| 00:15.20 | *** join/#asterisk jsidhu (~jsidhu@c-76-103-82-26.hsd1.ca.comcast.net) |
| 00:16.08 | *** join/#asterisk resist0r (~resist0r@69.31.131.51) |
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| 00:45.37 | jermey_g | hi |
| 00:45.54 | WIMPy | lo |
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| 00:56.36 | *** part/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 00:57.10 | csd-199 | Hi. I have a CentOS 6 x64 and want to yum asterisk... when will I be able to do that? |
| 00:57.31 | *** part/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 01:00.02 | *** join/#asterisk coppice (~chatzilla@183178203098.ctinets.com) |
| 01:04.58 | csd-199 | Hi. I have a CentOS 6 x64 and want to yum asterisk... when will I be able to do that? |
| 01:05.18 | WIMPy | After you found out, how. |
| 01:09.50 | *** join/#asterisk darkbasic (~quassel@host37-245-static.119-2-b.business.telecomitalia.it) |
| 01:10.40 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-yughepvzswxheddq) |
| 01:13.03 | csd-199 | ok, how? |
| 01:16.01 | jsidhu | http://pastebin.com/raw.php?i=60TvB0XH anyone have a few minutes? I can't get any of my SIP devices to register.. |
| 01:18.36 | *** join/#asterisk celord (~cesar@celord.ice.co.cr) |
| 01:24.37 | pabelanger | jsidhu: nice debug log, just missing one thing, sip.conf for your peer settings |
| 01:25.22 | jsidhu | ah |
| 01:26.22 | jsidhu | http://pastebin.com/raw.php?i=WCAVXgGG |
| 01:29.00 | jsidhu | does that give any clues? |
| 01:29.36 | *** join/#asterisk coppice (~coppice@m121-202-32-188.smartone.com) |
| 01:32.02 | *** join/#asterisk random800 (442596fb@gateway/web/freenode/ip.68.37.150.251) |
| 01:32.46 | random800 | Hey guys, I have a stupid problem with asterisk |
| 01:32.47 | pabelanger | jsidhu: Hmm, this might be a bug. This does not look right: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4f18baa6" |
| 01:32.59 | pabelanger | notice the space / tab in nonce |
| 01:33.02 | random800 | I have upgraded it from 1.6 to 1.8.8 |
| 01:33.48 | random800 | now it does not start. Any ideas on how to debug this? |
| 01:33.50 | pabelanger | jsidhu: what if you try type=peer |
| 01:33.53 | jsidhu | yes, i see it WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4f18baa6" |
| 01:33.56 | jsidhu | let me see |
| 01:34.08 | pabelanger | random800: is asterisk crashing? |
| 01:34.14 | pabelanger | and how are you starting it |
| 01:35.02 | random800 | I am using ubuntu. I tried to ways to start it: 1. sudo service asterisk start, and 2. sudo asterisk |
| 01:35.30 | jsidhu | pabelanger: same issue. 401 Unauthorized and I still see the tab in the nonce, nonce="07ad126\t" |
| 01:35.47 | random800 | I can't tell if it crashes or exits because of some bad config (left from the previous version) |
| 01:36.19 | random800 | in dmesg the last line is: [Jan 19 20:10:57] WARNING[15090] translate.c: empty buf size, you need to supply one |
| 01:39.14 | pabelanger | jsidhu: what OS are you using? |
| 01:39.32 | jsidhu | debian wheezy/sid on arm. |
| 01:39.38 | pabelanger | Hmm |
| 01:39.57 | jsidhu | check my earlier post header http://pastebin.com/raw.php?i=60TvB0XH |
| 01:40.07 | jsidhu | wheezy/sid |
| 01:40.17 | jsidhu | on ARM |
| 01:40.26 | *** part/#asterisk csd-199 (~asdf@189.237.68.82) |
| 01:41.14 | jsidhu | brb |
| 01:41.56 | *** join/#asterisk jsidhu (~jsidhu@c-76-103-82-26.hsd1.ca.comcast.net) |
| 01:42.00 | jsidhu | back |
| 01:42.21 | pabelanger | jsidhu: able to pb your config.log file? |
| 01:42.39 | jsidhu | sure |
| 01:43.11 | jsidhu | what exactly do u need? I could trim and start new, from what point on do u want to see the log |
| 01:43.38 | jsidhu | everything, from startup to register? with verbose/debug set to high? I think 255 is max? |
| 01:44.12 | pabelanger | jsidhu: no, the config.log. The output from ./configure |
| 01:44.14 | pabelanger | actually |
| 01:44.23 | pabelanger | I don't think you have it, a package install right? |
| 01:44.32 | jsidhu | sorry, yes |
| 01:44.59 | jsidhu | maybe there's a -dev package |
| 01:45.26 | jsidhu | http://pastebin.com/raw.php?i=xM9mjp5K |
| 01:45.28 | pabelanger | jsidhu: ls -la /dev/urandom |
| 01:45.31 | pabelanger | what is the output |
| 01:45.53 | jsidhu | crw-rw-rwT 1 root root 1, 9 Jan 19 00:55 /dev/urandom |
| 01:46.10 | pabelanger | k |
| 01:52.36 | jsidhu | yea i don't think the config.log is provided as part of the -dev package, only the headers and other include files |
| 01:55.17 | *** join/#asterisk coppice (~coppice@m121-202-46-227.smartone.com) |
| 01:57.13 | pabelanger | jsidhu: do you have a cleaner debug.log? It looks like the output is wrapped or something |
| 01:58.49 | *** join/#asterisk resist0r (~resist0r@69.31.131.51) |
| 02:00.19 | pabelanger | random800: try starting asterisk via the command line, sudo asterisk -vvvvvc |
| 02:04.26 | pabelanger | random800: please don't mesg me, just make your posts here. More people will help |
| 02:04.32 | pabelanger | ~collectdebug |
| 02:04.33 | infobot | somebody said collectdebug was a method of collecting logs allowing others help troubleshoot an issue. Refer to https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 02:04.38 | pabelanger | random800: ^ start here |
| 02:04.42 | jsidhu | pabelanger: I think I'm going to step back and try the older release, Squeeze. I think Wheeze is too bleeding edge |
| 02:05.18 | pabelanger | jsidhu: do me a favor, create an issue in JIRA and upload your debug log. It is defiantly a bug in asterisk |
| 02:05.28 | random800 | <PROTECTED> |
| 02:06.01 | jsidhu | ok |
| 02:06.29 | WIMPy | jsidhu, pabelanger: I'm sure I have seen that before. |
| 02:06.31 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 02:06.36 | WIMPy | But I don;t know if it was here or on Jira. |
| 02:07.26 | pabelanger | WIMPy: I seen it on asterisk-user mailing list a while ago. |
| 02:07.41 | *** join/#asterisk jsidhu_ (~jsidhu@c-76-103-82-26.hsd1.ca.comcast.net) |
| 02:07.53 | jsidhu | at what verbosity do you guys recommend for the logs? |
| 02:08.17 | WIMPy | pabelanger: I don't read that. |
| 02:10.33 | WIMPy | jsidhu: Is the platform big or little endian? |
| 02:10.36 | *** join/#asterisk F2Knight (~ben@c-98-246-210-98.hsd1.or.comcast.net) |
| 02:12.41 | jsidhu | Im not sure |
| 02:12.44 | jsidhu | let me check |
| 02:14.25 | F2Knight | Q: working on trying to get a channel bank working.... |
| 02:14.26 | jsidhu | The device supports both Big Endian and Little Endian byte ordering, as defined in the ARM |
| 02:14.27 | jsidhu | Architecture Reference Manual, Second Edition. It also supports hardware features for performing |
| 02:14.27 | jsidhu | data conversion on some of its interfaces |
| 02:14.49 | jsidhu | thats from the cpu func. spec pdf |
| 02:15.00 | WIMPy | Yes, I know ARMs do both. That's why I ask what your architecture uses. |
| 02:15.02 | F2Knight | Would I be correct to assume that I should From asterisk... configure my second port as an fxsks to send to the channel bank? |
| 02:15.31 | jsidhu | Debian currently only supports little-endian ARM systems. |
| 02:15.35 | jsidhu | so its Little Endian |
| 02:15.38 | WIMPy | (at least the modern ones) |
| 02:15.53 | jsidhu | Sorry, I was copy/pasting as I was figuring it out myself. |
| 02:16.05 | WIMPy | Ok, so that's not the cause of the issue then. |
| 02:16.38 | F2Knight | it was working on a 1.2 box as fxoks.. but on the 1.8 it does not give dialtone.. tried for giggles to set it as fxs and while it got not dialtone or ring I could still get a call through it |
| 02:16.52 | F2Knight | *confused* |
| 02:17.29 | jsidhu | maybe I should ditch the debian package and recompile |
| 02:17.53 | WIMPy | Surely worth a try. |
| 02:18.03 | jsidhu | only if the issue has been fixed, i should check jira |
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| 02:36.14 | pabelanger | jsidhu: cat /proc/cpu/alignment |
| 02:36.16 | pabelanger | if you don't mind |
| 02:37.27 | jsidhu | Just read this on plugpbx forums regarding asterisk 1.8 on arm: -md5sum math/logic broken. This breaks SIP Authentication (can't use passwords). |
| 02:37.36 | jsidhu | rebooting the device, give me one sec |
| 02:37.49 | pabelanger | jsidhu: link? |
| 02:39.46 | jsidhu | http://forums.plugpbx.org/index.php/topic,234.0.html |
| 02:39.55 | jsidhu | http://forums.plugpbx.org/index.php/topic,234.msg1110.html#msg1110 |
| 02:41.33 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 02:43.02 | jsidhu | cat /proc/cpu/alignment ---> http://pastebin.com/raw.php?i=iMDqtHqh |
| 02:43.42 | pabelanger | jsidhu: thank, attach that to the JIRA issue too if you don't mind |
| 02:55.36 | *** join/#asterisk ninor (~ninor@50.34.232.22) |
| 02:56.09 | ninor | can asterisk handle video over ip too? |
| 02:56.10 | ninor | or just voice |
| 02:56.13 | WIMPy | jsidhu: What arm is it exactely? |
| 02:56.46 | WIMPy | ninor: As far as forwarding goes. |
| 02:56.58 | ninor | ok |
| 02:57.00 | ninor | so just voice then |
| 02:57.15 | WIMPy | Did I say that? |
| 02:57.24 | jsidhu | CPU - Marvell Kirkwood 88F6281 @ 1.2GHz |
| 02:57.38 | jsidhu | http://www.marvell.com/embedded-processors/kirkwood/assets/88F6281-004_ver1.pdf |
| 02:58.11 | WIMPy | Guruplug or something? |
| 02:58.53 | ninor | WIMPy, pardon me. i must have misunderstood. please clarify |
| 02:59.09 | WIMPy | It says ARMv5, so that can't do misaligned acces. |
| 02:59.34 | WIMPy | ninor: It can forward video, but it cannot process it. |
| 03:00.03 | WIMPy | Except for the very limited ConfBridge support, which just switches by talk detection. |
| 03:01.09 | jsidhu | Dreamplug |
| 03:01.42 | ninor | could it forward around h.264 video calls? |
| 03:01.49 | ninor | basically i wanna replace skype, and i need a server |
| 03:01.52 | ninor | i can code the client myself |
| 03:02.24 | WIMPy | I don't think there's any restriction on what can be forwarded. |
| 03:02.40 | ninor | thanks |
| 03:02.41 | *** part/#asterisk ninor (~ninor@50.34.232.22) |
| 03:03.43 | WIMPy | should also play around with plugs or minicubes. But I already have trouble to find time for anything :-( |
| 03:05.50 | jsidhu | ok, i've collected all the logs, and will create a issue on jira later, I'm gonna try and work with .16 |
| 03:05.51 | jsidhu | 1.6 |
| 03:06.19 | WIMPy | Mention that it's an ARMv5 |
| 03:06.24 | jsidhu | ok |
| 03:06.29 | WIMPy | Or ARMv5TE exactely |
| 03:06.38 | jsidhu | ok |
| 03:06.56 | WIMPy | But I don't think the extensions matter in any way. |
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| 04:50.46 | damex | hello, im was using now elastix for asterisk (used before debian and gentoo with asterisk) and i wanted to interest about functionality that i want. for example, ppl calling 153 and adding number like 153*9 (directly from phone) and then call going automatically to extension "9". how can i do that? |
| 05:40.31 | *** join/#asterisk reatoik (~atoik@217.10.42.164) |
| 05:44.53 | [TK]D-Fender | damex: You haven't explained what the 153 does or how there is an "extension" after it. |
| 05:46.12 | damex | [TK]D-Fender, its number on analog line on FXO port. and extension is the extension for the USER sip phone. |
| 05:47.00 | [TK]D-Fender | damex: Please redescribe the entire scenario |
| 05:47.02 | *** join/#asterisk zopiac (~zopiac@c-68-40-13-61.hsd1.mi.comcast.net) |
| 05:49.42 | damex | [TK]D-Fender, i have now "analog telephone call to 153 number => FXO port on openvox a800e from system with asterisk => IVR menu => select 9 => go straight to user's extension number 9". what i want "analog telephone call to 153*9 number => FXO port on openvox a800e from system with asterisk => go straight to user's extension number 9" |
| 05:49.55 | F2Knight | When connecint to a channel bank device over T1 Interfaces... should asterisk be the Fxs or the Fxo? The channel bank connects to real telephones, and then back to asterisk over a Dual Port T1, (span2) Span1 already connects to a pri |
| 05:50.39 | [TK]D-Fender | F2Knight: * is the FXO |
| 05:50.52 | [TK]D-Fender | * = office, phones = stations |
| 05:51.14 | p3nguin | *headscratch* |
| 05:51.45 | p3nguin | Phones have FXO ports in them. |
| 05:52.30 | F2Knight | [TK]D-Fender, that is what I thought.. but does that mean in dahdi/system.conf I should define the 2nd span like fxoks = 25-48, or fxsks = 25-48? |
| 05:53.01 | F2Knight | Currently it is at fxoks.. but I get no dialtone on the channel bank and when I send a call over it gets no ringing. |
| 05:55.38 | damex | asked about trouble and gone :( |
| 05:55.56 | F2Knight | ?? |
| 05:55.56 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 05:56.42 | [TK]D-Fender | F2Knight: Pb your configs and CLI status dump |
| 06:02.47 | *** join/#asterisk singler (~singler@84.15.129.49) |
| 06:04.39 | F2Knight | [TK]D-Fender, http://pastebin.com/mAU4xn0X |
| 06:06.14 | [TK]D-Fender | F2Knight: You masked bit from chan_dahdi. Don't. |
| 06:06.17 | [TK]D-Fender | new PB |
| 06:07.22 | F2Knight | [TK]D-Fender, I removed the caller ID's and all the duplicate lines from chan_dahdi |
| 06:07.34 | [TK]D-Fender | Also include the proper status dump from CLI that I requested |
| 06:08.12 | [TK]D-Fender | F2Knight: I want to see what you really have, not some hacked up version |
| 06:10.48 | F2Knight | [TK]D-Fender, http://pastebin.com/gU1FyTSL Thats the full chan_dahdi |
| 06:11.29 | [TK]D-Fender | signalling=auto <- should be fxo_ks |
| 06:11.56 | [TK]D-Fender | and you didn't set any standard parms for 3-way calling, cid, transfer, etc. |
| 06:12.11 | [TK]D-Fender | Doesn't look like you tried specifying what you were configuring |
| 06:13.40 | [TK]D-Fender | F2Knight: And still no status dump |
| 06:14.00 | F2Knight | status dump? |
| 06:14.13 | F2Knight | I pasted the CLI output is that what you were wanting? |
| 06:14.27 | [TK]D-Fender | Status <------ |
| 06:14.30 | [TK]D-Fender | dahdi show cstatus |
| 06:14.37 | [TK]D-Fender | dahdi show channels |
| 06:14.40 | F2Knight | didn't knwo about that . will do. |
| 06:17.37 | zopiac | Blagh, trying to set up Asterisk using the online documentation but the commands seem to be deprecated |
