| 00:01.00 | *** join/#asterisk moy_ (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
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| 00:44.14 | Jouva | Welp, finally found what I needed from debian to compile the dahdi module so that works now |
| 01:02.07 | *** join/#asterisk lost_soul (~noymfb@cpe-67-249-141-227.twcny.res.rr.com) |
| 01:03.32 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.72) |
| 01:45.35 | *** join/#asterisk zilasb (~zilasb@c-24-99-28-174.hsd1.ga.comcast.net) |
| 01:46.04 | zilasb | needs some help on agi... |
| 01:48.06 | zilasb | in agi I need to call url to add payment into asp. I am using $ch = curl_init('https://url.com/AddCardPayment?username=john&Password=4490&Ticket=178934&Amount=1.00'); |
| 01:48.06 | zilasb | curl_exec($ch); |
| 01:48.06 | zilasb | curl_close($ch); |
| 01:48.20 | zilasb | but agi hangs up on me |
| 01:48.33 | zilasb | any ideas? |
| 01:49.23 | jql | seems like curl is printing something to stdout |
| 01:49.30 | *** join/#asterisk voxter (~voxter@S01060024012ad159.vc.shawcable.net) |
| 01:49.32 | zilasb | it does |
| 01:49.39 | jql | yeah, asterisk dislikes that |
| 01:49.40 | zilasb | is that a reason to hangup? |
| 01:49.42 | jql | yes |
| 01:49.50 | jql | asterisk is listening to what you're printing out |
| 01:49.57 | zilasb | any ideas to work around it? |
| 01:49.58 | voxter | any of you guys use a speed dial in a polycom phone that sends "#" ? Apparently this used to work but now its sending the # escaped as hex |
| 01:50.33 | jql | zilasb: umm... either tell curl to stop printing to stdout, or redirect stdout for curl |
| 01:50.51 | jql | voxter: it's kinda supposed to hex it |
| 01:51.11 | zilasb | jql: thanks. no idea how but will ask our god google about it |
| 01:51.28 | jql | zilasb: good luck |
| 01:52.06 | voxter | jql: its supposed to hex a pound sign? that is a valid telephone digit. |
| 01:52.53 | jql | voxter: but it's sending it as part of a URI, where hex codes are equal to the character it represents |
| 01:53.10 | jql | URI rules apply in SIP |
| 01:53.18 | voxter | jql: i wonder if i disable uri dialing if it will work right |
| 01:54.00 | jql | I dunno. My phones have it disabled, but my dialplan has both # and percent-encoded variants |
| 02:00.50 | voxter | jql: is there a special way to add something with a # in it into a speed dial key and have it actually pass "#" ? |
| 02:00.51 | voxter | \# ? |
| 02:02.22 | jql | lord no |
| 02:02.29 | jql | it's a layering issue |
| 02:02.46 | voxter | I get why urlencode is happening on characters, but do you have a workaround? |
| 02:03.37 | jql | the _phone_ says dial "45#", but the signaling layer says INVITE sip:45%23@somewhere.example.com SIP/2.0 |
| 02:03.44 | zilasb | jql: man i owe you a []). you a life saver!!!! |
| 02:03.53 | jql | zilasb: cool. :) |
| 02:04.18 | zilasb | jql: simple thing like curl_setopt($ch,CURLOPT_RETURNTRANSFER,1); did the job. can't believe I wasted 2-3 hours on it.... |
| 02:04.30 | voxter | jql: ya, no, i get it when you look at the problem in reverse, but, how do i modify the phone's config to pass what i actually want? |
| 02:04.38 | jql | zilasb: I have no idea what that does, but I'm glad it works for you. :) |
| 02:04.43 | voxter | jql: there has to be a way to send a proper "#" as a dtmf digit |
| 02:04.49 | jql | voxter: why would it be possible to turn off? |
| 02:05.20 | jql | voxter: it's like asking if there's an option to turn off the ability to hang up a call |
| 02:05.47 | jql | well, dialing as dtmf is a different issue |
| 02:05.59 | jql | you can probably do *that* |
| 02:06.05 | voxter | jql: except, its a reasonable expectation to program a speed dial key, and when pressed, have it send the digits you asked it to. |
| 02:06.22 | jql | voxter: it does. %23 is exactly equivalent to # in a URI |
| 02:06.25 | jql | it's in the spec. :) |
| 02:06.40 | voxter | okay |
| 02:06.41 | jql | they are interchangeable, rather |
| 02:06.46 | voxter | let me rephrase my question |
| 02:07.18 | voxter | I want to program a button on one of my side cars, that when pressed, parks a call |
| 02:07.36 | voxter | The logical idea would be to make a button send "#900" # being transfer and 900 being park call |
| 02:08.12 | jql | did I forget to mention that a legal extension is: _%23X,1,Goto(#${EXTEN:3},1) ? |
| 02:08.16 | jql | just sayin' |
| 02:08.32 | voxter | jql: # is a dtmf digit that asterisk picks up from features.conf and interprets |
| 02:08.36 | *** join/#asterisk coppice (~chatzilla@245.168.17.210.dyn.pacific.net.hk) |
| 02:08.38 | voxter | im not trying to dial an extension that begins with # |
| 02:08.45 | voxter | # = blind xfer |
| 02:09.16 | jql | so you want an enhanced feature key which dials dtmf |
| 02:09.19 | jql | that's doable |
| 02:09.39 | voxter | thats in the emkeys stuff right? |
| 02:10.55 | jql | efk, I think |
| 02:11.00 | voxter | it looked to me like doing that would put a park key at the bottom of the screen, not assignable on a sidecar |
| 02:11.14 | jql | ahh, $Tdtmf$ |
| 02:11.28 | jql | efks can be assigned to speed-dial |
| 02:11.32 | voxter | oh i see how it works |
| 02:11.35 | jql | rather, macros can |
| 02:11.55 | voxter | I have a efk set to |
| 02:11.56 | voxter | <PROTECTED> |
| 02:12.27 | voxter | I guess then i put on a sidecar "!callpark" as the exten |
| 02:12.36 | voxter | (i labelled it "callpark" in efk) |
| 02:12.38 | jql | yeah, that sounds right |
| 02:12.50 | voxter | let me try this |
| 02:13.17 | voxter | should <efk> be inside of <sip> or no? |
| 02:13.56 | jql | no, I generally trust the attribute names to specify the xml path |
