
| 16:43.56 | Katty | hugs raden |
| 16:44.00 | *** join/#asterisk m_tadeu (~quassel@89.181.27.165) |
| 16:45.12 | m_tadeu | hi...I'm using php for agi. my problem is I want to include a file that is not in the agi scripts director. is it possible to work around this? |
| 16:45.47 | raden | how is Katty today ? |
| 16:45.51 | raden | and how the weather down there ? |
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| 16:50.00 | *** join/#asterisk thehar (~thehar@diddlebox.thehar.com) |
| 16:54.37 | Katty | raden: doing a bit better today. weather is cold )= |
| 16:56.10 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
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| 16:58.25 | raden | good to hear |
| 16:58.42 | *** part/#asterisk babilen (~babilen@unaffiliated/babilen) |
| 16:59.39 | Katty | raden: how're you dear? |
| 17:00.04 | raden | Doing fine, cold and snowy up here, working on graphics arts stuff today ( blah ) |
| 17:00.13 | *** part/#asterisk viraptor (~viraptor@212.11.65.66) |
| 17:12.00 | *** join/#asterisk raden (~J@66-168-15-100.dhcp.stpt.wi.charter.com) |
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| 17:20.24 | *** part/#asterisk UQlev (~yuriy@212.50.99.8) |
| 17:20.55 | ray_dominick | russellb, I've seen the options. I might not be looking for full fledged voice recognition, rather recognizing simple words. Does anything exist for that? |
| 17:25.07 | frigidzephyr | As far as I am aware, recognizing simple words requires full fledged voice recognition |
| 17:25.40 | frigidzephyr | I'm not aware of any packages out there for Asterisk besides lumenvox and vestec |
| 17:25.45 | frigidzephyr | I"ve heard they both work well |
| 17:29.29 | *** join/#asterisk path (path@gateway/shell/bshellz.net/x-dwlvcnzjqbexeiqj) |
| 17:29.33 | *** part/#asterisk sekil (~sekil@80.93.247.26) |
| 17:29.37 | path | hey guys |
| 17:29.53 | path | I need some guidance regarding this: http://pastebin.com/gEXEb74x |
| 17:30.34 | path | the issue is, if someone tries to dial anything (dtfm) before sound3 finishes it goes to an invalid extension |
| 17:31.05 | path | s/dtfm/dtmf |
| 17:31.23 | *** part/#asterisk wesphillips (~wphillips@adsl-75-53-139-49.dsl.hstntx.sbcglobal.net) |
| 17:31.34 | path | any ideas will be appreciated |
| 17:34.31 | tzafrir | path, why don't you use the 'n' "priority"? |
| 17:34.49 | tzafrir | You missed priority 2 |
| 17:34.56 | tzafrir | s,2 |
| 17:35.13 | *** join/#asterisk blee (~blee@99-117-188-231.lightspeed.dybhfl.sbcglobal.net) |
| 17:35.23 | blee | is it possible to set the order that musiconhold plays files? |
| 17:37.46 | path | I noticed the missing '2' tzafrir |
| 17:37.57 | path | 'n' will it make a difference? |
| 17:41.53 | *** join/#asterisk JerJer (~JerJer@asterisk/original-h323-guy/JerJer) |
| 17:49.04 | tzafrir | path, the problem is that '2' is missing |
| 17:49.26 | path | I fixed tzafrir |
| 17:49.32 | JerJer | who stole the 2 ? |
| 17:49.44 | JerJer | :) |
| 17:49.45 | tzafrir | The one who fixed tzafrir |
| 17:49.54 | tzafrir | ;-) |
| 17:50.03 | path | but thats not the main issue, the problem is if dial anything before sound3 it goes to an invalid extension |
| 17:50.29 | path | in some way I need to not consider any dtmf |
| 17:50.38 | path | in a period of time |
| 17:51.49 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 17:53.38 | *** part/#asterisk bobdole369 (4694a992@gateway/web/freenode/ip.70.148.169.146) |
| 17:56.04 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 18:01.41 | *** join/#asterisk kalimc (~mcurry@proxy.hostopia.com) |
| 18:01.49 | Katty | sways |