| 06:18.56 | F2Knight | [TK]D-Fender, implimented that fxo_ks.. what do I put for the standard parms you were talking about? |
| 06:19.42 | [TK]D-Fender | F2Knight: Look at any other standard FXS doc on this |
| 06:20.13 | [TK]D-Fender | zopiac: https://wiki.asterisk.org/wiki/display/AST/Home |
| 06:20.52 | F2Knight | http://pastebin.com/7pqHsWuj |
| 06:20.54 | [TK]D-Fender | zopiac: "the" documentation? There are dozens upon dozens of different sites with "documentation" on it. To which were you referring? |
| 06:21.03 | zopiac | [TK]D-Fender: That's where i am, but the Getting Started tells me to use the 'sip show peers' command but that says command not found |
| 06:21.31 | [TK]D-Fender | YEL <--- yellow alarm doesn't look too good. |
| 06:22.22 | [TK]D-Fender | zopiac: that isn't deprecated. If it doesn't work then you either have a very broken sip.conf preventing chan_sip from loading, or the port was taken by another device and it couldn't bid, or you're missing a modules.conf to tell * what modules to load, etc |
| 06:22.24 | F2Knight | just noticed that.. it was ok earlier. |
| 06:22.33 | [TK]D-Fender | F2Knight: Go look into it |
| 06:22.35 | F2Knight | being playing with config changes doing a restart |
| 06:22.51 | zopiac | [TK]D-Fender: hm, all right, I'll look into that |
| 06:22.58 | [TK]D-Fender | zopiac: Are you running any other SIP software on your server? |
| 06:23.16 | zopiac | [TK]D-Fender: Asterisk as the server and Ekiga as the client, nothing else |
| 06:23.26 | [TK]D-Fender | zopiac: that is the problem |
| 06:23.36 | [TK]D-Fender | Make sure to tell ekiga to bind to a different port |
| 06:23.38 | zopiac | [TK]D-Fender: That's why it can't execute a command? |
| 06:24.10 | [TK]D-Fender | zopiac: Ekiga stole port 5060 for itself. * wants to bind to it and Ekiga got it first so chan_sip bombs out and never loads |
| 06:24.22 | [TK]D-Fender | Get Ekiga to use a different port |
| 06:24.28 | zopiac | [TK]D-Fender: will do |
| 06:25.48 | zopiac | [TK]D-Fender: well, I would, if I could find how to modify the port it's trying to use |
| 06:25.59 | [TK]D-Fender | Google-Fu |
| 06:27.31 | zopiac | all right port changed |
| 06:28.12 | zopiac | fixed the modules problem as well, woo |
| 06:30.49 | F2Knight | [TK]D-Fender, I think the guy that was on site earlier when we were trying to figure this out might have left it unpluged. |
| 06:31.32 | [TK]D-Fender | Lack of pluggage is a definite potential issue |
| 06:31.56 | zopiac | still getting "Could Not Register (remote party host is offline) |
| 06:31.58 | zopiac | back to the docs again |
| 06:32.00 | F2Knight | [TK]D-Fender, yes this would be.. But we were having the same issues alday. |
| 06:32.55 | F2Knight | originally this gateway sent the span2 out as a pri_net to a nother asterisk box that was already setup to work with the channel bank, but that system got zapped today |
| 06:33.24 | F2Knight | we were hoping to just put the channel bank on to the span2. as the older system was an aserisk 1.2 box. |
| 06:33.30 | F2Knight | and this is now 1.8 |
| 06:36.56 | F2Knight | [TK]D-Fender, okay just confirmed they unpluged the channelbank connection because it was causing all the phones to do an odd ringing when connected. So can we try and resolve with just looking at configs and try to adjust |
| 06:37.24 | [TK]D-Fender | You mean ... like plug it in.... and like really actually kinda try? |
| 06:37.28 | [TK]D-Fender | ...ish? |
| 06:37.55 | F2Knight | no like can we review the configs and make any changes you might see that need to be made to make it work> |
| 06:39.26 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 06:43.12 | [TK]D-Fender | but on that note..... it's bedtime here. Run with it and hopefully I'll be back on early tomorrow to hear that it's all running smoothe |
| 06:43.17 | [TK]D-Fender | ciao for now |
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| 07:02.31 | zopiac | still having difficultied getting ekiga to connect |
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| 07:34.04 | ChannelZ | takes his pants off |
| 07:35.51 | shamelessn00b | lol |
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| 07:42.03 | schmidts | morgen |
| 07:42.13 | schmidts | good morning i meant ;) |
| 07:42.24 | ChannelZ | I was gonna say, cuz my name is Bob |
| 07:59.33 | ollii | schmidts: moin ;) |
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| 08:37.04 | bulkorok | juten morgen... ^^ hi |
| 08:43.32 | *** join/#asterisk jkroon (~jkroon@dsl-244-35-76.telkomadsl.co.za) |
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| 09:00.17 | *** join/#asterisk ujjain (~ujjain@unaffiliated/ujjain) |
| 09:00.27 | ujjain | Hello! :) Do people have experience with Voipbuster? |
| 09:00.54 | ujjain | The call quality lately seems pretty horrible. Do people here have good experiences? I am looking for a cheap voip provider, but Voipbuster, call quality is horrible. |
| 09:03.22 | *** join/#asterisk as001 (~uros@82.117.198.142) |
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| 09:04.27 | *** join/#asterisk petaflot (~root@85-218-19-88.dclient.lsne.ch) |
| 09:05.25 | as001 | Hello do you know why asterisk sometimes give status 0 (Unknown) to just logged in agent, when it should be 1 (not in use), right ? I get status via QueueStatus manager event. |
| 09:07.14 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 09:07.48 | petaflot | hello! I have an issue with my asterisk server. it was workign fine, and now I have some phones that fail registering. for some reason, nmap shows port 5060 as closed |
| 09:10.01 | *** join/#asterisk jacc0 (~jacc0@D522448D.static.ziggozakelijk.nl) |
| 09:10.02 | kaldemar | petaflot: you're scanning a tcp portm right? asterisk uses UDP by default for SIP. |
| 09:10.09 | jacc0 | Goodmorning all! |
| 09:10.54 | kaldemar | petaflot: enable sip debug and verbosity in the asterisk CLI and look at a registration attempt. the commands are "sip set debug on" and "core set verbosity 10". |
| 09:10.56 | as001 | I use Asterisk 1.8.8.0, my queues are realtime |
| 09:11.47 | rjvvliet | ujjain: not using voipbuster, but an affiliate, uses 2 servers one of them had trouble for a while, so check if there are more servers maybe it's only one of them. |
| 09:12.06 | jacc0 | I'm doing a 'make menuselect' on assterisk 1.8.9.0-rc2 and it is showing that `cdr_mysql` is deprecated; is that true? why? and is there a replacement? |
| 09:12.07 | ujjain | rjvvliet: Hmm, I have no better performance with 12voip unfortunately. |
| 09:12.18 | *** join/#asterisk IsUp (~nocturne@unaffiliated/isup) |
| 09:12.25 | IsUp | hi |
| 09:12.33 | jacc0 | hi IsUp |
| 09:12.36 | rjvvliet | jacc0: i'am not mistaken is replaced for odbc. |
| 09:12.44 | jacc0 | okay, ty |
| 09:17.41 | jacc0 | How would one execute sql querys from dialplan using ODBS; I dont see a app_odbc ? |
| 09:17.52 | jacc0 | s/ODBS/ODBC |
| 09:18.03 | *** join/#asterisk qakhan (~qakhan@203.130.22.202) |
| 09:18.07 | qakhan | hi all |
| 09:18.09 | bulkorok | jacc0: use func_odbc |
| 09:19.27 | qakhan | i want to make a front end for user. where they can create exts make changes in dialplan etc |
| 09:20.27 | qakhan | any help... |
| 09:21.29 | jacc0 | @qakhan; load your extensions from database (realtime) |
| 09:21.47 | jacc0 | and give these users access to this database table only |
| 09:21.55 | jacc0 | is that what you are looking form |
| 09:21.58 | qakhan | how |
| 09:21.59 | jacc0 | *for |
| 09:22.27 | qakhan | no |
| 09:22.28 | jacc0 | http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions |
| 09:22.43 | qakhan | i want to make a front end like trixbox has |
| 09:23.04 | jacc0 | did you have a look at asterisk-GUI ? |
| 09:23.31 | qakhan | i want to make front end which has access on .conf files and make changes in .conf files |
| 09:23.57 | jacc0 | that is what asterisk-GUI does |
| 09:25.23 | kaldemar | qakhan: what do you want people to help you with? |
| 09:25.46 | petaflot | kaldemar: "core set verbosity 10" is not recognized as a command.. |
| 09:25.57 | kaldemar | petaflot: what version of asterisk are you using? |
| 09:26.37 | kaldemar | petaflot: sorry, it was my mistake, the command is "core set verbose 10". verbose, not verbosity. |
| 09:27.06 | rjvvliet | jacc0: you answer to ODBC diaplan query see : core show function ODBC_SQL |
| 09:27.21 | *** part/#asterisk as001 (~uros@82.117.198.142) |
| 09:28.22 | petaflot | kaldemar: 1.8.7.1 |
| 09:28.44 | *** join/#asterisk reatoik (~diak2k@217.10.42.10) |
| 09:31.16 | petaflot | what does "jaK" mean? |
| 09:33.11 | jacc0 | is there a escape functino for func_odbc; to prevent sql injection? |
| 09:33.26 | jacc0 | s\functino\function |
| 09:34.25 | *** join/#asterisk jkroon (~jkroon@dsl-244-35-76.telkomadsl.co.za) |
| 09:35.33 | qakhan | kaldemar i need an idea is there any help. i can create php front end which handle the .conf files |
| 09:47.42 | jacc0 | :S |
| 09:48.13 | jacc0 | I am getting an error when doing `make` of 1.8.9.0-rc2 |
| 09:48.16 | jacc0 | asterisk.c:147:25: error: ../defaults.h: No such file or directory |
| 09:48.41 | wdoekes2 | make -j ? |
| 09:48.57 | jacc0 | make -j9 i did |
| 09:49.08 | wdoekes2 | drop the -j |
| 09:49.11 | wdoekes2 | fixed in trunk |
| 09:49.23 | jacc0 | hmm, why is that? |
| 09:49.31 | jacc0 | okay |
| 09:50.13 | jacc0 | :) IT WORKS WITHOUT THE -J |
| 09:50.16 | kaldemar | qakhan: do you have a question about it? if not, just make your php front end that modifies the files. |
| 09:51.07 | wdoekes2 | jacc0: '${SQL_ESC(${var_with_quotes})}' iirc |
| 09:51.23 | *** join/#asterisk black187 (~black187@93-103-22-42.static.t-2.net) |
| 09:51.47 | jacc0 | does SQL_ESC still exist? I will have a look |
| 09:51.59 | jacc0 | compiling first |
| 09:52.15 | jacc0 | I couldn't find SQL_ESC in 1.8.8 |
| 09:52.27 | qakhan | kaldemar yes i have question how to make front end with php which make changes in .conf files |
| 09:53.28 | black187 | hello, did anybody patched Asterisk 1.8 with T38 faxgateway? I've patched it, compile it (just res_fax and res_spandsp - without app_fax). Now I'm having trouble of setting up T.38 protocol in SIP, I've just made changes in the dialplan (exten => _X.,n,Set(FAXOPT(gateway)=yes)), but is this enough. Wireshark trace show's no T.38 |
| 09:53.40 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:84ac:b918:ac6c:39a3) |
| 09:55.23 | jacc0 | SQL_ESC is there :) |
| 09:57.10 | *** join/#asterisk fulcan (~brads@li345-191.members.linode.com) |
| 09:58.54 | fulcan | inbound huntgroup/busy signal, is this only handled by the telco or is there a trick you can do to trip the call over to another line like a forward and free up the preceeding line for new calls? |
| 09:59.58 | jacc0 | how whould I retrieve multiple lines with ODBC_SQL? |
| 10:00.08 | jacc0 | and multi column? |
| 10:01.21 | rjvvliet | jacc: https://wiki.asterisk.org/wiki/display/AST/Function_ODBC_FETCH |
| 10:02.52 | jacc0 | ty :) |
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| 10:07.53 | *** join/#asterisk prasanthprabalan (~root@218.248.24.19) |
| 10:08.27 | prabalan | Hi all |
| 10:08.29 | bulkorok | black187: how about using t38modem != |
| 10:08.33 | bulkorok | ?! |
| 10:08.48 | fulcan | I.E. line 1 is verizon, line 2 is google voice. If verizon line is busy, send call to google voice. hunt grouping across carriers, is this possible? |
| 10:11.05 | rjvvliet | black187: udptl enabled in you sip.conf ? |
| 10:12.28 | jacc0 | is there no need to open and close the odbc connection ? |
| 10:13.06 | rjvvliet | fulcan: i think you need to create your own fail trough if trunk1 reports congestion. |
| 10:15.16 | rjvvliet | fulcan: take a look at GROUP() and GROUP_COUNT() functions to count the number of calls on a sip peer. |
| 10:15.33 | fulcan | rjvvliet my problem is that conceptually I cannot visualize how the inbount rtp stream could cross networks transparently (without a forward) and that first line being made available for new calls. I keep seeing hits that there is a way around this, but I am just not seeing it. It would be a really neat trick if it could be done. |
| 10:17.15 | *** join/#asterisk ccesario (~ccesario@187.17.166.162) |
| 10:17.36 | rjvvliet | fulcan: with asterisk you will have canreinvite=no enable the SIP and RTP will always go trough asterisk, othwise incall DTMF stops working. |
| 10:18.26 | rjvvliet | fulcan: i also think that its not the RTP you eed to think about but the SIP signalling, thats the real channel. |
| 10:20.27 | petaflot | f***. when I try to connect a windows 7 box to my wifi, it DHCPREQUESTs on 192.168.1.12 - server makes an offer on the real subnet (172.16.32.0) and informs the iwnodws host "unknown subnet". the windows client then keeps his crappy IP and therefore has no internet |
| 10:20.31 | petaflot | any clue? |
| 10:20.34 | fulcan | rjvvliet so what you are say is that with canreinvite=no, In an office of 20 phones I can cut 9 of my expensive verizon hardwire lines out and leave only 1, connect my old 9 line to free googlevoice phone number and I can have all the inbound (up to 10 simultainious calls) to the 1 single verizon hardwire number? |
| 10:21.03 | petaflot | sorry wrong chanel |
| 10:21.23 | fulcan | rjvvliet my brain hits a malfunction at the thought of it. |
| 10:21.59 | *** join/#asterisk s[X] (~s_x_@ppp59-167-157-96.static.internode.on.net) |
| 10:22.19 | rjvvliet | fulcan: hahah, yea same here, lets see if i can still follow... |
| 10:24.40 | *** join/#asterisk master_of_master (~master_of@p57B55AFA.dip.t-dialin.net) |
| 10:24.53 | fulcan | rjvvliet cross network inbound hutgroups. :) In England they call it Centrex I believe. But the Centrex system has access to the their pstn to from the 'outsourced/inhouse' MLH (don't quote me in this). |
| 10:25.09 | rjvvliet | fulcan: canreinvite= is just for SIP, is will always keep Asterisk in the call Path, of you set is to Yes, then phone are allowd to reINVITE the other end while oncall and the call stream will be direct phone2phone soa asterisk does not know ANY status. |