| 02:16.41 | voxter | so, sorry for the paste |
| 02:16.45 | voxter | <PROTECTED> |
| 02:16.45 | voxter | <PROTECTED> |
| 02:16.45 | voxter | <PROTECTED> |
| 02:16.45 | voxter | <PROTECTED> |
| 02:16.45 | voxter | <PROTECTED> |
| 02:16.46 | voxter | <PROTECTED> |
| 02:16.47 | voxter | <PROTECTED> |
| 02:16.54 | voxter | i took the space out between dt and mf |
| 02:17.44 | voxter | then in a speed dial i set tthe contact to "!callpark" |
| 02:17.46 | voxter | we'll see if that works. |
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| 02:17.54 | *** join/#asterisk cesar_CR (~cesar@201.201.41.242) |
| 02:18.22 | jql | well, should be interesting, at least |
| 02:18.44 | voxter | thanks for the help |
| 02:19.06 | voxter | seems odd that they wouldnt allow a series of numbers to be speed dialed just as-is as digits, and force it to be a url encoded sip string.. |
| 02:19.27 | voxter | should be able to define a speed dial or some kind of button that simply sends whatever you set as dtmf digits in a call |
| 02:19.39 | jql | well, the sip string implies that you were actually dialing that destination as a separate call |
| 02:19.43 | jql | and/or transferring to it |
| 02:19.50 | jql | when you actually wanted inline dtmf |
| 02:20.04 | voxter | right |
| 02:20.13 | voxter | i had set this key up intended to be a 'one key park' |
| 02:20.18 | jql | yeah, speed dial by default is dial. never been a problem for me, but I understand the complaint |
| 02:20.20 | voxter | rather than telling people to hit transfer, then 900 |
| 02:20.47 | jql | I have a one-key park as well, but it's a blind-transfer macro |
| 02:20.55 | voxter | using efk? |
| 02:20.55 | jql | so I never expected dtmf behavior |
| 02:20.57 | jql | yes |
| 02:21.01 | *** join/#asterisk russ (foobar@ip70-176-251-1.ph.ph.cox.net) |
| 02:21.27 | voxter | we sent it as dtmf with # (asterisk's built in blind transfer) so that it reads the park extension to the parker as well |
| 02:21.38 | voxter | then on a group of like 10 phones they all have hints on parking slots 1-10 |
| 02:21.45 | voxter | its like a simulated Shared line system |
| 02:22.12 | jql | yeah. I'm considering using the Warning header as a message system back to the phones in order to allow dynamic park extensions |
| 02:22.17 | jql | right now I have manual selection |
| 02:22.26 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 02:22.31 | voxter | id really love to see some more advanced implementation on polycoms |
| 02:22.40 | voxter | i have all but abandoned them in favor of aastras with their xml integration |
| 02:22.53 | voxter | the polycoms at the time, at least, were far too limited to be able to deliver the same experience |
| 02:23.22 | jql | polycom mainly pisses me off with their software release policy |
| 02:23.44 | voxter | in terms of requiring to be polycom authorized to get the latest release? |
| 02:23.51 | jql | their firmware release cycle spits in the face of a heterogeneous network |
| 02:24.02 | jql | no, they give the firmware away now, or near enough |
| 02:24.06 | voxter | oh, good. |
| 02:24.11 | voxter | so they just dont release often enough? |
| 02:24.19 | jql | they just keep end-of-lifing on the one end, and then requiring the newest fucking release on the other |
| 02:24.20 | voxter | It also pisses me off that now i have to go grab two versions |
| 02:24.27 | voxter | ahhh yeah |
| 02:24.28 | voxter | exactly |
| 02:24.32 | jql | new phones require bleeding edge firmware. old phones are dead and gone |
| 02:24.36 | voxter | you need a version of the new firmware, then a legacy firmware for the others |
| 02:24.40 | voxter | and some do one thing, some dont |
| 02:24.56 | jql | I'm actually up to three separate firmwares |
| 02:25.08 | jql | "stable", "old", and "required for new phone X" |
| 02:25.16 | voxter | polycom is by far worlds behind aastra in terms of managing them |
| 02:25.19 | jql | fucking X |
| 02:27.48 | rue_mohr | polycom sucks if you ask me |
| 02:28.02 | jql | what's your preference? |
| 02:28.07 | rue_mohr | so far, aastra |
| 02:28.16 | rue_mohr | but everyone seems to be making voip sets |
| 02:28.35 | rue_mohr | panasonic, .... some I cant pronounce... |
| 02:29.25 | rue_mohr | anyone ever tried * with a panasonic voip set? |
| 02:29.44 | jql | the list is a lot shorter as a carrier. remote provisioning excludes a lot of fly-by-night handsets. :( |
| 02:29.51 | jql | they try. really they do |
| 02:31.10 | voxter | yeah. |
| 02:31.16 | voxter | i do aastra, polycom |
| 02:31.19 | voxter | thats pretty much it |
| 02:31.28 | rue_mohr | I'v only worked with the 2 |
| 02:31.30 | voxter | i'll do cisco or linksys if someone insists and or begs but i dont advertise them |
| 02:31.34 | rue_mohr | I hear grandstream is junk |
| 02:31.46 | voxter | whats the latest polycom firmware on a 601? |
| 02:32.03 | rue_mohr | Our isp's aren't reliable enough to do wan voip, so I'm doing localized systems |
| 02:32.06 | voxter | I just tried to get this one to update but still seems stuck on 3.0.3 |
| 02:32.18 | rue_mohr | just getting into aastras aastralink 160 |
| 02:33.03 | voxter | i passed on that one |
| 02:33.06 | rue_mohr | no, I gave up on the 601's cause polycom *cant* tell me the gain limits, the defaults are so badly messed that even in a quiet office my users cant hear anything |
| 02:33.08 | voxter | I partnered with a local ADSL provider |
| 02:33.30 | rue_mohr | no aastra audio problems |
| 02:33.43 | voxter | and deliver "point to point adsl" utilizing peering arrangements |