| 18:03.22 | kalimc | When dialing a 11 digit number (with a leading 1) I am getting Called Number (len=14) [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) '19059180739' ] - however when I dial with a 10 digit I get the expected response: [Span 0 D-Channel 0]< Called Number (len=13) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '4165395610' ] -- whats with that? |
| 18:03.41 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 18:07.50 | kalimc | why do I get "Unknown Number Type" ? |
| 18:12.52 | raden | how can i allow directory listing in apache ? |
| 18:13.25 | leifmadsen | #apache? |
| 18:16.31 | tuxx- | lolz |
| 18:16.32 | tuxx- | ;d |
| 18:20.10 | tzafrir | raden, make sure you don't have a index file |
| 18:20.24 | tzafrir | and that DirectoryIndex is enabled there |
| 18:20.44 | tzafrir | and yeah, ask in #apache , I guess |
| 18:21.49 | *** join/#asterisk Dovid (Dovid@213.8.118.62) |
| 18:22.34 | raden | thanx |
| 18:24.14 | *** join/#asterisk lanning (~lanning@208.87.235.224) |
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| 18:36.52 | *** join/#asterisk kb3ien (~quassel@static-72-80-25-34.nycmny.fios.verizon.net) |
| 18:37.11 | kb3ien | i'm getting no love from queue application is there a demo of an all-dynamic members queue anywhere? |
| 18:39.45 | kb3ien | http://pastebin.com/bbLfu58x |
| 18:43.36 | _Corey_ | kb3ien: Have you tried to name the queue without a "." in it? |
| 18:44.16 | _Corey_ | there's nothing particularly special about an all-dynamic member queue |
| 18:48.56 | *** join/#asterisk oej (~olle@2001:470:1f15:d79:225:ff:fe44:74ec) |
| 18:57.42 | *** part/#asterisk ray_dominick (~dominick@59.164.98.180) |
| 19:02.15 | jaytee | is externnotify= in voicemail.conf a global setting or can it be set to a specific context or single mailbox? |
| 19:04.15 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
| 19:05.55 | kb3ien | is . a reserved char? |
| 19:07.06 | _Corey_ | Don't know, just looked strange |
| 19:07.26 | _Corey_ | I've never used anything more exotic than an _ or - in a queue name |
| 19:08.20 | leifmadsen | same |
| 19:08.43 | leifmadsen | kb3ien: would be best to just simplify the name and find out if that is the issue |
| 19:10.26 | kb3ien | here goes |
| 19:12.53 | kb3ien | heh typo in my SIP 'addrr never mind... |
| 19:13.39 | paulc | What's the big fat obvious thing I need to have installed in order for res_jabber to be available in make menuselect? |
| 19:19.52 | leifmadsen | paulc: iksemel |
| 19:20.04 | leifmadsen | ~newbook |
| 19:20.04 | infobot | Help review the 3rd edition of the Asterisk book published by O'Reilly at http://ofps.oreilly.com/titles/9780596517342/ Released under a Creative Commons license: http://creativecommons.org/licenses/by-nc-nd/3.0/us/ |
| 19:20.09 | leifmadsen | documents res_jabber |
| 19:20.39 | paulc | @leifmadsen: Yeah, that's what I thought.. installed fine on Ubuntu, and CentOS is giving me a headache (for a change - it pisses me off) |
| 19:21.00 | leifmadsen | paulc: well iksemel is only available from a 3rd party repo I believe on CentOS |
| 19:21.07 | leifmadsen | check the docs because it should tell you how to do it |
| 19:21.50 | paulc | doffs hat to @leifmadsen - once again, a gentleman and a scholar - thanks :-) |
| 19:21.58 | leifmadsen | :) |
| 19:23.58 | JustinCampbell | bootstrap and ec2 instance: https://gist.github.com/raw/d781d6a48a95f821fdfb/c547ffd86579b585eb949ab28d338efa19cc5f06/bootstrap.sh |
| 19:24.13 | JustinCampbell | an* |
| 19:28.03 | paulc | @leifmadsen: Download source for iksemel, compile, install, and now my res_jabber is available for compilation. Thanks again :-) Off to 2 hours of meetings but I might hit you up later with a Jabber_Status related question ;-) |