| 10:26.15 | rjvvliet | fulcan: Aaa, so when you Sinle Hardline is Busy the privoder will try a second number? |
| 10:26.32 | rjvvliet | fulcan: witch even can be another provider. |
| 10:27.44 | Azrael808 | Hi guys, we have been using a Siemens Gigaset C460IP (cordless DECT phone) for a while, but it's been a bit problematic - I wonder if anyone has any alternative phones they could suggest? |
| 10:29.52 | Azrael808 | Needs to be cordless/wireless |
| 10:30.00 | fulcan | rjvvliet yes, try a second number. but there is usually a cost, or it would have to be an outside service that has a deal cut with the provider to provide huntgrouping extensions. |
| 10:31.46 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 10:32.54 | rjvvliet | fulcan: ok i get it, unfortunately no experience with this service. |
| 10:33.00 | Goldwing | Azrael808, why problematic? i have a couple of s685 basestations and 450/470 bases connected without problems |
| 10:35.11 | Azrael808 | Well, the first problem we kept experiencing was that the phone kept entering a DnD mode... |
| 10:35.42 | Azrael808 | And because it doesn't seem to support syslog, I found it hard to diagnose why. |
| 10:35.58 | Azrael808 | Now the battery appears to be shot, because it's not holding charge :( |
| 10:36.16 | Azrael808 | Understand the battery is just unlucky |
| 10:36.46 | Azrael808 | But the DnD problem was annoying - we could only seem to resolve it by unplugging the base station and re connecting it. |
| 10:37.08 | Goldwing | Syslog, never tried to use it, so can't help with that, the DnD is strange, as i said, i have 5+ Siemens bases connected and none of them go into DnD |
| 10:38.16 | Goldwing | i did have one problem with 1.8 and lower versions of asterisk, somehow asterisk crashes the network in the base station every now and then, but that was solved with 10.0 |
| 10:39.09 | *** join/#asterisk ollii (~risker@2001:470:1f15:1384:62eb:69ff:fe31:b4) |
| 10:41.28 | Azrael808 | Ok - well, we do have an older version of Asterisk I think |
| 10:41.33 | Azrael808 | Let me just check version |
| 10:42.52 | Azrael808 | ok... yeah, we're well out of date: 1.4.24 |
| 10:43.11 | Azrael808 | I'll have to try an upgrade... eek! |
| 10:43.32 | Goldwing | hehehehe |
| 10:45.30 | Azrael808 | Thanks for the info though, good to know the phone isn't a problem :) |
| 10:45.48 | Goldwing | y/w |
| 10:46.11 | Tamo | 7 days |
| 10:52.15 | *** join/#asterisk gego (~quassel@b238085.customer.hansenet.de) |
| 10:57.35 | *** part/#asterisk ujjain (~ujjain@unaffiliated/ujjain) |
| 10:58.01 | *** join/#asterisk [gnubie] (~gnubie]@cm61.sigma15.maxonline.com.sg) |
| 10:58.09 | [gnubie] | waves |
| 10:59.11 | *** join/#asterisk sekil (~sekil@78.24.104.82) |
| 11:01.53 | [gnubie] | i am running "Asterisk 1.8.8.1-1digium1~squeeze" and i am trying to configure a p2p sip uri dialing (like an e-mail address).. |
| 11:01.55 | *** join/#asterisk Ice_Strike (~Ice_Black@host213-120-117-212.in-addr.btopenworld.com) |
| 11:02.52 | [gnubie] | but, i am getting this message from my asterisk console: [Jan 20 18:55:58] NOTICE[2080]: chan_sip.c:22147 handle_request_invite: Call from '' (11.22.33.44:61026) to extension 'gnubie' rejected because extension not found in context 'default'. |
| 11:03.29 | [gnubie] | the thing is, i do not have a "default" context in my extensions.conf . any clue on how to fix this? thank you. |
| 11:06.57 | rjvvliet | gnubie: doe you have an 'exten => gnubie,' in the dedault section? or maybe included? |
| 11:09.09 | rjvvliet | gnubie: sorry mis read... set the context= in the SIP peer to a context that does exist with that extension. |
| 11:14.20 | *** join/#asterisk phpboy (~shane@blowfish.x86.co.za) |
| 11:14.22 | phpboy | hi |
| 11:14.42 | phpboy | How do I set a var to the response of a script I run using system() ? |
| 11:15.30 | kaldemar | phpboy: you don't. use function SHELL for that. |
| 11:17.13 | *** join/#asterisk sekil (~sekil@78.24.104.82) |
| 11:26.26 | Ice_Strike | Does _6[0-2]XX mean Start with 6 and it can be 0, 1 or 2 and then any number of two digits? |
| 11:30.07 | sekil | yes |
| 11:34.29 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 11:35.19 | *** part/#asterisk Cain (~Geek@unaffiliated/cain) |
| 11:35.40 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 11:39.50 | Ice_Strike | thx |
| 11:47.05 | Ice_Strike | What is MACRO_EXTEN? |
| 11:48.48 | *** join/#asterisk ph8 (~ph8@unaffiliated/ph8) |
| 11:51.25 | kaldemar | Ice_Strike: it has the value of the extension that executed the macro. |
| 11:52.40 | Ice_Strike | ahhh got it. |
| 11:55.19 | Ice_Strike | Let say I have 20 hardware phones, it would be better the username and the extention number to be the same on each phone? |
| 11:56.43 | Ice_Strike | So for example: |
| 11:56.46 | Ice_Strike | exten => _3[0-2]XX,1,Dial(SIP/${EXTEN}) |
| 11:56.53 | Ice_Strike | That should work I think |
| 11:57.28 | kaldemar | http://svn.digium.com/svn/asterisk/tags/10.0.0/README-SERIOUSLY.bestpractices.txt |
| 11:58.23 | kaldemar | sure that would work, but see "proper device naming" in that document. |
| 11:58.58 | Ice_Strike | Thanks! |
| 11:59.11 | kaldemar | and you'd have side effects when someone decides to dial a number that is not mapped to an existing device in sip.conf. |
| 12:04.31 | *** join/#asterisk Neptu (~Neptu@mail.avtech.aero) |
| 12:09.37 | *** join/#asterisk s[X] (~s_x_@ppp59-167-157-96.static.internode.on.net) |
| 12:09.45 | Ice_Strike | kaldemar I have just read it, thanks. It say mac address can be used for device name. But then I would have to defince extention number for each device |
| 12:11.01 | Ice_Strike | define* |
| 12:11.42 | Ice_Strike | ? |
| 12:15.59 | bulkorok | Ice_Strike: It's a bit hard for beginning, but if you make realtime peers you can handle extension-mapping with func_odbc and nice sql-querries... |
| 12:19.32 | kaldemar | Ice_Strike: yes. but if you script it, it's no work at all. |
| 12:38.36 | Ice_Strike | Hmmmm |
| 12:40.57 | petaflot | kaldemar: thanks for the help... things are getting to work again |
| 12:45.21 | *** join/#asterisk singler (~singler@81-7-123-162.static.zebra.lt) |
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| 12:47.59 | jacc0 | here , http://www.voip-info.org/wiki/view/Asterisk+func+func_odbc , under the 'Tips and tricks' header it staets a problem with getting the insert id |
| 12:48.21 | jacc0 | how can I make sure it doesn't happen |
| 12:48.22 | jacc0 | ? |
| 12:52.54 | *** join/#asterisk shamelessn00b (~bwahahaha@host-130-net-107-160-119.mobilinkinfinity.net.pk) |
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| 12:54.42 | Ice_Strike | I can't find the reference what is ${TECHNOLOGY} |
| 12:56.50 | kaldemar | Ice_Strike: where did you find such a variable? |
| 12:56.51 | jacc0 | could be 'sip/', 'dahdi/' or some of the other technologys I guess |
| 12:57.45 | jacc0 | how can I uses odbc transaction in asterisk 1.8 ? |
| 12:58.43 | Ice_Strike | kaldemar the link you have provided |
| 12:58.52 | Ice_Strike | I am trying to find a way to get a mac address |
| 12:59.14 | Ice_Strike | exten => _3[0-2]XX,1,Dial(SIP/?????) |
| 12:59.20 | Ice_Strike | replace ????? to mac address |
| 12:59.30 | kaldemar | Ice_Strike: it is just a variable that is set from a database value. |
| 12:59.45 | Ice_Strike | I already define mac address device in sip.conf |
| 13:00.17 | leifmadsen | Ice_Strike: ${TECHNOLOGY} isn't a channel variable built in. In documentation "Technology" is usually in reference to DAHDI/, SIP/, IAX2/, etc |
| 13:00.21 | kaldemar | that example assumes that astdb is used to store additional information. what are you trying to do? |
| 13:00.57 | leifmadsen | Ice_Strike: you need to provide Asterisk the MAC address from somewhere ( a database for instance) |
| 13:02.02 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 13:03.58 | Ice_Strike | @leifmadsen In the sip.conf - I will be having a couple of [mac-address] names |
| 13:04.13 | jacc0 | should I use 'pre-connect => no' to make asterisk odbc use a new connection for every channel? |
| 13:04.18 | leifmadsen | Ice_Strike: ok, that's step one |
| 13:04.34 | Ice_Strike | @leifmadsen in the extensions.conf - i don't want to define each mac address, - it is time consuming. |
| 13:04.34 | leifmadsen | jacc0: that's basically what that would do, yes |
| 13:04.46 | leifmadsen | Ice_Strike: you need to define it somewhere though, like a database |
| 13:05.02 | leifmadsen | Ice_Strike: you need to map an extension or something to the MAC and then look up the MAC when someone dials an extension |
| 13:05.10 | jacc0 | so it will overcome the problem that is stated here , http://www.voip-info.org/wiki/view/Asterisk+func+func_odbc , under the 'Tips and tricks' header it staets a problem with getting the insert id |
| 13:05.13 | jacc0 | ? |
| 13:06.10 | jacc0 | do I need to connect before every query? or wil asterisk auto-connect when a new channel is created? |
| 13:07.25 | leifmadsen | jacc0: no, use multirow mode I believe, then ODBCFinish() to close the multirow connection |
| 13:07.26 | Ice_Strike | @leifmadsen Thanks, I think I understand now.. |
| 13:08.38 | *** join/#asterisk vipkilla (~t_dot_zil@unaffiliated/t-dot-zilla/x-2830497) |
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| 13:09.52 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 13:21.15 | makmak78 | should one use 1.6, 1.8 or 10? |
| 13:22.22 | schmidts | makmak78 1.8 or 10 |
| 13:22.42 | makmak78 | stable wise? |
| 13:22.51 | makmak78 | which one |
| 13:22.52 | schmidts | 1.8 |
| 13:22.57 | makmak78 | alright |
| 13:23.43 | *** join/#asterisk Tuju (~tuju@176.75.219.213.sta.estpak.ee) |
| 13:23.46 | leifmadsen | well, 1.6.{0,1,2} are no longer supported, 1.8 is a long-term-support releaese, and 10 is a standard (1 year support only) release |
| 13:24.01 | makmak78 | alright |
| 13:24.28 | makmak78 | are there any differences regarding performace on the 1.8 vs 10 |
| 13:24.42 | leifmadsen | likely not |
| 13:24.54 | schmidts | leifmadsen ?! ahm wrong ;) |
| 13:25.03 | leifmadsen | depends where you're using it I guess |
| 13:25.03 | schmidts | 10 is faster than 1.8 |
| 13:25.13 | leifmadsen | schmidts: that's a pretty generalized statement |
| 13:25.34 | schmidts | leifmadsen yes thats true, for example sip processing 10 is much better |
| 13:26.03 | leifmadsen | shakes his fist at polycom configurations |
| 13:26.22 | makmak78 | :-) |
| 13:28.40 | rjvvliet | leifmadsen: mmm, i think i got my answer on the status of your quest! |
| 13:28.52 | leifmadsen | rjvvliet: orly |
| 13:28.55 | jacc0 | will ${SQL_ESC(${ARG1})} only escape singe quotes? or will it also escape other special/dangerous chars like % ? |
| 13:29.20 | leifmadsen | jacc0: might want to check the code to be sure how it is handling the escapes, or just test it and see what it does. |
| 13:29.54 | *** join/#asterisk lcat (~lcat@187.45.253.153) |
| 13:30.08 | makmak78 | how about sql support in dialplan in 1.8, is it there? we use mssql |
| 13:30.28 | jacc0 | that's a good idea, while I'm very into SQL-injection maybe I can make some sugestions how improve escaping |
| 13:30.43 | [TK]D-Fender | jacc0, http://xkcd.com/327/ |
| 13:30.43 | *** join/#asterisk s[X] (~s_x_@ppp59-167-157-96.static.internode.on.net) |
| 13:31.17 | [TK]D-Fender | makmak78, fun_odbc <- |
| 13:31.23 | [TK]D-Fender | func_odbc |
| 13:31.30 | makmak78 | okay |
| 13:32.27 | makmak78 | whats the best way of using Fastagi using c# |
| 13:32.56 | makmak78 | should one make a agiserver or is there any good out there |
| 13:33.14 | leifmadsen | rjvvliet: heh ya, I just tested it with _Corey_'s configs as well with 3.2.3 (version he was using) and same thing, so I must still be missing a configuration option somewhere |
| 13:33.50 | jacc0 | @[TK]D-Fender: yes, my last name is ;Drop tables; |
| 13:34.16 | [TK]D-Fender | jacc0, If you care enough do it in AGI |
| 13:35.49 | rjvvliet | leifmadsen: mm yea, i gave up, it's no fun whitout a polycom at hand... i Also think they did not understand the meaning if 'Directed call pickup' look more to me like a 'General call pickup' |
| 13:36.19 | leifmadsen | rjvvliet: ya -- also I may see something. This may just be a bug, or lack of functionality in the SLAStation() application! |
| 13:36.42 | leifmadsen | I see a lot more information from _Corey_'s SIP trace, but he is using Dial() |
| 13:36.44 | leifmadsen | going to try that too |
| 13:37.07 | rjvvliet | leifmadsen: WOW SLA and saterisk, gave that up after a day trying... |
| 13:37.17 | leifmadsen | rjvvliet: ya, I have that part working actually :) |
| 13:37.23 | [TK]D-Fender | * doesn't support SLA... |
| 13:37.33 | leifmadsen | once you understand how the applications work, it's not too bad |
| 13:38.01 | leifmadsen | rjvvliet: I plan on writing more documentation once I understand all of it in terms of device handling and all that |
| 13:38.04 | rjvvliet | leifmadsen: Yeah, but i merely gave up because of callerid funcionality. |
| 13:38.34 | leifmadsen | rjvvliet: gotcha, ya, I might have to re-architect it all in the dialplan, but I think the scenario I'm setting up might work CallerID wise. |
| 13:38.44 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 13:38.58 | leifmadsen | rjvvliet: I won't have any transfers or anything crazy, just inbound and outbound calls, so CallerID only needs ot be populated once |
| 13:39.02 | rjvvliet | leifmadsen: Wow, would be great, at is to the cookbook ;-) or the Def. Guide 4th |
| 13:40.10 | rjvvliet | leifmadsen: mm, was trying to rebuild an in incoming Key-System function that had with the current system. directed them to queues instead. |
| 13:40.29 | leifmadsen | rjvvliet: Definitive Guide 3rd edition has SLA configuration in it |
| 13:40.35 | leifmadsen | I think I need to fix a couple of things to be honest though |
| 13:40.54 | leifmadsen | rjvvliet: ah ya, that could get tricky -- we're not going to be using it for queues and such. |
| 13:41.16 | rjvvliet | leifmadsen: yeah i know, but you can update with your new knowledge ;-) |