| 02:33.50 | rue_mohr | and the users can see th digits their dialing, which is a big complaint on the polycom |
| 02:33.55 | jql | 601? http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html |
| 02:34.08 | jql | the winner is: 3.1.6! |
| 02:35.22 | voxter | was just on my way there :) |
| 02:35.43 | rue_mohr | does polycom do a canned pbx? |
| 02:36.10 | jql | oh lovely, the 430 was end-of-lifed with 3.3.x |
| 02:36.30 | rue_mohr | ALSO the polycoms have a memory leak and need to be rebooted every few months |
| 02:36.38 | rue_mohr | had some STRANGE issues |
| 02:36.51 | rue_mohr | like calls transfered to them having no audio |
| 02:38.10 | jql | that would be among the reasons I dislike living on polycom's bleeding edge. I keep a nicely aged release around for anyone who doesn't feel like they need an IP5000 or something |
| 02:38.11 | rue_mohr | I think the polycoms adjust the mic level with the earpiece volume, but I dont ahve a sip-spy-scope to prove it |
| 02:38.21 | voxter | Hmm... jql, i downloaded the 3.1.6 legacy combined, and my 601 here is downloading a sip.ld file and claiming it is the same as the one it has (3.0.3) - any ideas? |
| 02:39.01 | jql | voxter: hmm. I dunno how the combined works. I've been using split for years, now |
| 02:39.21 | voxter | oh, should i be doing that? |
| 02:39.26 | voxter | im still confused as to whats the deal with them. |
| 02:39.32 | voxter | i thought combined just included an updated bootrom too |
| 02:39.36 | jql | and by years, apparently I mean more than one, and less than 2. :) |
| 02:39.49 | rue_mohr | but I was just looking at the gstreamer software and It think I can make a program to do eavesdropping and give you audio levels in db |
| 02:40.19 | rue_mohr | as currently there isn't a way to watch sip audio levels |
| 02:40.26 | jql | no, combined has all the phone models' firmwares in a single file |
| 02:40.34 | jql | while split branches out into individual models |
| 02:41.40 | voxter | ok ill download split |
| 02:42.22 | *** join/#asterisk lost_soul (~noymfb@cpe-67-249-141-227.twcny.res.rr.com) |
| 02:43.00 | jql | split requires bootrom 4.0, is the main thing |
| 02:44.00 | voxter | pretty sure all my phones are on bootrom 4.0 or higher |
| 02:44.23 | rue_mohr | and you can download the resultant config from an aastra phone to help you write configurations, most things can be set from the webpage unlike polycoms, "there is a webpage, what more do you want?" |
| 02:45.11 | jql | heh, I disable the webpage on all of our phones, so I guess I'm out of touch with that. :) |
| 02:47.04 | voxter | I dont do web-provisioning unless its a pinch or a test |
| 02:47.15 | voxter | its asking for trouble later. i guess if you only deal with 10 phones who cares |
| 02:47.18 | voxter | but i have 1000s deploeyd |
| 02:47.20 | voxter | deployed |
| 02:47.37 | jql | yeah. if a user has touched the phone config, it might as well be destroyed |
| 02:47.47 | jql | wields the iron fist of God |
| 02:48.04 | voxter | haha |
| 02:48.17 | voxter | I like how you think jql, where do you live? do you need a job? :P |
| 02:48.38 | jql | San Diego, and I'm expensive. :) |
| 02:48.48 | voxter | Do you do this for a living already? |
| 02:48.49 | rue_mohr | I'm waiting for the day our main product is not a Panasonic TDA30 |
| 02:48.51 | voxter | I'm in vancouver |
| 02:48.52 | jql | yep |
| 02:49.04 | rue_mohr | I'm in 'vancouver' |
| 02:49.13 | rue_mohr | what country is your vancouver? |
| 02:49.15 | voxter | Vancouver washington? :P |
| 02:49.17 | jql | lol |
| 02:49.24 | voxter | I'm in canada. |
| 02:49.24 | rue_mohr | mines canada |
| 02:49.26 | jql | I've been to both of those. I'm from Seattle |
| 02:49.26 | voxter | "Real" vancouver. |
| 02:49.35 | rue_mohr | I'm out of van, a ferry trip |
| 02:49.40 | voxter | rue_mohr: island? |
| 02:49.52 | rue_mohr | the 'other' ferry, 'Langdale' |
| 02:50.13 | voxter | oh yeah. |
| 02:50.21 | voxter | I own/operate a business voip company here in the lower mainland |
| 02:50.46 | voxter | the client im working on at the moment which is one of my last on polycoms is Warner Brother Studios |
| 02:50.48 | rue_mohr | do you have a few hundred thousand to start a sip provider up here? |
| 02:50.54 | voxter | haha |
| 02:50.59 | rue_mohr | damn |
| 02:51.03 | jql | heh, WB |
| 02:51.11 | voxter | rue_mohr: we just opened pops in seattle, miami, phoenix, LA |
| 02:51.14 | jamko | Just to confirm, ODBC must be used for storage of voicemail messages in a mysql db??? Using * 1.6.2.10 |
| 02:51.17 | *** join/#asterisk lost_soul (~noymfb@cpe-67-249-141-227.twcny.res.rr.com) |
| 02:51.22 | voxter | and just inked a deal doing integrated voip with a large firewall provider. |
| 02:51.26 | jql | my company keeps getting pinged by Canadians wanting service, but we always have to say no |
| 02:51.26 | rue_mohr | I cant even get isdn or fractional T1 for less than $1300/mo |
| 02:51.31 | jql | serious, wtf Canada? |
| 02:51.53 | jql | it's like the ghost of ma bell is rattling her chains up there. it's sad |
| 02:51.55 | rue_mohr | yea |
| 02:52.09 | rue_mohr | the lack of isdn was a surprise to me |
| 02:52.32 | jql | I can offer e911 service to the yukon territories, but I have to pay 2c/min origination in Toronto? |
| 02:52.32 | rue_mohr | one of my customers wanted ... 2 of them..... wanted 2 lines and 5+ numbers |
| 02:52.37 | jql | facepalms |
| 02:53.57 | rue_mohr | how much is T1 thre? |
| 02:54.56 | voxter | fuck isdn, fuck t1.. I mean, if it were cheap |
| 02:55.29 | rue_mohr | our isp is not a reliable carrier |
| 02:55.30 | voxter | I just get DSL and peer with the dsl provider and provide my own PPPoE |