| 19:30.50 | leifmadsen | :) |
| 19:33.43 | *** join/#asterisk Buttons840 (~buttons@c-24-10-183-136.hsd1.ut.comcast.net) |
| 19:38.46 | Buttons840 | anyone have experience sending text messages from asterisk to cell phones (say, a typical verizon cell phone)? |
| 19:42.56 | leifmadsen | probably needs to be done outside of asterisk |
| 19:43.13 | leifmadsen | in the US, you would probably have to use an email-to-sms gateway |
| 19:47.11 | Buttons840 | leifmadsen: thanks for your interest; asterisk has a sms application i see; i thought it might be useful but i expect you would need a correct service provider (forgive my loose terminology); perhaps no such providers exist? otherwise the email-to-sms is pretty simple given i can get a phone number and know which service provider they have (at&t or verizon, etc) |
| 19:47.38 | leifmadsen | Buttons840: that is only useful in Europe on specific carriers |
| 19:47.51 | leifmadsen | you can't use app_sms in North America |
| 19:49.18 | Freeaqingme | leifmadsen, why is that? |
| 19:50.51 | leifmadsen | Freeaqingme: because that's how it was programmed |
| 19:51.17 | Freeaqingme | but is the US protocol for texts so much different from what they use in in the eu? |
| 19:51.34 | leifmadsen | I don't know |
| 19:51.44 | Freeaqingme | kk |
| 19:53.21 | Buttons840 | i see "sms gateway providers" are a service you can buy -- i'm still researching but any comment on these? |
| 19:53.52 | leifmadsen | I have no further comments |
| 19:54.09 | carrar | Buttons840, please write one, k thanks |
| 19:54.14 | Freeaqingme | they are indeed a service. but it _may_ be that your voip trunkprovider can also act as sms gateway |
| 19:55.25 | chuckf | recalls something about Vitelity starting to offer SMS services a few months back.... |
| 19:57.08 | Buttons840 | carrar: what do you mean? |
| 19:57.23 | Buttons840 | actually, i'm probably getting off topic, so nevermind; thanks all |
| 19:57.44 | carrar | What do you mean by what do you mean? |
| 19:59.15 | Buttons840 | carrar: you said "please write one"; write what? |
| 20:00.23 | carrar | Write what you were talking about |
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| 20:13.58 | helloritesh | <PROTECTED> |
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| 20:47.25 | twanny796 | what happened to x-lite for linux, any links? |
| 21:04.09 | keith4 | ugh. x-lite is a miserable pile of code. |
| 21:05.25 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 21:05.58 | _Corey_ | My licensed copy of Bria has a "Show Advertisements" option. Ads are nice. Everyone likes ads. |
| 21:12.16 | Qwell | _Corey_: there was actually a study done, and they found that some people actually like seeing ads.. as odd as that sounds. |
| 21:12.47 | _Corey_ | I guess if we needed any proof the human race is doomed... |
| 21:14.06 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
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| 21:26.42 | twanny796 | sip protocol book? |
| 21:33.48 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 21:33.53 | leifmadsen | ... http://www.ietf.org/rfc/rfc3261.txt ? |
| 21:36.45 | Pan3D | Qwell: what study? |
| 21:36.59 | Qwell | Pan3D: dunno, something I read about randomly |
| 21:37.03 | leifmadsen | 7 out of 10 studies never happened |
| 21:37.10 | Pan3D | is suspicious |
| 21:37.26 | leifmadsen | 87% of stats are also made up |
| 21:37.43 | Pan3D | was it like "Do you prefer A) ads in software, B) getting your nuts chopped off" |