| 13:41.24 | leifmadsen | indeed :) |
| 13:42.03 | leifmadsen | wish _Corey_ was online, I need to ask him a question now. He calls SIP/2042, but I think that's just his hint... |
| 13:42.17 | *** join/#asterisk ccesario (~ccesario@187.17.166.162) |
| 13:44.41 | rjvvliet | leifmadsen: But did you find a way to distinct wich BLF is pressed ? |
| 13:45.11 | leifmadsen | rjvvliet: that's what I'm working on right now |
| 13:45.16 | leifmadsen | once I get that part working, I'll be gold |
| 13:45.40 | rjvvliet | leifmadsen: Yep thats 99% of the function.... |
| 13:45.59 | leifmadsen | yep for sure -- and I'm seeing someone got it working :) |
| 13:46.55 | rjvvliet | leifmadsen: A, that something that keeps you going. |
| 13:47.16 | leifmadsen | ya that's the worst part because now I'm obsessive about making it work |
| 13:47.57 | rjvvliet | leifmadsen: And _thats_ something i can fully understand.. |
| 13:51.14 | leifmadsen | rjvvliet: and I'm an idiot |
| 13:51.16 | leifmadsen | rjvvliet: working now |
| 13:51.25 | leifmadsen | notifycid=yes in sip.conf was missing |
| 13:51.33 | rjvvliet | leifmadsen: WOW... |
| 13:52.02 | rjvvliet | leifmadsen: please wait..... reading manual sip,conf....... |
| 13:52.06 | leifmadsen | rjvvliet: yep, although I don't at all see the *97XXXX invite at all :) |
| 13:52.33 | leifmadsen | it just calls Asterisk with the same line button info -- it might work with legacy mode like that |
| 13:53.01 | *** join/#asterisk tamiel (~tamiel@85-171-171-179.rev.numericable.fr) |
| 13:53.37 | rjvvliet | leifmadsen: yep, the button support two modes, legacy using INVITE and normal using the Code. |
| 13:53.49 | leifmadsen | rjvvliet: I think it's the other way around,b ut ya |
| 13:53.54 | leifmadsen | legacy == the star code |
| 13:54.13 | rjvvliet | leifmadsen: mm, was not clear headed yesterday. |
| 13:54.28 | leifmadsen | ya no worries, thanks for the links and encouragement :) |
| 13:54.45 | rjvvliet | leifmadsen: But still want to share this http://www.excaliburtech.net/archives/147 |
| 13:54.48 | leifmadsen | it makes so much sense now |
| 13:55.03 | rjvvliet | leifmadsen: OEPS.... that mensioning the enablecid= |
| 13:55.09 | rjvvliet | sorry, just found that |
| 13:55.21 | rjvvliet | 3sec before you epifany |
| 13:55.27 | leifmadsen | rjvvliet: ya I read that too, and the idiot part was that I thougth I had that enabled already |
| 13:55.33 | leifmadsen | ya I read that YESTERDAY |
| 13:55.37 | leifmadsen | stupid me |
| 13:55.43 | leifmadsen | oh well, figured out now I guess heh |
| 13:56.58 | leifmadsen | rjvvliet: so ya, thanks again for your help, much obliged |
| 13:56.59 | rjvvliet | leifmadsen: Well i think its google related, sometimes you just keep on searching not knowing you have it already. |
| 13:57.16 | leifmadsen | rjvvliet: ya for sure, it was my bad in not verifying my sip.conf was setup right |
| 13:57.39 | leifmadsen | it makes sense now, I need to notify the device (phone) about information so that it can respond appropriately I guess |
| 14:01.57 | *** join/#asterisk ilj (~ilj@sourcemage/grimoire/apprentice/ilj) |
| 14:03.59 | ilj | I use pgsl as a backend for cdr and in cdr.conf I have for csv backed userfield=yes. however neither in pgsql nor in Master.csv userfields are recorded. I'm not sure whether this is something Aterisk generates on its own or it expects some input from user scripts or whatnot. Any pointers would ... |
| 14:04.05 | ilj | ... be nice. |
| 14:04.15 | *** join/#asterisk serafie (~erin@nat/digium/x-ehjmawtrhngbjwoe) |
| 14:05.26 | *** join/#asterisk Bob_Pierce (~WESTMAN\p@216.36.132.162) |
| 14:07.03 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 14:07.05 | *** join/#asterisk bbourdage (~bbourdage@174.33.93.154) |
| 14:07.25 | schmidts | ilj do you set the userfield in your dialplan? |
| 14:07.54 | schmidts | ilj like exten => _X.,Set(CDR(userfield)=123-something) |
| 14:08.35 | ilj | hm let me check |
| 14:11.41 | ilj | schmidts, nope, apparently I don't. Man, wish I asked earlier lol Anyway, I learned quite a bit of new things in the process which is good anyway :) |
| 14:14.10 | jkroon | hi guys, callgroup and pickupgroup - it looks like it's only possible to set that in the range from 1 to 63? I take it this is a bitmask in the code? what if I need 300 such groups? |
| 14:15.34 | kaldemar | jkroon: i guess you'd need to modify source or implement the groups yourself in dialplan. |
| 14:16.26 | [gnubie] | waves |
| 14:16.33 | jkroon | using the Pickup() dialplan calls :) |
| 14:16.42 | [gnubie] | [Jan 20 22:16:06] ERROR[2459]: codec_dahdi.c:578 find_transcoders: Failed to open /dev/dahdi/transcode: No such file or directory |
| 14:16.59 | [gnubie] | but i already bought the g.729 codec license |
| 14:17.28 | jkroon | kaldemar, that can work! thank you very much. |
| 14:17.31 | Ice_Strike | Is it possible to output the value of ${EXTEN} ? |
| 14:17.35 | Ice_Strike | For debugging, etc. |
| 14:17.36 | WIMPy | Hi jkroon! I came across your blog the other day. |
| 14:17.52 | [TK]D-Fender | Ice_Strike, NoOp(whatvever you want) |
| 14:17.53 | jkroon | [gnubie], that error is for the hardware transcoder - ignore it. if g729 show channels works, you're fine. |
| 14:18.03 | rjvvliet | Ice_Strike: NoOp( ${EXTEN}) |
| 14:18.07 | jkroon | WIMPy, ... not sure if that's a good thing ... ? |
| 14:18.21 | Ice_Strike | Ah I see, thanks. |
| 14:18.32 | WIMPy | jkroon: Normal patch cables should always work and that "flat cross" should never be used, even if it may work. |
| 14:19.03 | jkroon | WIMPy, jip, in theory. have you ever bumped into Telkom equipment? |
| 14:19.14 | [gnubie] | jkroon: ok. but, somehow transcoding also does not work properly |
| 14:19.31 | jkroon | [gnubie], "core show translations" ? |
| 14:19.49 | jkroon | WIMPy, in theory yes, but I had that one case where it just would not work if I did not "flat cross" it. |
| 14:20.06 | jkroon | not an electrical engineer, but yes, should not ever be required. |
| 14:20.08 | [TK]D-Fender | [gnubie], Show us the module is loaded and that * acknowledges your licenses |
| 14:20.11 | WIMPy | jkroon: That flat cross thing is a bit of a legend. Thing is that on ptmp you need to have all cables the same. So if all are reversed that's ok, but if you mix correct and reversed cables, you get i |
| 14:20.39 | WIMPy | jkroon: With only one device connected, it shouldn't matter. |
| 14:20.47 | jkroon | WIMPy, that could perhaps be what screwed me over. I'll definitely update the entry with the info you just gave. |
| 14:20.53 | jkroon | if i remember ... |
| 14:21.25 | WIMPy | jkroon: I will put a section about cabling on my to do list as well. |
| 14:22.10 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 14:22.41 | jkroon | please do. when i wrote that I struggled to find information. over the last while I found some really misleading information, but also some more reasonably accurate info. would be very good if there is an accurate source, something which people without much cabling experience can use. |
| 14:23.32 | Katty | hello my asterisk does not work |
| 14:23.37 | WIMPy | I guess I have to draw up something or go hunt for pictures. |
| 14:24.06 | Katty | how to fix plz?? |
| 14:25.47 | leifmadsen | Katty: I suggest you tune the flux capacitor 1/8th of a degree clockwise |
| 14:25.53 | Katty | :> |
| 14:25.56 | [gnubie] | [TK]D-Fender: kindly check http://www.pastie.org/3219192 |
| 14:26.00 | leifmadsen | Katty: but only the northern hemisphere |
| 14:26.01 | Katty | that was original!!! YAY! |
| 14:26.43 | [TK]D-Fender | [gnubie], Ok, so far everything looks fine. Now show us a problem. |
| 14:27.26 | chuckf | leifmadsen: what if she's in the northern hemisphere and the asterisk is in the southern? |
| 14:28.15 | Katty | southern is not northern |
| 14:28.26 | Katty | and his directions were very specific |
| 14:28.44 | leifmadsen | chuckf: then she's screwed |
| 14:31.49 | [gnubie] | [TK]D-Fender: what i noticed if a caller over pots to the trunk port of my home pbx reaches the Background() of my auto-attendant, the audio is so fast that i can't understand |
| 14:32.14 | [TK]D-Fender | [gnubie], What about normal calls? |
| 14:32.33 | [TK]D-Fender | [gnubie], Or any other audio files? did you test with *'s stock G.729 sounds? |
| 14:33.26 | [gnubie] | [TK]D-Fender: meaning, 2 end points on a call session using sip/g729 and dahdi/ulaw ? |
| 14:33.49 | [TK]D-Fender | [gnubie], G.729 end-to-end. Other files in other formats, etc. |
| 14:34.00 | [TK]D-Fender | [gnubie], or "I don't trust YOUR file" |
| 14:34.02 | [gnubie] | [TK]D-Fender: i will try changing the Background() audio file with a g.729 codec for testing |
| 14:34.13 | jkroon | how can I switch off the *8 from features.conf? or do I just need to set it to a different value? |
| 14:34.33 | *** join/#asterisk rossand (~aross@foundation-yow.eclipse.org) |
| 14:34.33 | [TK]D-Fender | jkroon, Ctrl-K |
| 14:34.42 | [gnubie] | [TK]D-Fender: if both end-points uses g.729, isn't it just a pass-through? |
| 14:34.45 | jkroon | :p you're asuming i'm using nano. |
| 14:34.52 | jkroon | [gnubie], run a timing test. |
| 14:35.17 | [TK]D-Fender | [gnubie], fair point. so trancode a call, and use *'s stock ulaw & G.729 files to test |
| 14:35.19 | Ice_Strike | Why there is nothing displaying on the console? |
| 14:35.20 | Ice_Strike | exten => 121,1,Answer |
| 14:35.20 | Ice_Strike | exten => 121,2,NoOp(${EXTEN}) |
| 14:35.35 | [TK]D-Fender | Ice_Strike, Because you didn't set the console verbose level |
| 14:35.38 | jkroon | [gnubie], just for interest's sake: if the file on-disk is already g.729 then it's pass-through too. |
| 14:35.41 | [gnubie] | and, does dahdi on the trunk port to to pots uses ulaw or alaw codec only? |
| 14:35.43 | [TK]D-Fender | Ice_Strike, "core set verbose 10" |
| 14:35.49 | Ice_Strike | Ah |
| 14:35.50 | Ice_Strike | thanks |
| 14:36.05 | [TK]D-Fender | ic_You should notive fo the fact of not seeing any of the call processing whatsoever |
| 14:36.41 | [TK]D-Fender | jkroon, Yeah.. that's catching up to me... Not on all cylinders just yet... |
| 14:37.31 | [gnubie] | jkroon: yes, that's also my assumption. and since i don't have a g.729 codec Background() audio file, i am assuming that asterisk will transcode it |
| 14:38.00 | jkroon | it will. and i've never seen it break (permitting you have the appropriate codecs installed, and that the original file is actually sane) |
| 14:38.20 | [TK]D-Fender | [gnubie], What is your endpoint? |
| 14:39.48 | [gnubie] | [TK]D-Fender: first scenario is only between a caller over pots to the dahdi (fxo) port and asterisk only |
| 14:40.48 | [TK]D-Fender | [gnubie], No, what is the phone that is talking G.729 |
| 14:40.58 | [gnubie] | so the caller is either calling from another house using an analog phone or maybe using a mobile phone.. but both calls are incoming to the fxo of my home pbx |
| 14:41.53 | [TK]D-Fender | [gnubie], Ok, 2 parts : #1: DAHDI playing back g.729 = too fast. What about to a G.729 phone? |
| 14:42.23 | [TK]D-Fender | [gnubie], #2 : G.729 phone to a ULAW file. Too fast? |
| 14:42.27 | [gnubie] | the g.729 phone via sip has no problem |
| 14:42.35 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 14:42.39 | [TK]D-Fender | [gnubie], and use only stock recordings in those formats |
| 14:42.53 | [TK]D-Fender | [gnubie], Sounds like your file is bad then. |
| 14:43.01 | [TK]D-Fender | [gnubie], because * has to translate jsut the same |
| 14:43.48 | [gnubie] | i see.. let me check my configs and audio files first.. it's been a long that time that i didn't touched this home pbx of mine. :D |
| 14:44.26 | jkroon | home pbx ... 10 years back i would never have thought that you'd have a pbx for home ... now i can't imagine having a phone without one :p |
| 14:44.54 | [gnubie] | :D |
| 14:48.53 | *** join/#asterisk _Corey_ (~chatzilla@mail1.infradapt.com) |
| 14:49.41 | [gnubie] | [TK]D-Fender and jkroon: i'm afraid, i have to leave for now. it's late already in here and my son want me to sleep with him. thanks guys. ;-) |
| 14:49.55 | [gnubie] | waves... Gong Xi Fa Cai |
| 14:51.18 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:51.18 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:51.24 | *** join/#asterisk newbe (~Newb@c-98-226-119-225.hsd1.il.comcast.net) |
| 14:53.14 | *** part/#asterisk asterisk-Tester (~RAMYT@193.227.170.247) |
| 14:53.42 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:53.42 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:56.16 | *** join/#asterisk ACiDV (cded2d81@gateway/web/freenode/ip.205.237.45.129) |
| 14:58.02 | ACiDV | Hi, have a small question ... does it possible to do something like : exten => s,n,Set(CMD=Macro) ... exten => s,n,Set(ARG=blabla) .... exten => s,${CMD}(${ARG}) ? |
| 14:58.39 | *** join/#asterisk mchou (~quassel@unaffiliated/mchou) |
| 15:01.52 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
| 15:03.58 | [TK]D-Fender | ACiDV, No. but you can use it in ExecIf() |
| 15:04.16 | [TK]D-Fender | ACiDV, You were also missing a priority there... but who's counting? |
| 15:05.05 | [TK]D-Fender | ACiDV, Aside from the raw exercise of what you've described, what is your practical application of this direction you're looking at? |
| 15:05.39 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 15:05.39 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 15:05.46 | ACiDV | [TK]D-Fender: I have an AGI script that execute command like Goto, etc... problem is that the AGI script doesn't exit until all command are executed |
| 15:05.58 | *** join/#asterisk moy (~moy@173.239.155.74) |
| 15:06.04 | ACiDV | [TK]D-Fender: so I try to find a way to exit the script and continue the task on the dialplan |
| 15:06.16 | [TK]D-Fender | ACiDV, So you want to line them up and "GTFO of AGI as fast as possible then? |
| 15:06.28 | [TK]D-Fender | Suppose that works.. to a point |
| 15:07.00 | [TK]D-Fender | ACiDV, So yeah, nest it in ExecIf's |
| 15:07.41 | ACiDV | [TK]D-Fender: ok, so ExecIf($CMD=Macro) .... Macro($ARG) .... ExecIf($CMD=Voicemail) ... Voicemail($ARG) ? |