| 02:55.43 | voxter | so i am single-hop back to the customer |
| 02:56.03 | voxter | I can give out my own ips and all , and they give me free "on-net" traffic since im not routing out to the internet at all. |
| 02:56.06 | rue_mohr | the data networks are flimly as gazoo here |
| 02:56.07 | jql | and what does *that* cost? heh |
| 02:56.15 | voxter | $20/circuit! |
| 02:56.19 | voxter | for 3mb/1mb |
| 02:56.24 | voxter | its perfect |
| 02:56.29 | jql | indeed |
| 02:56.31 | rue_mohr | wow, we cant get that up here for less than $500/mo |
| 02:56.46 | voxter | the thing is you have to partner with an ADSL provider and become a reseller of theirs |
| 02:56.53 | rue_mohr | via eastlink, telus.... no |
| 02:57.02 | rue_mohr | there is only telus up here |
| 02:57.08 | rue_mohr | they dont partner |
| 02:57.13 | jql | telus, telus, and sometimes telus |
| 02:57.16 | voxter | my resller is a subset of telus |
| 02:57.20 | voxter | AEBC |
| 02:57.30 | voxter | or smarttnet, or skyway west |
| 02:57.57 | rue_mohr | most of our area dosn't even have dsl coverage |
| 02:58.03 | voxter | ah that could be a problem then ;) |
| 02:58.18 | rue_mohr | I'v thought about building a buyout |
| 02:58.30 | rue_mohr | the cable isp is quite unreliable |
| 02:58.34 | rue_mohr | (eastlink) |
| 02:59.14 | rue_mohr | so for now, its aastralink 160 with pots lines |
| 02:59.26 | rue_mohr | or home-rolled asterisk |
| 02:59.38 | rue_mohr | which is a problem cause we need techs who can service that |
| 02:59.48 | rue_mohr | as opposed to a tech. |
| 03:01.23 | voxter | hrmmm... :/ |
| 03:01.43 | voxter | this 601 boots and then asks for 2345-11605-001.sip.ld |
| 03:01.47 | voxter | which hasnt been updated |
| 03:01.56 | voxter | in the <mac>.cfg file it just specifies "sip.ld" |
| 03:02.03 | voxter | the 3.1.6 split didnt contain an updated 2345-11605-001.sip.ld |
| 03:03.33 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 03:03.50 | jaytee | evening *ers |
| 03:04.32 | jql | err, weird |
| 03:05.22 | voxter | hmm. put the <APPLICATION_SPIP601 APP_FILE_PATH_SPIP601="sip_316.ld" CONFIG_FILES_SPIP601="phone1_316.cfg, sip_316.cfg" /> |
| 03:05.25 | voxter | inside of the <mac>.cfg file |
| 03:05.31 | voxter | it seems happier. fucked up kludge |
| 03:08.05 | drfreeze | Anyone have some docs on setting up haproxy with AGI? |
| 03:16.00 | p3nguin | [Aug 20 22:09:23] WARNING[18325]: app_voicemail.c:3547 make_email_file: Sox failed to re-encode /var/spool/asterisk/voicemail/default/202/INBOX/msg0000.WAV: An error occurred during file processing (have you installed support for all sox file formats?) |
| 03:16.04 | p3nguin | What's the fix here? |
| 03:16.12 | p3nguin | It worked prior to upgrading asterisk. |
| 03:16.49 | carrar | you installed support for all sox file formats? |
| 03:17.00 | *** join/#asterisk DarkNet (~FreeNoden@courriel-quebec.com) |
| 03:17.00 | p3nguin | No clue. |
| 03:17.11 | p3nguin | If I knew how that was done (or not done), I might know how to overcome it. |
| 03:17.52 | p3nguin | It isn't something I have intentionally changed, so I don't know. |
| 03:19.05 | p3nguin | The last time sox was upgraded was 2010-01-27. I've gone through multiple versions of Asterisk since that. |
| 03:19.39 | p3nguin | I guess I'll start rolling back Asterisk versions until it works again. |
| 03:22.09 | jql | odd |
| 03:24.38 | p3nguin | Okay, so it isn't the build of Asterisk that makes any difference. I just rolled back to a version where I know it was working, but the error still shows up. What could it be? Where do I need to look to fix it? |
| 03:24.40 | *** join/#asterisk lost_soul (~noymfb@cpe-67-249-141-227.twcny.res.rr.com) |
| 03:25.14 | jql | I'm guessing there's not an extension <-> format mismatch? |
| 03:25.39 | jql | file /var/spool/asterisk/voicemail/default/202/INBOX/msg0000.WAV # should say something about wav |
| 03:26.26 | p3nguin | Does not exist. |
| 03:26.49 | jql | oh. |
| 03:27.01 | jql | I can't blame sox for that |
| 03:29.59 | carrar | grep format voicemail.conf |
| 03:31.00 | p3nguin | I thought it could have been that I used to have more than one format specified, so I put it back to wav49|gsm, and made sure that every .WAV in the mailboxes have a .gsm to go along with it. |
| 03:31.38 | p3nguin | But I think I'm going to specify only wav49 and delete all the gsm files. |
| 03:34.40 | p3nguin | I don't understand the "support for all sox formats" thing, though. |
| 03:35.01 | p3nguin | It sounds like something is missing, but how would I include whatever is missing? |
| 03:43.27 | p3nguin | It'll record to .gsm just fine. I can use "file convert" on the CLI to convert the gsm to a WAV with no trouble after the .gsm is recorded. What the hell is wrong that it won't record to .WAV anymore? |
| 03:48.05 | *** join/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
| 03:48.23 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 04:00.15 | p3nguin | It doesn't make any sense to me. |
| 04:01.00 | p3nguin | I see plenty of results including this problem when a google it, but there is no solution. |
| 04:01.14 | p3nguin | when I google it, rather. |
| 04:04.32 | v1s | p3nguin: do the directories have the right permissions? |
| 04:06.47 | raden | Can i have a web app track extension status somehow ? |
| 04:12.39 | *** join/#asterisk lost_soul (~noymfb@cpe-67-249-141-227.twcny.res.rr.com) |
| 04:25.06 | MrHanMan | Has anyone gotten a Cisco 7925g wifi phone to work with Asterisk? I followed a howto on voip-info, and got as far as updating the firmware, but it doesn't seem to be registering to Asterisk. I'm not really sure where to go from here. |