| 21:38.24 | russellb | which nuts? I do like having my peanuts chopped off the vine before they get to me |
| 21:38.35 | russellb | hand picking them off the roots is a pain |
| 21:38.41 | russellb | B. |
| 21:38.54 | leifmadsen | +1 |
| 21:39.34 | *** join/#asterisk timholum1 (~chatzilla@68-117-120-138.static.eucl.wi.charter.com) |
| 21:40.23 | _Corey_ | :) |
| 21:41.55 | timholum1 | hello, I have an asterisk phone system, I have it set up to record every phone call, the state I live in only requires single party notification, is it legal to record a phone call with a company in a state that reqires 2 party noticfication? |
| 21:42.29 | Qwell | timholum1: consult your lawyer |
| 22:00.40 | *** join/#asterisk ChkDigit (~mike@static24-72-137-23.r.rev.accesscomm.ca) |
| 22:06.17 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
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| 22:16.17 | Corydon76-home | timholum1: does your company have a physical presence in the state? |
| 22:17.31 | *** join/#asterisk E-bola (~bola@188.120.76.228) |
| 22:17.47 | Corydon76-home | timholum1: if not, federal law takes precedence, since you're talking about interstate commerce. |
| 22:18.34 | Corydon76-home | If you do, better ensure you have complied with 2-party notification |
| 22:21.59 | Corydon76-home | and by physical presence, I mean anything that could possibly tie you into that state. For example, a store, a building, a storage rental, a computer server... |
| 22:22.49 | Corydon76-home | A contract whose terms constrain you to sue for breach only in that state |
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| 22:39.09 | citywok | does anybody ahve experience with the plantronics s11/s12? how's durability/reliability? |
| 22:41.05 | *** join/#asterisk golikwid|mac_ (~chrislees@rrcs-67-78-200-57.se.biz.rr.com) |
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| 22:55.48 | *** join/#asterisk pecenipicek (~pecenipic@cpe-109-60-81-151.zg3.cable.xnet.hr) |
| 22:55.52 | pecenipicek | g'day |
| 22:56.46 | pecenipicek | got a question. how can i get asterisk to determine on which tech is the extension that is called on? |
| 22:57.23 | *** join/#asterisk ariel_ (~chatzilla@c-98-254-138-7.hsd1.fl.comcast.net) |
| 22:57.41 | pecenipicek | i have 2 boxes set up and running and sip calls going back and forth without problem, but how can i get it to check registered phones and determine if it should call the phone using SIP or SCCP? |
| 23:01.17 | *** join/#asterisk chasing`Sol (~rc4@smtp.master-zone.net) |
| 23:02.09 | *** join/#asterisk Tozz_ (Tozz@hardwire.duocast.net) |
| 23:06.31 | *** join/#asterisk sekil (~Ognjen@78.24.104.73) |
| 23:07.04 | _Corey_ | pecenipicek: You could dial both simultaneously, one will just fail |
| 23:07.35 | _Corey_ | i.e: Dial(SIP/1234&DAHDI/g1/1234567,28,r) |
| 23:07.37 | pecenipicek | that is bound to make the CDR fill up the db rather... quickly, wouldnt you say? :p |
| 23:07.49 | pecenipicek | i'm looking for a bit more elegant solution |
| 23:08.21 | *** join/#asterisk guilhermebr (~Guilherme@189.63.56.86) |
| 23:08.21 | _Corey_ | It's one call, so the one that works will get logged |
| 23:08.22 | pecenipicek | since i can pretty much match the number arbitrarily, i find it odd that i cant match the tech also... |
| 23:08.31 | _Corey_ | it's not like it's spawning two real calls |
| 23:08.45 | pecenipicek | my current call exten is like this. exten => _XXX,n,Dial(SIP/${EXTEN},10) |
| 23:09.05 | _Corey_ | You could check ${DIALSTATUS} |
| 23:09.12 | _Corey_ | on CHANUNAVAIL, dial SCCP |
| 23:09.18 | _Corey_ | or vice-versa |
| 23:09.25 | pecenipicek | mm. |
| 23:09.30 | pecenipicek | thats definitely a possibility. |