| 15:08.08 | [TK]D-Fender | ACiDV, With proper var references, etc. |
| 15:08.25 | ACiDV | [TK]D-Fender: thanks for hint ! |
| 15:08.28 | [TK]D-Fender | ACiDV, Clearly gets messy fast as you have to pile on the parms, etc |
| 15:08.43 | bbourdage | Does anyone have experience with a Cisco SPA525G and 722 codec ?, we are having a performance issue with this model ? |
| 15:09.11 | [TK]D-Fender | ACiDV, Does the actual call flow (app order) realy have to be that variable? |
| 15:10.11 | *** join/#asterisk tully` (Tully@r74-192-179-36.htvlcmta01.hnvitx.tl.dh.suddenlink.net) |
| 15:10.33 | Ice_Strike | Is this secure enough? |
| 15:10.36 | Ice_Strike | In the Sip file: [Agent-101], [Agent-102], [Agent-103] |
| 15:10.44 | Ice_Strike | and in the extensions.conf I have: exten => _[1-2]XX,1,Dial(SIP/Agent-${EXTEN}) |
| 15:11.19 | leifmadsen | Ice_Strike: it's not great... still pretty much an extension number that can be searched easily enough; make sure you have strong, unique passwords |
| 15:12.32 | Ice_Strike | @leifmadsen Each device will have a strong unique passwords |
| 15:13.13 | Ice_Strike | so device name still need to be stroner? |
| 15:13.22 | Ice_Strike | stronger* |
| 15:13.33 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 15:13.38 | [TK]D-Fender | dangnammit |
| 15:13.44 | ACiDV | [TK]D-Fender ... it work perfectly using : exten => s,n,ExecIf($["${DIAL_NEXT_CMD}xxx" != "xxx"]?${DIAL_NEXT_CMD}(${DIAL_NEXT_ARG})) |
| 15:14.27 | [TK]D-Fender | ACiDV, jsut use 1? so it's always true. Or incremental var names, etc |
| 15:14.48 | [TK]D-Fender | ACiDV, I might suggest something else depending on what it is this is really doing... |
| 15:15.01 | [TK]D-Fender | ACiDV, What is the "blocking' part you're looking to avoid? |
| 15:15.57 | ACiDV | [TK]D-Fender: I want to kill the AGI script when I do a Macro/Dial/etc from the AGI script ... so I want to "externalize" some functions |
| 15:16.36 | [TK]D-Fender | ACiDV, I generally recommend you splitting your AGI into multiple different AGI's and jumping back in when needed. |
| 15:16.45 | [TK]D-Fender | ACiDV, But not needing dynamic dialplan... |
| 15:17.12 | ACiDV | [TK]D-Fender: Ok, thanks, will continue testing :) ttyl |
| 15:17.51 | *** join/#asterisk bananapie (~david@70.49.65.154) |
| 15:18.23 | bananapie | I would like to see the output of 'dahdi show channel' on all my dahdi channels at once. Is there a way to do this without calling asterisk 23 times using seq in bash ? |
| 15:19.28 | [TK]D-Fender | bananapie, Nope, X iterations of whatever method is what you've got |
| 15:19.47 | bananapie | Ok, bummer. THanks :) |
| 15:19.48 | *** join/#asterisk ollii (~risker@2001:470:1f15:1384:62eb:69ff:fe31:b4) |
| 15:20.35 | *** join/#asterisk chendy (~Alex_Chan@216.24.198.48) |
| 15:23.00 | *** join/#asterisk hurdman (~ygcheny@cylon.r0b0t.fr) |
| 15:23.03 | hurdman | hi |
| 15:23.16 | rjvvliet | bananapie: is aan AMI DAHDIShowChannels usable? |
| 15:24.03 | rjvvliet | bananapie: Just tested this gives me all my channels in parsable format. |
| 15:24.10 | hurdman | i'm looking for a solution to know if my channel is ever open into an asterisk agi ( into a while, if there was an hangup or something like a crash ) to exit, any idea ? |
| 15:24.50 | bananapie | Hmm, I am running a script in bash, I didn't think I could use AMI from bash |
| 15:24.59 | WIMPy | rjvvliet: "command" |
| 15:25.40 | bananapie | I think I'll use the AMI interface, it'll probably be faster |
| 15:31.07 | [TK]D-Fender | Doubt it. |
| 15:31.27 | *** join/#asterisk rrittenhouse (~rrittenho@unaffiliated/rrittenhouse) |
| 15:31.47 | bananapie | in show channel status it has PRI Flags, can I get this in DAHDIShowChannels ? |
| 15:32.27 | tully` | Does anyone know of a good asterisk->skype solution? I've found a program called siptosis but it requires the actual skype client. I'm curious if anyone knows of anything similar to what digium had. |
| 15:32.35 | rrittenhouse | Is there a specific channel for asterisk scf or is this the correct channel? |
| 15:35.16 | [TK]D-Fender | rrittenhouse, Not released yet and it's more than foreign enough that it'll have its own channel when the time comes |
| 15:36.53 | _Corey_ | um, no... it's #asterisk-scf actually |
| 15:36.56 | rjvvliet | bananapie: Sorry, had some server troubles. yep AMI wil be faster than X times same command. and with a local shell its easy. |
| 15:37.34 | bananapie | ok. I had a look, I can modify chan_dahdi.c to add more information to the AMI interface. Should I submit such a patch to asterisk ? |
| 15:38.36 | bananapie | I noticed that since I updated to 1.8.8.1 from 1.6 occassionnally glare will crash a channel and it will freeze on 'PRI Flags: resetting', the only way to free the channel is to restart asterisk. Is this a driver, asterisk, libpri, dahdi or provider issue ? |
| 15:38.42 | Ice_Strike | I want to develop a call manager website that will communicate with asterisk server. When Agent enter username and password on the website to login - I want the phone to ring to confirm it has logged in |
| 15:38.45 | Ice_Strike | how can that be done? |
| 15:39.21 | Ice_Strike | Is it important to store which phone the agent are using? |
| 15:40.18 | [TK]D-Fender | Depends if you consider it "import" and what your definition of "logged in" is |
| 15:40.22 | [TK]D-Fender | important* |
| 15:40.28 | WIMPy | bananapie: chan_dahdi and/or libpri. |
| 15:40.52 | [TK]D-Fender | Ice_Strike, And as for "phone ring", that will be an actual call, there is no generic "just ring because" |
| 15:41.07 | [TK]D-Fender | Ice_Strike, So you can Originate a call to it, etc |
| 15:41.24 | bananapie | thanks |
| 15:41.51 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 15:44.49 | rrittenhouse | [TK]D-Fender: Thanks. I am reading the asterisk wiki info on it now. I would like to watch it develop. I currently use a Cisco CUCM cluster and I am waiting on something like SCF to come along. |
| 15:45.30 | rjvvliet | bananapie: don't forget call files dial a number and playback(Agent-loginok) |
| 15:45.32 | _Corey_ | rrittenhouse: In case you missed it, the channel for SCF is #asterisk-scf |
| 15:45.36 | *** join/#asterisk Lantizia (~lantizia@cpc22-stok16-2-0-cust96.1-4.cable.virginmedia.com) |
| 15:45.58 | Lantizia | Other than LumenVox does anyone know of any other speech recognition software that works well with asterisk? |
| 15:46.00 | rrittenhouse | _Corey_: Oh sorry. I didn't scroll up! Thanks :) |
| 15:46.19 | _Corey_ | rrittenhouse: No problem |
| 15:46.57 | *** join/#asterisk devmikey (~irc@unaffiliated/devmikey) |
| 15:48.56 | Ice_Strike | [TK]D-Fender I am trying to think of ideas how the call manager can be done. Agent can use any phones on the desk. On the website they will enter their username and password. If they have sucessfully logged in, phone will ring (loud speaker) to confirm and they need to press answer/accept on the phone.. |
| 15:49.15 | Ice_Strike | Maybe include extention number textbox before loggin in. |
| 15:49.38 | [TK]D-Fender | Ice_Strike, As I said, just Originate a call to the phone |
| 15:49.45 | [TK]D-Fender | Ice_Strike, the rest is up to you and your dialplan |
| 15:50.39 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 15:52.12 | lcat | hi |
| 15:54.13 | lcat | I spended two days with segment fault with static realtime on mysql asterisk-1.8.8.1 |
| 15:55.52 | lcat | and the problem was that column var_val was null |
| 15:57.31 | lcat | it's a bug? or it's expected? |
| 16:00.42 | *** join/#asterisk jkroon (~jkroon@dsl-244-34-217.telkomadsl.co.za) |
| 16:06.06 | *** join/#asterisk navaismo (~navaismo@fixed-203-96-202.iusacell.net) |
| 16:06.19 | *** join/#asterisk jsjc (~Adium@181.Red-83-35-52.dynamicIP.rima-tde.net) |
| 16:07.01 | jsjc | I am having trouble troubleshooting a cisco 7912 phone that sometimes via SIP I can hear and sometimes not I know it must be a codec issue… what is the best way to troubleshoot this type of things? |
| 16:08.28 | [TK]D-Fender | jsjc, "sometime hear" vs not is not a codec issue. If there was a mismatcht he call would drop like a rock. |
| 16:08.59 | [TK]D-Fender | jsjc, And the description is rather loos right now. Show us something we can actually debug and maybe we'll be able to advise you on it. |
| 16:10.25 | jsjc | what sort of info will be nice to debug? Because I do not know where to start looking at due to theissue of sometimes SIP clients can hear and sometimes they do not hear nothing. |
| 16:11.41 | *** join/#asterisk mchou (~quassel@unaffiliated/mchou) |
| 16:12.58 | [TK]D-Fender | jsjc, Dump your SIP channels at the time it happens. Show use the complet call from beginning to end with SIP DEBUG enabled, etc |
| 16:13.23 | [TK]D-Fender | jsjc, Give us actual details about the circumstances of the call. What is on both ends, all networking in between, etc |
| 16:14.02 | jsjc | ok Let me gather everything. |
| 16:15.46 | Katty | anyone have a yummy pot pie recipe? |
| 16:16.39 | [TK]D-Fender | Katty, I've heard of a few good brownie ones though... |
| 16:16.41 | *** part/#asterisk hurdman (~ygcheny@cylon.r0b0t.fr) |
| 16:17.34 | Katty | brownie pot pie? ^_- |
| 16:18.39 | [TK]D-Fender | partly.... I'm sure you'll figure it out ;) |
| 16:20.49 | Katty | ... |
| 16:20.52 | Katty | yes, yes i did. |
| 16:21.32 | *** join/#asterisk mort_gib (~mort_gib@16.Red-83-36-63.staticIP.rima-tde.net) |
| 16:21.39 | mort_gib | Hi all |
| 16:21.56 | Katty | hello dear |
| 16:23.06 | mort_gib | Hey Katty, how are you?? |
| 16:23.26 | *** join/#asterisk bbourdage (~bbourdage@174.33.93.154) |
| 16:23.47 | Katty | livin the dream, how're you mort? |
| 16:24.03 | mort_gib | Anyway, any thoughts on Avaya IP phones Like/Hate |
| 16:24.20 | mort_gib | A friend of mine wants to do an installation with them |
| 16:24.37 | mort_gib | they "fell off the back of a truck" kind of approach |
| 16:24.57 | *** join/#asterisk dimon00 (~chatzilla@2.229.25.130) |
| 16:25.31 | Katty | i've never used avaya phones, unless that's what talkswitch uses. lemme check |
| 16:25.35 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v000-101.mobile.uci.edu) |
| 16:25.39 | Katty | infobot: avaya? |
| 16:25.39 | infobot | avaya is probably some big company that equals Micro$oft in phone systems |
| 16:25.51 | mort_gib | Infobot is right |
| 16:26.01 | dimon00 | asterisk 1.8: my icoming calls are from ISDN BRI channels |
| 16:26.05 | mort_gib | But they DID (almost) come free |
| 16:26.07 | dimon00 | my internal phones are SIP |
| 16:26.17 | dimon00 | all the incoming calls are from anonymouse |
| 16:26.40 | dimon00 | I'd like to see the called number on my display |
| 16:26.44 | dimon00 | is there an easy solution? |
| 16:26.59 | Katty | hmm, no, talkswitch uses aastra. |
| 16:27.06 | Katty | so i've not used them. |
| 16:27.51 | mort_gib | dimon00 depends on your ISDN card, but mostly it comes with the default config |
| 16:27.58 | mort_gib | katty Ok, thanks |
| 16:28.14 | mort_gib | Some 40 phones for free, I'd like to see how well they work |
| 16:28.46 | dimon00 | mort_gib: I can see the called number in my console: CALLERID(num) is set... but it seems the SIP header is not |
| 16:29.04 | WIMPy | mort_gib: What can the card do about that? |
| 16:29.11 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 16:29.12 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 16:29.12 | Katty | mort_gib: you should snag one and put it on your desk for a couple days |
| 16:29.20 | Katty | mort_gib: give it a For Real test drive |
| 16:29.36 | mort_gib | katty Already done that (getting crowded here) |
| 16:30.08 | mort_gib | dimon00 When you see it in the console then the ISDN card part has been done, like waiting for the callerid |
| 16:30.46 | Katty | mort_gib: what do you think of it? |
| 16:32.09 | *** join/#asterisk dimon00 (~chatzilla@2.229.25.130) |
| 16:32.13 | mort_gib | Looks nice |
| 16:32.18 | dimon00 | sorry disconnected |
| 16:32.29 | mort_gib | Lost you there Dimon00 |
| 16:32.35 | dimon00 | my CALLERID(did) is set |
| 16:32.43 | mort_gib | When you see it in the console then the ISDN card part has been done, like waiting for the callerid |
| 16:32.54 | dimon00 | I agree |
| 16:32.59 | mort_gib | But you see "Asterisk" on the phones? |
| 16:33.04 | dimon00 | it is something with the SIP part which is wrong |
| 16:33.19 | mort_gib | -Yes, like I wrote, mostly you get this with the default config |
| 16:33.21 | mort_gib | for the cards |
| 16:33.23 | dimon00 | you mean on the display? No. |
| 16:33.29 | mort_gib | But still sometimes.... |
| 16:33.37 | mort_gib | What do you get on the display?? |
| 16:33.49 | mort_gib | What phones are you using?? |
| 16:34.00 | dimon00 | the account name or the time |
| 16:34.13 | *** join/#asterisk mahlon (mahlon@martini.nu) |
| 16:34.22 | mort_gib | So when the phone rings, you see the account name?? |
| 16:34.26 | mort_gib | What phones?? |
| 16:36.09 | Katty | oh wait |
| 16:36.12 | Katty | i've seen this before |
| 16:36.21 | Katty | incoming call from Asterisk shows up when there's no callerid coming in |
| 16:36.29 | Katty | when there's nothing to display |
| 16:36.40 | Katty | reads up |
| 16:36.46 | mort_gib | He sees the callerid in the cli |
| 16:36.57 | Katty | ah |
| 16:37.00 | Katty | yeah i just saw that |
| 16:37.01 | mort_gib | If no callerid is set I get asterisk |
| 16:37.14 | Katty | would help if i read the full conversation ;) |
| 16:39.02 | mort_gib | :-) Possibly |
| 16:39.41 | mort_gib | Some of the really shitty phones just don't do this easy |
| 16:39.51 | mort_gib | I came across some "noname" ip phones |
| 16:40.01 | mort_gib | Jesus, what a precious waste of time |
| 16:40.19 | mort_gib | i couldn't find ANY kind of branding on them |
| 16:40.39 | mort_gib | nmap came up with too many options, so I gave up |
| 16:40.45 | WIMPy | There might be a reason for that. |
| 16:41.10 | mort_gib | Yeah, if I had developed those phones I would NOT like my name on them |