| 04:27.10 | MrHanMan | i'm not sure even how to see if Asterisk loaded chan/sccp |
| 04:28.22 | MrHanMan | sccp show devices return no devices, but also no errors, so i'm assuming that means the chan was loaded |
| 04:33.01 | MrHanMan | here is my SEP[MAC].cnf.xml - http://pastebin.com/ek7gVtgb |
| 04:34.48 | MrHanMan | here is my sccp.conf - http://pastebin.com/BSZu22wW |
| 04:35.32 | *** join/#asterisk Ziaeon (~Alchemist@adsl-074-166-171-132.sip.mia.bellsouth.net) |
| 04:37.48 | p3nguin | v1s: As far as I can tell, yes. |
| 04:38.30 | p3nguin | v1s: app_voicemail is able to create the voicemails in gsm format, but spews that crap when trying to create them as WAV. |
| 04:40.00 | p3nguin | mrhanman: You installed chan_sccp-b already? |
| 04:40.08 | MrHanMan | yes |
| 04:40.19 | MrHanMan | compiled v3rc1 |
| 04:40.20 | p3nguin | Did you run module load chan_sccp? |
| 04:40.37 | MrHanMan | i added it to the modules.conf and rebooted |
| 04:41.00 | p3nguin | That's a bit of overkill, but it should have picked it up nevertheless. |
| 04:41.19 | p3nguin | Check it. sccp show version |
| 04:41.57 | MrHanMan | Skinny Client Control Protocol (SCCP). Release: 3.0 RC1 - (built by 'root' on '2010-08-21 03:18:56 UTC') |
| 04:43.00 | p3nguin | I'm successfully using SCCP channel Release: v2 - 1246. |
| 04:43.41 | p3nguin | I don't see anything on your sccp.conf that's sticking out as being wrong. |
| 04:44.08 | MrHanMan | the problem may be in extension.conf |
| 04:44.12 | MrHanMan | let me paste bin it |
| 04:45.00 | p3nguin | You mentioned that the phone isn't registering to Asterisk. That's not an extensions.conf issue. |
| 04:46.19 | MrHanMan | oh, ok...how would i determine where the breakdown is? |
| 04:47.00 | p3nguin | Increase the debug level in sccp.conf and reload it. |
| 04:47.20 | p3nguin | In v2, I have to unload and load... there is no reload feature. |
| 04:47.45 | p3nguin | Maybe v3 has the reload working in it. module reload chan_sccp |
| 04:48.40 | MrHanMan | what should i increase it to? |
| 04:48.48 | p3nguin | 1 or higher |
| 04:49.05 | raden | is it possible to monitor line status via a browser ? |
| 04:49.07 | p3nguin | I'd probably try 2 or 3. |
| 04:49.35 | MrHanMan | i set it to 3 |
| 04:49.48 | MrHanMan | btw, it doesn't support reload |
| 04:50.11 | p3nguin | Too bad, I suppose. unload it and then load it. |
| 04:50.16 | MrHanMan | done |
| 04:50.28 | p3nguin | It should be producing debug output already. |
| 04:50.29 | MrHanMan | powering on phone |
| 04:52.53 | MrHanMan | the phone says "opening [asterisk ip]", and that's it |
| 04:53.11 | p3nguin | So it isn't loading the necessary files, I would say. |
| 04:53.52 | MrHanMan | it downloads the SEP[MAC].cnf.xml, at least |
| 04:54.36 | p3nguin | There are probably four firmware files, XMLDefault.cnf.xml and SEP<MAC>.cnf.xml that should be in your tftp directory. |
| 04:55.51 | MrHanMan | yes, it downloaded and upgraded the firmware |
| 04:56.23 | p3nguin | I also have imageversion specified in my [devices] section. I don't see it on yours. Not sure if that's important or not. |
| 04:57.23 | p3nguin | imageversion = P00308010200 for my older equipment. |
| 04:58.39 | MrHanMan | what's the iptables command to add 2000 to the allow list in the INPUT chain? |
| 04:58.42 | raden | How do i make it so when people dial out asterisk does not pickup the 1 before they dial ? my call records are all messed up |
| 04:58.51 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 04:59.02 | raden | i have 10 digit call logs with a 1 at the beginning |
| 04:59.51 | raden | another issue comes up what happened when someone dials out of country using like 14 digits ? |
| 05:02.29 | p3nguin | iptables -I INPUT -p udp --dport 2000 -j ACCEPT |
| 05:05.33 | MrHanMan | ok, that's in there |
| 05:06.19 | MrHanMan | would the time/date being way off on the phone hurt? |
| 05:06.28 | p3nguin | I wouldn't think so. |
| 05:06.54 | p3nguin | I expect once you get things working chan_sccp will update that time anyway. |
| 05:08.13 | MrHanMan | still no sccp debugging |
| 05:08.20 | p3nguin | hmm |
| 05:08.35 | *** part/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
| 05:08.44 | p3nguin | Change it to debug level 10. |
| 05:09.16 | p3nguin | If nothing shows up with it on 10, it's borked. |
| 05:09.52 | p3nguin | Mine, by default, was on 1 and it showed plenty of debug info. |
| 05:11.36 | raden | am i invisible tonight ? |
| 05:11.54 | jql | anyone hear something? |
| 05:12.00 | p3nguin | Stupid crap... they closed down Jack in the Box and turned it into a Mexican restaurant. Now I don't have any place to get a raspberry smoothie. |
| 05:12.33 | p3nguin | jql: Not voicemail, that's for damn sure. |
| 05:13.07 | jql | loss of a jack in the box is tragic. what does one do without their curly fries? |
| 05:13.22 | jql | delicious, crispy, curly fries. *mmmmm* |
| 05:15.30 | p3nguin | While recording the voicemail message, the message exists as .WAV in the tmp directory, but once I press 1 to save it, the error pops up and the message file is lost. |
| 05:16.51 | *** join/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
| 05:17.02 | MrHanMan | still nothing in sccp debugging. now what should i do? |
| 05:17.25 | p3nguin | Oh, here's some progress on the voicemail... |
| 05:17.52 | p3nguin | The problem was: app_voicemail.c:3547 make_email_file: Sox failed to re-encode blah blah blah |
| 05:18.32 | p3nguin | So I changed attach = yes to no, and left another voicemail. No problem with it. The file is saved as .WAV and the error did not appear. |