| 23:10.14 | citywok | http://www.voip-info.org/wiki/view/Asterisk+standard+extensions#app_devstate |
| 23:14.08 | pecenipicek | basically the problem is that the call attempt doesnt die when it tries to dial the SCCP phone via SIP. |
| 23:15.11 | Chainsaw | pecenipicek: Is this Asterisk 1.8? |
| 23:15.18 | pecenipicek | 1.6 |
| 23:15.28 | pecenipicek | wouldnt be using SCCP if i was on 1.8, now would i? :p |
| 23:15.35 | Chainsaw | pecenipicek: ChanAvail wasn't broken for SIP there yet, _Corey_'s suggestion should work. |
| 23:15.52 | Chainsaw | pecenipicek: No idea, haven't bothered with SCCP in a long time. |
| 23:16.09 | pecenipicek | the problem is that the SIP attempt of the call doesnt die when it hits an cisco phone. |
| 23:16.33 | pecenipicek | its trying to call, the server waits till it times out internally (20 seconds), then complains that it didnt work. |
| 23:17.06 | Chainsaw | pecenipicek: You could try SIPPEER functions instead. |
| 23:17.32 | Chainsaw | pecenipicek: If the SIPPEER has no context, you should probably be using SCCP for that peer instead. |
| 23:17.45 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 23:19.10 | pecenipicek | despite the fact that the information for a registered phone should be availible and accessible via other paths? |
| 23:19.44 | Chainsaw | pecenipicek: I'm the last person to ever call Asterisk perfect. That workaround has helped me out in the past. |
| 23:19.55 | Chainsaw | pecenipicek: So you may want to try it. |
| 23:20.17 | pecenipicek | its on the list of things to try when brickwall is thoroughly encountered. |
| 23:20.46 | pecenipicek | hm. |
| 23:21.02 | pecenipicek | another option is doing it partly via pattern matching and a goto command. |
| 23:21.18 | *** join/#asterisk atommat (~atommat@atommat.plus.com) |
| 23:21.23 | pecenipicek | but that would neccesitate manually punting all extensions into the extensions.conf file. |
| 23:21.36 | pecenipicek | so that kindof defeats the purpose of pattern matching in my head... |
| 23:21.40 | pecenipicek | hrrmmm.... |
| 23:22.45 | atommat | im having a real headache trying to get dialout to work with my b410p. i can dial in fine but im just not understanding channels, extensions and contexts too well. question to follow. |
| 23:24.07 | atommat | firstly, the extensions.conf has globals where there is a line: span_1 = DAHDI/g0 and span_2 = DAHDI/g1 |
| 23:24.11 | atommat | is this correct? |
| 23:25.07 | atommat | and how does this translate into a dialplan? |
| 23:26.23 | pecenipicek | if you are running asterisk, go to the console and write "dialplan show" (without the quotation marks). i think that that might answer your question a bit |
| 23:26.38 | atommat | im also not sure where the g0 and the g1 is defined. where can i see this realting back to the spans in the dahdi config? |
| 23:28.39 | *** join/#asterisk LemensTS (~matthew@adsl-70-238-158-17.dsl.stlsmo.sbcglobal.net) |
| 23:30.17 | atommat | '_9!' => 1. Macro(trunkdial-failover-0.3,${span_1}/${,${EXTEN:1})},,span_1,) [pbx_config] |
| 23:30.21 | atommat | this doesnt look right to me |
| 23:30.35 | atommat | any suggestions? |
| 23:30.44 | atommat | i could do without the failover |
| 23:32.41 | LemensTS | what do you think the best tts is on asterisk? |
| 23:37.28 | pecenipicek | i think its time i read up on the availible functions in asterisk |
| 23:47.45 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 23:59.44 | citywok | hmm. asterisk-mib and digium-mib.txt have disappeared from the 1.8 releases. the doc/ folder appears to have been cleaned out after 1.6.2.xx. any reason/ |