| 16:41.24 | Katty | why are sales reps so annoying? i emailed a dealership to see if they still had a 350z in stock that was on their website...and now they're email me every bloody day |
| 16:41.27 | mort_gib | All the same been there... T-shirt etc |
| 16:41.45 | mort_gib | Uhm um a Nissan?? |
| 16:42.54 | Katty | yes. |
| 16:42.59 | Katty | it was very shiny in the photo |
| 16:43.10 | Katty | sadly i don't think there is going to be room in it for laundry baskets. |
| 16:43.24 | Katty | or ginormous puppy + passenger |
| 16:44.45 | mort_gib | And would that be in any way important |
| 16:44.53 | Katty | yes, very important. |
| 16:45.00 | Katty | how else will i get my laundries to the laundrymat? |
| 16:45.13 | mort_gib | Get your bf to do it |
| 16:45.19 | Katty | pff, boyfriend. |
| 16:45.24 | Katty | who needs one of those. |
| 16:45.38 | mort_gib | Ups, sorry :-) |
| 16:47.03 | Katty | hehe, s'ok |
| 16:47.07 | Katty | i'm just a /little/ bitter |
| 16:47.14 | Qwell | glomps Katty |
| 16:47.20 | Qwell | bitter THIS |
| 16:47.38 | *** join/#asterisk AliRezaTaleghani (~taleghani@unaffiliated/AliRezaTaleghani) |
| 16:47.56 | AliRezaTaleghani | hi all |
| 16:48.43 | AliRezaTaleghani | I have a question about the number of processors core, which is sufficient for asterisk to handle high loads? |
| 16:48.58 | leifmadsen | define: high loads |
| 16:48.59 | Qwell | It depends. |
| 16:49.20 | AliRezaTaleghani | I had install Asterisk on a DL360 with Doal Xeon X5670 |
| 16:49.21 | WIMPy | And what form of load. |
| 16:49.33 | AliRezaTaleghani | but all the times, just one of it's core is under the load |
| 16:49.54 | AliRezaTaleghani | about the load, we have to plan |
| 16:50.20 | *** join/#asterisk nW44b (~Schnitzel@unaffiliated/benwa) |
| 16:50.32 | AliRezaTaleghani | first our company CallCenter which handle 8 E1 ( as incoming calls) |
| 16:51.31 | AliRezaTaleghani | and we about about 1400 Lines of dialplan ( menus, IVRs, and some Queues) ... |
| 16:51.33 | WIMPy | AliRezaTaleghani: Do you use SWEC? |
| 16:51.55 | AliRezaTaleghani | WIMPy: :-. no, I didn't know it |
| 16:52.18 | WIMPy | SoftWare Echo Cancellation |
| 16:52.30 | WIMPy | Or do you use MeetMe? |
| 16:52.43 | AliRezaTaleghani | how, right now ! no |
| 16:53.12 | AliRezaTaleghani | cos I have 2 Cisco AS5300 to convert the calls from PSTN to SIP for me |
| 16:53.23 | AliRezaTaleghani | and they do echo canceletion |
| 16:53.59 | WIMPy | You should tell us what your Asterisk does, not what other equipment does. |
| 16:54.09 | AliRezaTaleghani | :-/ the main problem is that why just one of the Cores are under the load. |
| 16:54.22 | AliRezaTaleghani | WIMPy: ok, what should i exactly explain? |
| 16:54.42 | AliRezaTaleghani | the way i handles the calls? |
| 16:54.59 | WIMPy | Tell us what your Asterisk does. |
| 16:55.24 | AliRezaTaleghani | I will Answer the calls, have some Read functions to get customer indentifying |
| 16:56.02 | AliRezaTaleghani | ** I will identify the customers with some SOAP calls to check them on our CRM |
| 16:56.31 | WIMPy | That does not sound like it could produce much load. |
| 16:57.01 | AliRezaTaleghani | the i will handle some of the customers via IVRs and try to solve their problem,,, finally, if it's nessesery will put them on Queues, to be answered by agents. |
| 16:57.11 | WIMPy | Maybe if you have DTMF detection. Do you have out-of -band DTMF from your Cisco? |
| 16:57.46 | *** join/#asterisk hfb (~hfb@pool-98-119-109-145.lsanca.dsl-w.verizon.net) |
| 16:58.50 | WIMPy | wonders if an AS5300 can do DTMF detection at all. |
| 16:58.52 | AliRezaTaleghani | WIMPy: :-. I am not sure... |
| 16:59.00 | AliRezaTaleghani | :D |
| 16:59.44 | AliRezaTaleghani | WIMPy: really they just catch the calls, and pass them to me by H.323 |
| 17:00.18 | WIMPy | Above you said SIP. |
| 17:00.49 | AliRezaTaleghani | :-. maybe It's my mistake... at the first we work on SIP on that side |
| 17:00.59 | AliRezaTaleghani | but hade some problem about the Timers |
| 17:01.04 | *** join/#asterisk emora (~emora@213.236.9.114) |
| 17:01.06 | AliRezaTaleghani | so changed it to H.323 |
| 17:01.19 | AliRezaTaleghani | but my agents are working on SIP |
| 17:02.06 | *** join/#asterisk emora (~emora@213.236.9.114) |
| 17:02.14 | WIMPy | But you aren't (accidentally) transcoding, are you? |
| 17:02.35 | AliRezaTaleghani | really no, |
| 17:02.42 | WIMPy | Otherwise DTMF is the only thing so far I can see producong load, and that shouldn't be too much. |
| 17:02.49 | AliRezaTaleghani | we just use alaw |
| 17:03.52 | WIMPy | Unfortunatly I have no idea, how expensive DTMF detection is. |
| 17:04.10 | AliRezaTaleghani | must of the time, as the callers count goes over 120 the server load will rise up to 4~5 in 1 , 5, 15 min.... |
| 17:04.34 | AliRezaTaleghani | and the asterisk service will crash |
| 17:04.48 | [TK]D-Fender | * H.323 sucks. |
| 17:05.03 | [TK]D-Fender | By most people accounts |
| 17:05.06 | WIMPy | Where does the time go? |
| 17:05.13 | WIMPy | [TK]D-Fender: The Asterisk implementation? |
| 17:05.21 | [TK]D-Fender | WIMPy, Yes, all of them |
| 17:05.40 | [TK]D-Fender | WIMPy, instability, load issues, etc. |
| 17:05.45 | WIMPy | Otherwise it is magnitudes better than SIP. |
| 17:06.56 | *** join/#asterisk Carlos_PHX1_ (~Carlos@ip68-2-227-192.ph.ph.cox.net) |
| 17:08.37 | leifmadsen | Qwell: ping |
| 17:08.56 | Qwell | pong |
| 17:09.36 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 17:09.45 | leifmadsen | Qwell: where might I find a .spec file for DAHDI? |
| 17:10.12 | *** join/#asterisk SimonMX (~simon@82-71-48-142.dsl.in-addr.zen.co.uk) |
| 17:10.22 | SimonMX | Hey guys, anyone familiar with RTPTIMEOUT ? |
| 17:10.32 | *** join/#asterisk hariom (~hariom@117.216.213.81) |
| 17:10.59 | SimonMX | We've got an interesting issue of calls where the caller doesn't speak for a few minutes being cut off |
| 17:11.18 | Qwell | leifmadsen: in an SRPM on packages.asterisk.org |
| 17:11.18 | hariom | How to build web based IVR and VOIP applications? |
| 17:11.18 | SimonMX | Does RTPKeepalive actually work? |
| 17:12.09 | leifmadsen | Qwell: thanks |
| 17:13.32 | [TK]D-Fender | hariom, What is a "web based IVR"? |
| 17:14.07 | [TK]D-Fender | hariom, " VOIP applications" is extremely vague. Please specify what you want |
| 17:14.11 | *** join/#asterisk nW44b (~Schnitzel@unaffiliated/benwa) |
| 17:15.09 | hariom | [TK]D-Fender, communication between two, Web browser <--> user on softphone or Browser |
| 17:15.44 | [TK]D-Fender | hariom, then that is just a softphone launched on the viewers browser |
| 17:15.55 | [TK]D-Fender | hariom, http://code.google.com/p/red5phone/ |
| 17:16.13 | [TK]D-Fender | hariom, Something like that. There are plenty of others. Go Google around for them. |
| 17:16.21 | [TK]D-Fender | ~wikis |
| 17:16.22 | infobot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 17:16.32 | [TK]D-Fender | hariom, ^^ some can be found ilsted here. |
| 17:16.43 | [TK]D-Fender | listed* |
| 17:16.51 | hariom | [TK]D-Fender: yea, that involves Red5 media server. Anything you know which is Just * and browser side technologies? |
| 17:17.16 | [TK]D-Fender | hariom, Go look at that WIKI and google around. |
| 17:17.53 | hariom | [TK]D-Fender: I have been googling around for sometime and then came here. I am trying to figure out how phono has done it |
| 17:18.23 | [TK]D-Fender | hariom, Maybe they just wrote their own. |
| 17:19.08 | hariom | Has anybody tried phono with *? Does that require any third party propertiery system to use? |
| 17:22.53 | *** join/#asterisk WebSprocket (~WebSprock@93-97-23-210.zone5.bethere.co.uk) |
| 17:23.42 | WebSprocket | hey guys, wondering if someone could help, restarted my asterisk box now cannot seem to make or recieve calls it just says == Using SIP RTP cos mark 5 |
| 17:23.46 | WebSprocket | then disconnect |
| 17:24.38 | *** join/#asterisk NightMonkey (~NightMonk@pdpc/supporter/professional/nightmonkey) |
| 17:27.29 | [TK]D-Fender | hariom, I'm sure pretty much nobody cares about "Phono" |
| 17:27.56 | [TK]D-Fender | WebSprocket, Enable SIP DEBUG and see what's actually coming in |
| 17:28.57 | WebSprocket | [TK]D-Fender is there any instructions on this bit of a newbie at all this. |
| 17:29.23 | [TK]D-Fender | asterisk -rvvvvvvvvvvvvvvvvv |
| 17:29.27 | [TK]D-Fender | sip set debug on |
| 17:31.12 | WebSprocket | http://pastebin.com/kU1t9gbR |
| 17:31.37 | WebSprocket | http://pastebin.com/kU1t9gbR |
| 17:31.45 | WebSprocket | oopy |
| 17:31.53 | WebSprocket | sorry was going to say Non-codec capabilities (dtmf), that doesnt look normal |
| 17:33.57 | [TK]D-Fender | WebSprocket, No user '03300883864' in SIP users list -- No matching peer for '03300883864' from '87.238.72.153:5060' <--- not matching a sip.conf entry |
| 17:34.13 | [TK]D-Fender | WebSprocket, Looking for 441785826400 in default (domain wh.gateway.ws) ---- SIP/2.0 404 Not Found |
| 17:34.31 | [TK]D-Fender | WebSprocket, Fall into [default] due to sip.conf [general] context entry and there is no dialplan match |
| 17:35.21 | WebSprocket | ty, let me have a look |
| 17:36.21 | WebSprocket | mad default had become default1 |
| 17:36.26 | WebSprocket | iv restarted and now working:) |
| 17:36.47 | WebSprocket | ty for your help:) |
| 17:41.43 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 17:43.56 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 17:57.17 | *** join/#asterisk sekil (~sekil@78.24.104.82) |
| 18:10.03 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 18:10.41 | *** join/#asterisk dimon00 (~chatzilla@host129-48-dynamic.116-80-r.retail.telecomitalia.it) |
| 18:11.06 | dimon00 | back hoping in a better connection... |
| 18:11.31 | dimon00 | I'm trying to show CALLERID(did) on my SIP phone display |
| 18:11.37 | dimon00 | any easy way to do it? |
| 18:11.38 | *** join/#asterisk AliRezaTaleghani (~taleghani@unaffiliated/AliRezaTaleghani) |
| 18:11.44 | dimon00 | I'm using asterisk 1.8 |
| 18:12.29 | dimon00 | inbound calls are through a isdn bri card and are redirected internally on SIP phones (siemens gigaset and x-lite softphones) |
| 18:15.36 | dimon00 | basically I want to show the incoming called number (the number the caller made to call my PBX) on my SIP phone. Any idea? |
| 18:19.15 | [TK]D-Fender | dimon00, "core show function CALLERID" |
| 18:19.25 | [TK]D-Fender | dimon00, Set the callerID |
| 18:19.59 | dimon00 | I tried set(CALLERID(num)=... |
| 18:20.14 | dimon00 | but it is not displaying on my SIP phones |
| 18:20.46 | *** join/#asterisk jsjc (~Adium@181.Red-83-35-52.dynamicIP.rima-tde.net) |
| 18:20.55 | dimon00 | gigaset and softphones show "anonymous" on their display when the phone is ringing |
| 18:21.54 | rjvvliet | dimon00: the incoming DID is the extension that is being processed. so first action in you EXTEN= > is to save the value of ${EXTEN} |
| 18:23.02 | dimon00 | ok. where do I set it? in another variable? and how do I display it? |
| 18:24.30 | rjvvliet | dimon00: Save it at the first priority like 'exten => 00239484,1,Set(__DID=${EXTEN})' |
| 18:25.05 | rjvvliet | dimon00: Now you have the dailed number of incoming number in the var DID avail for all spawed channels. |
| 18:25.31 | [TK]D-Fender | dimon00, Show us your call being processed including the SIP debug where it calls out to your phone |
| 18:25.52 | rjvvliet | dimon00: So in this case it contains 00239484 , as the DID |
| 18:26.03 | [TK]D-Fender | rjvvliet, No need to make assumptions of what he's doing, lets jsut see |
| 18:26.04 | *** join/#asterisk bdfoster (~bdfoster@unaffiliated/bdfoster) |
| 18:26.27 | dimon00 | D-Fender: unfortunately I'm not at the office now so I don't have access to the asterisk installation now :( |
| 18:27.25 | [TK]D-Fender | dimon00, Let us know when you are so we can have something to comment on. |
| 18:28.14 | dimon00 | not before tuesday... which is far away.. I'd looking for hints before going back at the installation |
| 18:29.19 | dimon00 | I tried something like 'exten => 00239484,1,Set(CALLERID(num)=${EXTEN}) and then I looked at the CALLERID(num) with noop |
| 18:29.50 | dimon00 | the variable contains the DID value correctly but it is not display on the SIP phones when ringing |
| 18:30.02 | dimon00 | they are simply showing anonymous |
| 18:30.29 | [TK]D-Fender | dimon00, We don't know what you're really doing, ro what is really happning, or what phone you're using. We have no evidence. CALLERID() is what you set to change the callerID> It works. If something else is happening, including having done it wrong, we can't see. There is evidence here. |
| 18:30.32 | dimon00 | it seems the value is not passed at the SIP header but I cannot find any way (from asterisk) to modify a SIP header |
| 18:30.59 | [TK]D-Fender | dimon00, So when you can show us what's happening on your side and are in a position to make changes and retest we can actually help you |
| 18:31.20 | rjvvliet | dimon00: i think we now need a sip trace ;-) , as a last sugestion do Set(CALLERID(num)=88888) does that work? |
| 18:31.28 | *** join/#asterisk timahvo1 (~rogue@197.178.182.107) |
| 18:31.46 | [TK]D-Fender | rjvvliet, I don't think you've been following. He's not at the office. We won't have any evidence for many days |
| 18:31.47 | dimon00 | rjvvliet: no, I tried it. It doesn't work |
| 18:32.00 | rjvvliet | dimon00: and don't forget Set(CALLERID(press)=passed_not_screened) |
| 18:32.34 | dimon00 | Set(CALLERID(press)=passed_not_screened) this I didn't tried |
| 18:33.05 | dimon00 | I tried even set(CALLERID(all)=John Smith <12345) nothing there as well :( |
| 18:33.09 | rjvvliet | [TK]D-Fender: Yep, a'am following, but he is stil asking some idea's so he know's what to try when hes there. |
| 18:33.16 | dimon00 | the variable are not passed to the SIP phones |
| 18:33.52 | rjvvliet | dimon00: Lets continue when your at the site, i think the SIP trace will tell _much_ more ;-) |
| 18:33.56 | dimon00 | I annot find any documents on this topic.... so I decided to ask in IRC |
| 18:35.21 | dimon00 | ok. Just going at the root of what I want to achieve |