| 05:19.38 | p3nguin | mrhanman: Consider using one of the v2 versions? |
| 05:20.34 | MrHanMan | worth a shot, i guess |
| 05:22.54 | AliRezaTaleghani | hi, all |
| 05:23.24 | AliRezaTaleghani | can some one tell me, how can i modify the incoming calls dialplan.. |
| 05:23.27 | AliRezaTaleghani | i mean: |
| 05:24.10 | AliRezaTaleghani | the call from the cisco, get in with this pattern 5xxx |
| 05:24.22 | AliRezaTaleghani | my internal extention is xxx (ex. 432) |
| 05:25.00 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.92) |
| 05:25.03 | AliRezaTaleghani | how can i route the incoming 5xxx into xxx extention (ex: 5432 => 432) |
| 05:25.04 | russ | :1 |
| 05:25.05 | russ | just like bash |
| 05:25.55 | p3nguin | The call is to 5432, but you want it to be routed to SIP/432? |
| 05:28.22 | AliRezaTaleghani | p3nguin: yes |
| 05:29.20 | p3nguin | If you're using patterns, it would be something like this: exten => _5XXX,1,Dial(SIP/${EXTEN:1},30) |
| 05:30.06 | AliRezaTaleghani | p3nguin: tnx.. will test it today :) |
| 05:30.06 | russ | or a jump ${EXTEN:1)@context |
| 05:30.08 | MrHanMan | ok, v2 is installed and still no joy...it's got to be something stupid |
| 05:30.09 | AliRezaTaleghani | uhuummm |
| 05:30.22 | AliRezaTaleghani | russ i get |
| 05:30.25 | AliRezaTaleghani | tnx |
| 05:31.29 | AliRezaTaleghani | russ: sorry, i wanted to copy your id too, but cleared the history ;) |
| 05:31.41 | AliRezaTaleghani | it was jump into pattern? |
| 05:32.18 | russ | if you have have a context that handles dialing the SIP extensions, you can jump to that |
| 05:32.37 | russ | passing ${EXTEN:1} as the extension to use in that context |
| 05:33.04 | AliRezaTaleghani | well, tnx 2 much |
| 05:33.41 | *** part/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
| 05:35.54 | MrHanMan | now, sccp show devices shows the device, but Reg. State shows None |
| 05:37.43 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 05:38.44 | gamedna | does the playback function depend upon dahdi? |
| 05:39.21 | gamedna | another way to put it is... can i compile asterisk w/o dadhi and still use playback? |
| 05:43.02 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
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| 05:54.44 | MrHanMan | just copied a config from the mailing list with the latest SVN v3 sccp module, and i'm still not seeing any debug info on the console. it's like the phone's not even connecting to asterisk at all. |
| 05:55.58 | *** join/#asterisk rishikesh (~Manipur@117.242.156.54) |
| 05:56.13 | rishikesh | hi |
| 05:56.50 | rishikesh | i need to setup musiconhold as extension so that anybody can call and listen as radio on pbx |
| 05:56.54 | rishikesh | plz help me |
| 05:58.07 | rishikesh | anybody? |
| 05:58.51 | rishikesh | i want to setup extension 301 as musiconhold so that my users can listen to radio by calling 301 |
| 05:59.10 | jql | what's hard about that? |
| 05:59.17 | rishikesh | tell me |
| 05:59.23 | rishikesh | how to do that? |
| 05:59.33 | jql | exten => 301,1,MusicOnHold(radio) |
| 05:59.41 | jql | questions? |
| 05:59.51 | rishikesh | (radio)? |
| 06:00.06 | rishikesh | where to add that config? |
| 06:00.16 | jql | radio is a moh configured in musiconhold.conf |
| 06:00.41 | jql | by you |
| 06:01.04 | rishikesh | ok, but 301 is not available when i dial |
| 06:02.21 | jql | if you setup the call to go to context [example], and add exten => 301,1,NoOp(Example Radio MOH) ; then you should reach it |
| 06:02.22 | rishikesh | it says extension is not available |
| 06:02.27 | jql | does that make sense? |
| 06:03.21 | rishikesh | that all configuration should be add to extension.conf rite? |
| 06:03.36 | jql | yes |
| 06:04.03 | jql | well, you configure sip.conf to actually direct to [example] via context => example |
| 06:06.52 | rishikesh | tell me with some example |
| 06:07.37 | rishikesh | do i need to add configuration on extension.conf and sip.conf |
| 06:08.40 | jql | well, http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf and http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Musiconhold will probably help more than I can |
| 06:09.26 | rishikesh | ok, thank |
| 06:09.32 | rishikesh | i wil check it out |
| 06:14.14 | rishikesh | it is not clear |
| 06:14.45 | rishikesh | first i have to add 301 extension in sip.conf or extension.conf? |
| 06:15.54 | rishikesh | what was that context [example] |
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| 06:47.35 | v1s | is there any oss for speech recognition that works with *? |
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| 07:28.51 | russ | why does itu.int have a html interface to a csv database of space objects? |
| 07:28.57 | russ | ,,USA,,VIKING 2 LAND,,,,SPA-AA,62,,,,1169,01.07.1975, |
| 07:29.11 | russ | 304520169,101520007,MLA,,MEASAT-46E,,46,11.06.2004,CR/C,911,M,1,,2544,17.05.2005, |
| 07:29.14 | russ | etc |
| 07:30.18 | russ | http://www.itu.int/ITU-R/space/snl/bresult/radvancedw_txt.asp |
| 07:30.40 | russ | you can put field specifiers in, like ?sel_satname=AUSSAT A 160E |
| 07:31.13 | russ | why don't they have this awesomeness for their other data that gets locked away in .pdf and .doc? |
| 07:34.35 | ectospasm | 'cuz that would make too much sense? |
| 07:34.50 | russ | their space department must be the cool department |
| 07:37.03 | russ | neato, they have an ftp with maps of all the satellites and the frequencies they use |
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| 08:07.22 | coppice | The ITU coordinates the world's commercial satellites, and stops them treading on each other toes |
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| 11:47.04 | garymc | anyone about? |
| 11:47.27 | garymc | just wanting to know if anyone has experience with Polycom phones |
| 12:04.35 | garymc | MWI message waiting indicator. Im that far upto now. Looking in the phone1.cfg file and cant work out how to turn it on :( |
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| 12:25.46 | MrHanMan | would the time on a Cisco 7925g prevent it from connecting to asterisk? |
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| 12:43.36 | MrHanMan | OK! finally some progress. The phone is now registering. I had enabled iptables to pass port 2000 traffic...but only udp. after a netstat, i saw that it was listening on tcp. as soon as i added a rule for tcp:2000, it registered |
| 12:44.10 | MrHanMan | now, after i place a call from the device, after a few seconds i get a busy signal |
| 12:47.54 | MrHanMan | here is the sccp debug, if anyone can help me decipher it - http://pastebin.com/LNqYR2MK |
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| 12:53.21 | MrHanMan | Do I need to have a custom extension setup? |
| 13:10.31 | MrHanMan | more progress! after fixing the settings in extension_custom.conf, i can now call the phone successfully! unfortunately, i still get the same behavior when calling from the phone. |
| 13:22.42 | MrHanMan | SIP to SCCP works great. I can even hit TransVM on the phone and it works as expected. I just can't figure out why SCCP to SIP isn't working. |
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| 14:01.26 | shapr | Good Morning #asterisk! Wassup? |
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| 15:08.43 | seanjohn | can you do [app-somefunction-${somevariable}] as the context? |
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| 15:15.24 | v1s | seanjohn: you mean like a macro? |
| 15:16.22 | v1s | http://www.voip-info.org/wiki/view/Asterisk+cmd+Macro |
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| 15:42.16 | MrHanMan | anybody had any luck setting up a Cisco 7925g? I can make calls to the device, but none from it. |
| 15:42.42 | MrHanMan | i noticed this in the debug output - "notify asterisk to set state to sccp channelstate INVALIDNUMBER (14) => asterisk: Device is invalid (4) on channel SCCP/705" |
| 15:43.15 | MrHanMan | here's a pastebin of the whole transaction - http://pastebin.com/LNqYR2MK |
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| 15:58.09 | Intel`` | hi guys may i know the command to show the table of conversion time between codecs? |
| 16:00.32 | v1s | core show translation |
| 16:00.48 | v1s | core show codecs |
| 16:01.02 | v1s | Intel``: that what u needed |
| 16:04.52 | nitram | MrHanMan: have you tried chan-sccp-b? |
| 16:05.09 | nitram | MrHanMan: i will have to do the same thing next week, setup a 7925... |
| 16:06.47 | MrHanMan | yes, i have v3preRC2 installed and working. I can receive calls on the 7925, but I can't place calls from it. I was hoping someone else had encountered a similar issue, as I don't think it's the phone itself. |
| 16:07.06 | rue_mohr | anyone know a linux app for voip w/ video? |
| 16:07.42 | MrHanMan | x-lite has a linux version, doesn't it? ekiga should work, too |
| 16:08.21 | nitram | MrHanMan: what device type did you use? |
| 16:08.34 | MrHanMan | 7925 |
| 16:08.38 | nitram | is there already one for 7925? |
| 16:09.28 | MrHanMan | yes, i believe so - it wouldn't work at all otherwise, right? |
| 16:10.23 | rue_mohr | k |
| 16:11.17 | *** join/#asterisk russ (foobar@ip70-176-251-1.ph.ph.cox.net) |
| 16:12.54 | MrHanMan | ah, it seems like it's trying to dial 01701 instead of just 701...why would that be? |
| 16:13.37 | MrHanMan | maybe...that's what it's showing on the display, anyway |
| 16:13.57 | nitram | MrHanMan: do you have some sort of diaplan on the phone itself? |
| 16:14.12 | MrHanMan | not unless it is there by default |
| 16:17.08 | nitram | i do not think so |
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| 16:18.54 | MrHanMan | nitram: that may be my problem |
| 16:19.04 | MrHanMan | nitram: looking into it |
| 16:19.23 | p3nguin | mrhanman: You got the phone online, finally? |
| 16:20.08 | MrHanMan | p3nguin: yep, it was the port...i allowed port 2000/udp, and it was listening on tcp |
| 16:20.25 | p3nguin | tcp? WTH? |
| 16:20.46 | MrHanMan | p3nguin: don't ask me why, but thank God for netstat or i never would have found it |
| 16:20.51 | p3nguin | The phone was TCP or the channel drvier was? |
| 16:20.59 | MrHanMan | i'm not sure |
| 16:21.11 | MrHanMan | the channel driver, sorry |
| 16:22.19 | MrHanMan | seems to be that way by default. i can't find where i specified it, anyway |
| 16:23.04 | p3nguin | I'm going to check mine. |
| 16:25.11 | MrHanMan | p3nguin: i can receive calls on it, but i can't place any |
| 16:26.02 | nitram | MrHanMan: anything specific to the 7925? |
| 16:26.12 | *** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk) |
| 16:26.22 | MrHanMan | nitram: what do you mean? |
| 16:26.37 | p3nguin | Mine appears to be on TCP only, too. So I have to apologize for telling you to add only -p udp in iptables. I assumed SCCP was UDP just like SIP and IAX2. |
| 16:27.00 | nitram | MrHanMan: anything different to 7975/7960? |
| 16:27.01 | MrHanMan | i made the same assumption |
| 16:27.29 | p3nguin | If your phone can get calls but not make calls, that's a context/extension problem. What context did you use for the phone in sccp.conf? |
| 16:27.35 | MrHanMan | nitram: honestly, i can't say. this is the first cisco phone i've setup. |