| 18:35.42 | dimon00 | I have 10 tel numbers and 3 companies |
| 18:35.53 | dimon00 | any companies has its numbers |
| 18:36.06 | dimon00 | the reception is the same for the 3 companies |
| 18:36.34 | dimon00 | when somebody call one of the 10 numbers the call goes to the reception |
| 18:36.47 | *** join/#asterisk resist0r (~resist0r@69.31.131.51) |
| 18:37.04 | dimon00 | I want the receptionist to see the called number on her display so she can asnwer "good morning this is <name of the company>" |
| 18:37.09 | dimon00 | is it possible? |
| 18:37.22 | bipolar | has anyone been able to build sangoma's wanpipe driver on dahdi 2.6? I might need to downgrade to 2.5 :( |
| 18:38.19 | rjvvliet | dimon00: Should be possible, doing similar things... let's continue then. gotta go sorry. |
| 18:38.29 | dimon00 | ok, next time then |
| 18:41.40 | [TK]D-Fender | <dimon00> I want the receptionist to see the called number on her display so she can asnwer "good morning this is <name of the company>" |
| 18:41.40 | [TK]D-Fender | <PROTECTED> |
| 18:42.21 | [TK]D-Fender | dimon00, I've been doing it with Asterisk for 8 years now. |
| 18:43.07 | [TK]D-Fender | dimon00, If you have some trouble with this, it is specifically on your end and we need real backup to confirm precisely what is preventing you from getting the outcome you want. |
| 18:43.33 | dimon00 | ok. Then I have to understand why I'm not able to pass the callerID to my SIP phones... |
| 18:44.10 | [TK]D-Fender | dimon00, And we can't help yuo because you cant show us. |
| 18:44.22 | [TK]D-Fender | dimon00, Come back soon with it |
| 18:44.24 | dimon00 | yep, I understand |
| 18:44.36 | dimon00 | I'm going to be back on tuesday, then |
| 18:45.06 | dimon00 | so far thank you |
| 18:51.53 | *** join/#asterisk jsidhu (~jsidhu@c-76-103-82-26.hsd1.ca.comcast.net) |
| 18:53.03 | Katty | herpderp |
| 18:54.24 | [TK]D-Fender | ~borkborkbork |
| 18:54.33 | [TK]D-Fender | ~borqborqborq |
| 18:54.35 | [TK]D-Fender | ~borq |
| 18:54.40 | [TK]D-Fender | ~bork |
| 18:54.51 | [TK]D-Fender | Ok, was sure one of those was around |
| 18:54.55 | [TK]D-Fender | dangnammit |
| 18:57.03 | *** join/#asterisk jsidhu (~jsidhu@c-76-103-82-26.hsd1.ca.comcast.net) |
| 18:57.12 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 19:01.00 | *** join/#asterisk titter (~Justin@c-76-101-15-40.hsd1.fl.comcast.net) |
| 19:01.33 | Katty | seems like i told infobot something about that once |
| 19:03.36 | *** part/#asterisk fulcan (~brads@li345-191.members.linode.com) |
| 19:08.15 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 19:08.19 | *** join/#asterisk nighty- (~nighty@static-68-179-124-161.ptr.terago.net) |
| 19:09.57 | titter | I have a queue with member extensions set directly to an IAX trunk ie. IAX2/gateway/1234. This works fine as it's a ringall. Everyone rings, everyone is happy. However if one of those IAX phones becomes unavilable, it instantly answers the queues and stops the other phones from ringing. Anyone know how to stop this? |
| 19:10.58 | [TK]D-Fender | titter, You said it. It answers. Call = answered. the end |
| 19:15.01 | titter | Why is it answering is more or less my question. SIP members of the queue who are unavailable don't cause the same issue. |
| 19:15.45 | *** join/#asterisk voipeng (~voipeng@70.44.195.22.res-cmts.brd2.ptd.net) |
| 19:16.43 | [TK]D-Fender | titter, You should look/ask on that end |
| 19:17.49 | titter | As I said the calls are just marked as answered by Asterisk on the far end. |
| 19:18.16 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 19:18.18 | titter | SIP clients local don't do this. SIP clients on that server in different queues don't do this. I am guessing this has something with IAX? |
| 19:18.30 | [TK]D-Fender | no, it's what you're doing with the call. |
| 19:19.48 | *** join/#asterisk sekil (~sekil@78.24.104.82) |
| 19:23.38 | titter | Nothing special ... it's quite simple, hence my confusion. Incoming DID from a PRI does a goto to context that executes a ringall queue with 12 or so members. When it IAX's to the other server, it rings those extensions. If one of those is unavaialble it answers the call per the CLI on that end. The extensions on the far end are dialed via SIP. |
| 19:24.09 | *** join/#asterisk AliRezaTaleghani (~taleghani@unaffiliated/AliRezaTaleghani) |
| 19:32.33 | Goldwing | Is it possible for Asterisk to listen to DTMF while it's ringing an extension (make it possible for the person calling in to leave a voicemail by pressing 0) |
| 19:32.59 | Goldwing | ? |
| 19:34.18 | [TK]D-Fender | titter, It is explicitly answering the call. Go look on their end |
| 19:35.04 | [TK]D-Fender | Goldwing, something like that, yes. "core show application dial" |
| 19:35.42 | Goldwing | [TK]D-Fender, Thx, reading.. |
| 19:41.44 | rossand | In asterisk 1.6.x, I want to add two variables. e.g. blah=${foo} + ${bar} What is the right syntax for this? |
| 19:47.22 | *** join/#asterisk _jamesf (~asdf@c-68-60-138-22.hsd1.mi.comcast.net) |
| 19:48.41 | _jamesf | I'm having an issue with a simple SIP setup where extension A can call B, B cannot call A, and the configurations are identical - but using different clients. Both are behind NAT and have NAT enabled as well. |
| 19:48.55 | _jamesf | the windows client (x-lite) can place calls, the android client (sipdroid) cannot |
| 19:49.13 | [TK]D-Fender | rossand, Set(blah=$[${foo} + ${bar}]) |
| 19:49.28 | [TK]D-Fender | rossand, go read THE BOOK for its sections on variables & expressions. |
| 19:49.30 | [TK]D-Fender | ~book |
| 19:49.30 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 19:49.50 | [TK]D-Fender | JamesJRH, show us : |
| 19:49.50 | [TK]D-Fender | ~pb |
| 19:49.51 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 19:49.52 | [TK]D-Fender | ^^^ |
| 19:50.00 | rossand | [TK]D-Fender: thank you kindly. And I especially appreciate the book ref. will do. |
| 19:51.25 | titter | [TK]D-Fender: If the phone is available it shows as - IAX2/gateway-18257 is ringing. If it's unavailable - IAX2/gateway-16858 answered DAHDI/i1/941xxxxxxx-3cc. The far end then shows - WARNING[23730]: app_dial.c:2218 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) |
| 19:51.48 | [TK]D-Fender | titter, Real pastebin of the complete call execution please.... |
| 19:51.54 | titter | K |
| 19:52.09 | [TK]D-Fender | both ends |
| 19:52.12 | titter | Ya |
| 19:55.18 | titter | [TK]D-Fender: http://pastebin.com/miAiN6PD |
| 19:56.41 | [TK]D-Fender | titter, Where does it said IAX2 answered? |
| 19:57.08 | titter | Line 51 |
| 19:57.34 | [TK]D-Fender | Ah, I see it.. |
| 19:58.00 | [TK]D-Fender | - Executing [s@unavailable:1] BackGround("IAX2/fortmyersro-8276", "vm_exten_option") in new stack <----- explicit answer right here |
| 19:58.12 | titter | Gotcha |
| 19:58.15 | [TK]D-Fender | they answered. Playing audio = answer. |
| 19:59.23 | titter | It's an extension to the way calls go to voicemail ... instead of going directly to voicemail it hits that context playing back the option to try another extension or continue to voicemail. So if I can that it should work. |
| 20:00.46 | Katty | hello. i am not dave. |
| 20:02.49 | [TK]D-Fender | eppigy[-1] |
| 20:03.24 | [TK]D-Fender | "So if I can that it should work." <- no |
| 20:03.54 | [TK]D-Fender | Playback= answer, background=answer, Voicemail=answer |
| 20:08.32 | titter | Well I would setup a different way to just dial those extensions |
| 20:09.06 | *** join/#asterisk jzaw (~jzaw@2001:8b0:7:0:216:cbff:febe:f5dd) |
| 20:09.13 | jzaw | lo peeps |
| 20:09.37 | jzaw | are there any mod_client_asterisk users for ejabberd here pls? |
| 20:09.40 | [TK]D-Fender | titter, You would have to. |
| 20:10.11 | titter | [TK]D-Fender: It's just doing this now: http://pastebin.com/Y40BAFVu |
| 20:10.33 | [TK]D-Fender | titter, exten => s,1,Dial(SIP/${ARG2},20,r) <------ r = EVIL |
| 20:10.38 | [TK]D-Fender | forced audio ringing. |
| 20:10.42 | [TK]D-Fender | don't do this |
| 20:11.00 | [TK]D-Fender | exten => s,n,Goto(unavailable,s,1) <--- guess what was here last time? Background... |
| 20:12.05 | titter | Why is r so evil? lol |
| 20:12.11 | *** join/#asterisk rdegges (u4891@gateway/web/irccloud.com/x-xvvbgkjvgsqlrlxs) |
| 20:12.30 | titter | That's what it's doing now, as in I need to fix that lol |
| 20:13.34 | rdegges | Hey guys, I had Asterisk crash on me last night (running Asterisk 1.8.8.1 + DAHDI (latest)) on ubuntu 11.04 64-bit. The errors I got (filled up my log files), were: |
| 20:14.04 | rdegges | "app_meetme.c: Failed to read frame: Bad file descriptor" |
| 20:14.12 | rdegges | I googled it, but I couldn't find any help. |
| 20:14.53 | rdegges | I had approximately 150 callers active on the server, (spread across 50 conference rooms or so). |
| 20:15.03 | *** join/#asterisk _mwoodj_ (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 20:33.00 | *** join/#asterisk jsidhu (~jsidhu@c-76-103-82-26.hsd1.ca.comcast.net) |
| 20:35.28 | *** join/#asterisk arekm (~arekm@pld-linux/arekm) |
| 20:36.02 | arekm | hi, I'm trying to find out what's the proper way to enable addons in asterisk 10.x at build time (non-interactively). Some make ENABLESOMETHING=1 flag? |
| 20:36.19 | arekm | I'm interested in ooh323c addon |
| 20:39.02 | titter | Can you share or import variables from IAX channels? |
| 20:40.25 | Katty | so... |
| 20:40.29 | Katty | i've been talking to this guy |
| 20:40.40 | Katty | and he's doing the typical let's impress the girl move. |
| 20:40.56 | Katty | he can build his own computer |
| 20:41.04 | Katty | ....should i act impressed, or laugh at him? |
| 20:41.13 | Katty | i mean, really? |
| 20:41.13 | leifmadsen | Katty: yes |
| 20:41.16 | Katty | really? |
| 20:41.36 | leifmadsen | Katty: I never try to impress a girl with my computer building skills... that's the wrong approach |
| 20:41.45 | Katty | agreed. |
| 20:41.59 | Katty | i was not impressed. |
| 20:42.00 | rdegges | I tried to impress my wife when we were in high school with computer building. |
| 20:42.07 | rdegges | But it backfired, and I was too nervous to get the damn lid off =p |
| 20:42.11 | rdegges | So I looked like an idiot! |
| 20:42.14 | rdegges | :( |
| 20:42.17 | Katty | LOL |
| 20:42.43 | rdegges | I just kept fumbling around with the little screws and I was unable to get the lid open for like 10 minutes. So I eventually mumbled something and kinda walked away :( |
| 20:43.03 | Katty | aww poor thing |
| 20:43.11 | rdegges | So, in that regard, I say at least act impressed because he probably put fourth a TON of effort and ginuwinely likes you ^^ |
| 20:43.17 | *** join/#asterisk jamesf (~asdf@c-68-60-138-22.hsd1.mi.comcast.net) |
| 20:43.32 | Katty | but....no |
| 20:43.42 | Katty | see he already knows i'm not intrested |
| 20:43.45 | jamesf | http://pastebin.com/VFwe2cUG - if i place a call from 6000 -> 6001 it fails, but 6001 -> 6000 works fine |
| 20:43.52 | Katty | but he's trying to impress me anyway |
| 20:43.59 | Katty | i'm already seeing a lovely boy |
| 20:44.06 | jamesf | simple setup in freepbx |
| 20:44.23 | Katty | rdegges: also, i just pictured the socially akward penguin when you walked away from the case. |
| 20:44.23 | rdegges | jamesf: what does `sip show peers` output? :o |
| 20:44.33 | leifmadsen | jamesf: sounds like an authentication problem or configuration issue -- impossible to know without delving into that verbose output for a few mins |
| 20:44.36 | rdegges | Katty: that's precisely how i felt, summed up by a reddit meme ^^ |
| 20:44.44 | Katty | *hee* |
| 20:45.03 | titter | leifmadsen: Can you share or import variables from over IAX? |
| 20:45.14 | jamesf | it shows (Unspecified) for host for 6001 |
| 20:45.17 | leifmadsen | titter: not sure, I don't use IAX2 |
| 20:45.20 | rdegges | jamesf: that's the prob i think |
| 20:45.23 | leifmadsen | jamesf: host isn't registered then |
| 20:45.30 | leifmadsen | impossible to call it |
| 20:46.05 | jamesf | leifmadsen: hmm.. if i place a call from 6001, while the call is in place (phone ringing next to me!) it still shows unspecified |
| 20:46.09 | jamesf | is that correct? |
| 20:46.18 | Katty | rdegges: just for you, sweet pea. http://memegenerator.net/instance/13442194 |
| 20:46.41 | rdegges | thanks :) |
| 20:46.50 | leifmadsen | shrugs |
| 20:46.52 | Katty | you're wife will appreciate it ;) |
| 20:46.59 | leifmadsen | your :) |
| 20:47.03 | rdegges | any of you guys going to SCALE by chance? |
| 20:47.08 | rdegges | I'll be there with the wife this year ^^ |
| 20:53.55 | *** join/#asterisk bn-7bc (bjarne@pdpc/supporter/active/bn-7bc) |
| 20:56.09 | *** join/#asterisk navaismo (~navaismo@189.249.54.230) |
| 21:00.26 | _Corey_ | rdegges: I'm not brave enough to take the fiance to a Linux event yet... :) |
| 21:00.33 | _Corey_ | kudos |
| 21:00.47 | rdegges | =p |
| 21:00.52 | rdegges | well she's not technical |
| 21:02.12 | _Corey_ | I'm taking mine to AsteriskWorld in Miami though in a couple weeks... She's an artist, so I doubt she'll come near a session :) |
| 21:06.05 | Katty | leifmadsen: yesh. |
| 21:13.34 | titter | Hmm AsteriskWorld in Miami ... short drive for me, maybe I shall go |
| 21:19.10 | titter | [TK]D-Fender: http://pastebin.com/VQYwBH7x -- Created another IAX trunk for just queue's to use. It lands in the first context here to set a variable (unless there is a better way to share vars over IAX), then it skips going to the eventual Background. All is well. Easy fix in my queues. Thanks for your help! |
| 21:19.36 | WIMPy | Share=transfer? |
| 21:20.05 | titter | Yes |
| 21:20.09 | titter | I a sense |
| 21:20.19 | titter | In a sense, I suppose ... stupid rdp. |
| 21:20.22 | WIMPy | 'core show function iaxvar' |
| 21:20.32 | WIMPy | 'core show function IAXVAR' |
| 21:20.40 | ketas | taking girls to events |
| 21:20.45 | ketas | ... |
| 21:21.21 | titter | I really should use that more often |
| 21:21.29 | ketas | ta(l)king |
| 21:22.14 | titter | WIMPy: So if I set a var on server A, I can use server B to retrieve it via iaxvar? |
| 21:22.43 | WIMPy | YOu set it with that function on one side an dan retrieve it on the other. |
| 21:22.56 | WIMPy | yikes |
| 21:23.08 | WIMPy | You set it with that function on one side and can retrieve it on the other. |
| 21:23.58 | titter | Awesome let me give it a go (core show needs to be implanted in my mind...) |
| 21:24.59 | titter | Set(IAXVAR(nvm)=1 |
| 21:25.02 | [TK]D-Fender | Checkout time, BBIAB |
| 21:25.05 | titter | Set(IAXVAR(nvm)=1) blarg |
| 21:25.44 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 21:25.59 | _Corey_ | titter: It's a good event if you've never been: http://www.tmcnet.com/voip/conference/digium-asterisk-world/east-12/ |
| 21:26.24 | titter | _Corey_: Never been. Not to far from me. 2 hour drive. |
| 21:26.37 | WIMPy | When is WMC? |
| 21:27.10 | _Corey_ | titter: There is a lot more going on at ITEXPO and all the other colocated events too, so definitely check it out |
| 21:27.20 | WIMPy | Much later. |
| 21:28.40 | titter | WIMPy: Thank you!!!!! Holy crap that was perfect. |
| 21:29.34 | leifmadsen | _Corey_: btw I solved my issue.... |
| 21:29.50 | leifmadsen | _Corey_: notifycid=yes <-- was missing in sip.conf |
| 21:30.31 | _Corey_ | leifmadsen: Seriously...? I wouldn't have guessed it would have been something different on the Asterisk end of it |
| 21:31.32 | _Corey_ | glad to hear it though... |
| 21:32.01 | leifmadsen | _Corey_: ya same here -- guess it kind of makes sense now. Asterisk needs to tell the device which line is being requested so it can request it back on the answer. |
| 21:32.39 | *** join/#asterisk LemensTS (~matthew@adsl-70-238-150-222.dsl.stlsmo.sbcglobal.net) |
| 21:32.41 | *** join/#asterisk nighty- (~nighty@static-68-179-124-161.ptr.terago.net) |
| 21:32.47 | *** join/#asterisk Linux4Eric (~chatzilla@cpe-71-72-172-65.woh.res.rr.com) |
| 21:32.54 | LemensTS | anyone have a preference on 48port POE switches for Polycom phones? |
| 21:33.53 | Linux4Eric | anyone ever see their FFA license show 1 license but 0 Concurrent CHannels? |
| 21:34.00 | voipeng | any fixes for echo... theres about a second delay the on our end, the wiresharks show a good rtp stream |
| 21:34.39 | _Corey_ | leifmadsen: Well, I guess I need to look at a before/after SIP trace.. I think either way the phone has enough info to know which line is ringing and therefore enough to construct a proper INVITE to pick it up |
| 21:35.06 | _Corey_ | I'm certainly not shocked to know that was the solution though... ;) |
| 21:40.15 | *** join/#asterisk timahvo1 (~rogue@197.178.182.107) |
| 21:43.56 | leifmadsen | _Corey_: ya I took your configuration and downgraded to 3.2.3 and tested, and it still didn't work |
| 21:44.06 | leifmadsen | so I'm pretty sure notifycid=yes is the necessary component |
| 21:44.45 | _Corey_ | oh, i don't doubt it... It would have taken me going through the firmware stuff before I even considered Asterisk though |
| 21:47.16 | _Corey_ | leifmadsen: anyway, congrats on figuring it out... i'm going to make a note here about it on our internal wiki ;) |
| 21:48.40 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:49.00 | leifmadsen | _Corey_: sounds good :) too bad the Polycoms don't do anything with the hold notification |
| 21:49.15 | leifmadsen | would be nice if when you put someone on hold that you could tell the line was on hold |
| 21:50.01 | [TK]D-Fender | You can. |
| 21:50.18 | [TK]D-Fender | Line key status is evident on all models |
| 21:51.15 | leifmadsen | [TK]D-Fender: BLF doesn't show hold status |
| 21:52.07 | [TK]D-Fender | Does on every phone I've ever used |
| 21:52.21 | _Corey_ | leifmadsen: yeah, i've gotten real bridged line appearances working OpenSER and they work pretty sweet like that |
| 21:52.23 | [TK]D-Fender | 30X, 50X, 60x, 32X, 335 |
| 21:52.40 | _Corey_ | [TK]D-Fender: He's talking about on a BLF |
| 21:52.45 | [TK]D-Fender | Ah... |
| 21:52.47 | [TK]D-Fender | oops |
| 21:52.48 | leifmadsen | hence.... |
| 21:52.52 | leifmadsen | <leifmadsen> [TK]D-Fender: BLF doesn't show hold status |
| 21:52.54 | [TK]D-Fender | No, no, I gotcha |
| 21:53.01 | [TK]D-Fender | Misinterpretation there |
| 21:53.37 | [TK]D-Fender | And actually I've never seen BLF report hold status.. if you are doing a BLF on an * hint... |
| 21:53.52 | _Corey_ | leifmadsen: The Polycom attendant thing is pretty darn close though... not sure what would need to happen so that the phone got an appropriate NOTIFY on the hold status |
| 21:53.56 | [TK]D-Fender | I know * doesn't seem to differentiate in-call VS ringing.... |
| 21:54.10 | leifmadsen | [TK]D-Fender: sure it does |
| 21:54.19 | leifmadsen | I get differnet notifications for ringing vs inuse |
| 21:54.35 | [TK]D-Fender | Sorry, meant "polycom" there |
| 21:54.35 | _Corey_ | I can't imagine on the SIP level (from the phone's perspective) that it'd be much different between the OpenSER BLA implementation |
| 21:54.35 | [TK]D-Fender | gah |
| 21:54.35 | [TK]D-Fender | soooo COOOOLD |
| 21:54.42 | leifmadsen | _Corey_: pretty sure Asterisk sends hold status... I just don't think the polycom does anything with it |
| 21:54.48 | _Corey_ | hmmm |
| 21:54.49 | leifmadsen | _Corey_: there is no icon for hold from what i read |
| 21:55.03 | _Corey_ | yeah, it flashes slow red |
| 21:55.17 | _Corey_ | same as a primary line appearance |
| 21:55.43 | _Corey_ | Although, now that I'm thinking about it that's not with the attendant config but with the shared registration |
| 21:55.48 | leifmadsen | _Corey_: hmmm, then I'm going to have to look at that because that should be working as I have notifyhold=yes enabled |
| 21:55.55 | leifmadsen | _Corey_: ah gotcha |
| 21:56.09 | leifmadsen | ya, from what I was reading attendant didn't have hold status for some reason and thought that was odd when I read it |
| 21:56.57 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-xvyrqcgdxorqnglf) |
| 21:57.57 | _Corey_ | leifmadsen: Yeah, I had notifyhold set to yes so I just tried it... no dice |
| 21:58.43 | _Corey_ | strange they didn't implement it the same as the shared line thing |
| 21:59.20 | leifmadsen | _Corey_: ya it made no sense to me either |
| 22:01.41 | titter | I believe it can be done, but requires SIP 3.2.x or later |
| 22:02.15 | *** join/#asterisk _Corey_ (~chatzilla@mail1.infradapt.com) |
| 22:05.24 | LemensTS | Anyone know if Polycom phones use the 2 data pairs as the POE pairs also? |
| 22:05.31 | titter | leifmadsen: Is this what you are looking for? http://support.polycom.com/global/documents/support/technical/products/voice/Static_BLF_TB62475.pdf |
| 22:11.52 | WIMPy | How do you call that thing on a modular plug that locks it in to the socket? |
| 22:11.55 | leifmadsen | titter: I'm using 3.3.0 |
| 22:18.02 | titter | leifmadsen: http://support.polycom.com/global/documents/support/technical/products/voice/SoundPoint_IP_Enhanced_BLF_QT37381.pdf -- Page 6. You are right. No icon for it, but it does change from green to red when a call is on hold. |
| 22:18.34 | leifmadsen | titter: it's already red when it is in use -- page 9 of the document shows only active, busy, ringing. |
| 22:18.49 | leifmadsen | when a call is active, it's already red, not green |
| 22:19.40 | titter | Correct, so it stays red if there is someone on hold as well ... so it supports it, just doesn't display it differently ... but if you presss and hold the line key, it should show all active and held calls |
| 22:20.07 | leifmadsen | titter: heh ok so it doesn't support it then :) |
| 22:20.19 | titter | Basically |
| 22:20.31 | titter | You aren't crazy ... it's a Polycom thing |
| 22:21.01 | leifmadsen | ya, holding the line button doesn't tell me if it's on hold or not -- regardless, there is no visual indicator that a call is on hold vs active |
| 22:21.08 | *** join/#asterisk navaismo (~navaismo@189.249.54.230) |
| 22:21.34 | _Corey_ | leifmadsen: Strictly speaking, I think the attendant implementation is even capable of picking up a call on hold should that information become available, so polycom probably decided not to confuse people |
| 22:21.58 | titter | Does it show different softkeys? I would think it should let you pick up where as if they were on that line, not give those options |
| 22:22.01 | leifmadsen | _Corey_: ya, like if the call is on hold and I press a line key on another phone, it picks it up |
| 22:22.43 | *** join/#asterisk s[X] (~s_x_@eth589.qld.adsl.internode.on.net) |
| 22:22.46 | _Corey_ | leifmadsen: That would be the intuitive behavior... which *is* exactly how their shared line implementation works |
| 22:23.09 | leifmadsen | _Corey_: yep and that's a good thing -- just would be nice if you could see when a call was on hold visually |
| 22:23.56 | _Corey_ | leifmadsen: eh, I can see the upside of not having to keep explaining that even though so-and-so's line is blinking you can't actually pick it up... ;) |
| 22:24.39 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 22:25.31 | leifmadsen | _Corey_: heh true :) |
| 22:31.46 | *** join/#asterisk plundra (1000@v0.article.se) |
| 22:34.54 | Goldwing | Just curious, has anyone ever tried to make a game for asterisk? |
| 22:35.28 | WIMPy | A "win up to $300 or more" game? |
| 22:35.56 | Goldwing | naah,.. just a braindead game for ppl that are waiting.. |
| 22:36.18 | Goldwing | i'm trying to design "Guess the number" |
| 22:36.46 | Goldwing | i know... waste of time.. but i helps me understand how asterisk works alot better |
| 22:37.11 | WIMPy | ably not a bad idea for that purpose. |
| 22:37.20 | WIMPy | Probably not a bad idea for that purpose. |
| 22:37.27 | Goldwing | yea.. |
| 22:38.03 | Goldwing | if only... i could figure out how to do a RND(0-9) into a string.. |
| 22:38.28 | WIMPy | A string? |
| 22:38.33 | Goldwing | sorry. value |
| 22:38.53 | Goldwing | it's late |
| 22:39.04 | WIMPy | Asterisk doesn't have types. You just don't care. |
| 22:39.11 | cmendes0101 | Is there a way to overlap audio during playback? Background waits to finish before moving to playback |
| 22:40.01 | Goldwing | Asterisk does have $(somename) for a string right? |
| 22:40.18 | Goldwing | doesn't it have %(somevalue) for a value? |
| 22:40.28 | WIMPy | If that string contains a number you can do maths on it. |
| 22:40.31 | Goldwing | or something like it |
| 22:40.38 | Goldwing | true |
| 22:40.38 | rjvvliet | Goldwing: does ${RND(0-9)} work? |
| 22:40.42 | WIMPy | And it's ${varname} |
| 22:41.27 | Goldwing | rjvvliet : hmm.. is it that simple?... lemme try |
| 22:41.28 | WIMPy | "0-9" is not a valid parameter for RND. |
| 22:41.37 | WIMPy | And it's called RAND. |
| 22:41.52 | rjvvliet | Goldwing: Yeah wasn't sure about the RND tough... never used it. |
| 22:41.57 | WIMPy | ${RAND(0,9)} |
| 22:42.16 | Goldwing | same here, to be honest, i've never used asterisk untill a couple of days ago.. |
| 22:42.32 | Goldwing | but.. it's fun (nerdy voice included) |
| 22:42.50 | rjvvliet | Goldwing: what wimpy sayd .. its RAND , core show function RAND |
| 22:43.29 | Goldwing | buntu*CLI> core show function rand |
| 22:43.30 | Goldwing | No function by that name registered. |
| 22:43.30 | Goldwing | Command 'core show function rand' failed. |
| 22:43.49 | Goldwing | hmm capital... |
| 22:43.54 | rjvvliet | Goldwing: is uppercase |
| 22:44.00 | Goldwing | yea.. got it |
| 22:44.22 | Goldwing | whowee....Set(junky=${RAND(1,8)}); Sets junky to a random number between |
| 22:44.22 | Goldwing | 1 and 8, inclusive. |
| 22:44.44 | rjvvliet | Goldwing: i'am leaving you with that last remark... its time to go... ttyl |
| 22:45.01 | Goldwing | seeya.. and sleep well |
| 22:45.09 | rjvvliet | Goldwing: thanks... |
| 22:52.42 | WIMPy | So can someone help me out with some english? |
| 22:52.50 | WIMPy | How do you call that thing on a modular plug that locks it in to the socket? |
| 22:53.21 | Goldwing | ??? |
| 22:53.40 | WIMPy | That thing you press to release it. |
| 22:54.17 | carrar | release tab? |
| 22:54.33 | Goldwing | uhmm.. jack-in-the-box... sorry.mate, i really dont know what you mean |
| 22:54.56 | Goldwing | ooh.. wait |
| 22:55.13 | Goldwing | you mean the connector on a telephone? |
| 22:55.23 | WIMPy | carrar: Sounds usable, thanks. |
| 22:55.32 | WIMPy | yes, or on a network cable, ot whatever. |
| 22:55.54 | Goldwing | yea, RJ11 (PSTN) or RJ45(UTP) connector |
| 22:55.58 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 22:56.25 | WIMPy | Well, RJ-* are not the connectors, but, yes, those thing. |
| 22:56.35 | Goldwing | ok? |
| 22:57.00 | Goldwing | RJ = cable |
| 22:57.13 | WIMPy | carrar was on the right track I was only searching for a name for the release / lock thing. |
| 22:57.40 | WIMPy | RJ = Some Bell demarcation point. |
| 22:57.42 | Goldwing | damn.. does that little plastic thing have a name??? |
| 22:57.54 | WIMPy | The connectors are known as modular or western plugs. |
| 22:58.11 | WIMPy | I could call it "nose", I guess. |
| 22:58.37 | Goldwing | or a " tightner" |
| 22:58.40 | WIMPy | So far "release tap" wins. |
| 22:58.51 | carrar | tab |
| 22:59.06 | carrar | not tap |
| 22:59.18 | WIMPy | Err, yes. |
| 22:59.30 | WIMPy | Tapping is done elsewhere. |
| 22:59.37 | Goldwing | i vote for SDB (self destruct button" |
| 22:59.41 | Goldwing | i vote for SDB (self destruct button) |
| 23:00.01 | Goldwing | why?.. it sounds funny |
| 23:00.51 | Goldwing | ma'm, i have te put a new plug on your telephone line, the TDB is broken off.. |
| 23:01.13 | Goldwing | arrgs.. i even misspell my own joke.. SDB |
| 23:01.32 | WIMPy | Use hot glue ;-) |
| 23:02.10 | Goldwing | WIMPy, if you know what my clients look like, you'll forget the word "Hot" .... |
| 23:03.11 | WIMPy | Time to find new clients. |
| 23:03.20 | Goldwing | i agree.... |
| 23:05.04 | Goldwing | does anyone know a US voip provider that hands out free/cheap US phonenumbers? |
| 23:05.30 | Goldwing | i need one for my holliday to FLA in 3 weeks |
| 23:06.40 | eZz | hm, what's a format of disallowed_methods if I have > 1 methods ? METHOD1, METHOD2 or METHOD1 METHOD2 ... METHODN ? |
| 23:07.18 | eZz | because I don't see it's really stripped, even if I see only one method |