| 16:27.49 | nitram | i see :) |
| 16:28.31 | MrHanMan | hmm...context is 'internal'. i think it should be 'from-internal' |
| 16:28.49 | p3nguin | Make it match your dialplan like you use on other phones. |
| 16:30.12 | nitram | btw. do you guys know what happened to the "privacy = full" feature? |
| 16:30.26 | *** join/#asterisk GoRK (~gork@c75-111-94-145.amrlcmta01.tx.dh.suddenlink.net) |
| 16:30.31 | nitram | that you could see incoming and outgoing numbers on the other lines? |
| 16:30.59 | nitram | that does not work anymore in trunk |
| 16:31.06 | GoRK | hello; is there a version of the digium g729 codec that works with 1.8 and/or trunk? |
| 16:32.27 | p3nguin | not yet |
| 16:32.38 | GoRK | i hear that there is not which is both disappointing and absurd |
| 16:33.05 | p3nguin | As soon as there is a release of 1.8, the commercial apps will be compiled and available. |
| 16:33.44 | GoRK | well thats too bad guess i wont be testing it then |
| 16:34.23 | pabelanger | GoRK: How is it absurd? |
| 16:34.25 | *** join/#asterisk war9407 (war@liquidswords.org) |
| 16:34.58 | MrHanMan | woohoo! it works! the context was the problem |
| 16:36.08 | p3nguin | I forgot to change my context setting at all when I first tried that channel driver. context sccp did not exist on my dialplan. |
| 16:37.10 | p3nguin | Easy fix, though. |
| 16:37.40 | GoRK | pabelanger: it's absurd because it limits testing and limits the ability to qualify products or configurations for an upcoming release; why have a public beta at all |
| 16:37.57 | MrHanMan | i should have noticed that, but i'm glad it works now. thanks alot for your help. |
| 16:37.59 | Naikrovek | GoRK: because there are other things, besides g729 to test |
| 16:38.27 | Naikrovek | GoRK: be angry all you like, but it's you who's being absurd. test it when its released |
| 16:38.56 | Naikrovek | if you're THAT desperate to deploy 1.8, then switch codecs |
| 16:39.21 | *** join/#asterisk nova911 (~Adium@59.161.12.201) |
| 16:39.31 | pabelanger | GoRK: g729 is not in beta, Asterisk is. Once stable, Digium will then test their product and release when ready. |
| 16:39.41 | MrHanMan | what exactly does 1.8 bring to the table? |
| 16:39.58 | chazzam | Calendars! |
| 16:40.06 | Naikrovek | check the CHANGES document |
| 16:40.08 | chazzam | and SRTP and IPv6 |
| 16:40.09 | Naikrovek | for new features |
| 16:40.23 | chazzam | for some of the big highlights, lots more stuff too though |
| 16:40.27 | GoRK | Naikrovek: what low bandwidth codec that has hardware support in a decent phone would you suggest i use for a user on satellite then? Have been waiting for SRTP for a while and would like to see how it goes |
| 16:40.46 | Naikrovek | GoRK: if you need 1.8 then wait until you can test it. i dont' see what the big deal is |
| 16:40.53 | Naikrovek | it's not out yet. you can't test it. |
| 16:40.55 | Naikrovek | wait until it's out |
| 16:41.15 | pabelanger | GoRK: Does your hardware support speex? |
| 16:41.18 | GoRK | no |
| 16:41.21 | Naikrovek | your logic says you can hop in a time machine and bitch that voip isnt out in 1850 |
| 16:42.26 | pabelanger | You could always contact Digium and see if they would release a version for $$$ |
| 16:44.54 | Corydon76-dig | GoRK: are you on 64-bit or 32-bit? |
| 16:46.11 | Corydon76-dig | GoRK: I'll build one for you, but please understand that it is built with NO OFFICIAL SUPPORT |
| 16:46.16 | GoRK | i never suggested it's not digium's right not to release it; i just suggested that i think its the wrong move becaue i think there are people with a use case for it that would benefit both themselves and digium in being able to test it. I think most of you are misunderstanding me here as complaining about as wanting 'something for nothing' |
| 16:47.01 | p3nguin | It's going to be hard to beat service like this. |
| 16:47.05 | GoRK | Corydon76-: Thanks for the offer and being the voice of reason. I appreciate it. I am 32 bit prescott |
| 16:48.16 | garymc | Ok anyone know how I can get my Polycom IP phones showing MWI again. Here is a copy of my sip.cfg http://pastebin.com/GZiMStBN |
| 16:52.29 | drmessano | GoRK: I was told G729 would begin to be build when RC1 is out |
| 16:52.34 | garymc | they dont tell the user that a voicemail has been left. Just dont know how to get them to work again. Is it an Asterisk or fpbx setting or the phones them selves |
| 16:53.05 | [TK]D-Fender | YES |
| 16:53.31 | garymc | [TK]D-Fender : ??? |
| 16:57.00 | Corydon76-dig | GoRK: please understand that it will take awhile; I need to work out the build issues |
| 16:57.36 | GoRK | Corydon76: no rush or urgency at all; again i appreciate it |
| 16:57.41 | *** join/#asterisk GhOnDiE (~GhOnDiE@92.7.179.66) |
| 16:57.58 | Corydon76-dig | GoRK: and it won't be published anywhere officially |
| 16:58.13 | garymc | anyone know if this sip.cfg is the place I need to be looking to fix my issue? |
| 17:00.42 | pabelanger | garymc: Enable a SIP trace and see what is going on. |
| 17:17.52 | garymc | pabelanger : How do I do a sip trace? |
| 17:18.02 | garymc | sip debug in the cli? |
| 17:18.15 | pabelanger | ~collectdebug |
| 17:18.16 | infobot | hmm... collectdebug is a method of collecting logs allowing others help troubleshoot an issue. Refer to doc/HOWTO_collect_debug_information.txt or http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt |
| 17:29.47 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
| 17:37.37 | *** join/#asterisk newasterx (newasterx@114.199.97.116) |
| 17:37.40 | newasterx | Helloo |
| 17:37.58 | newasterx | why i alway got this message |
| 17:38.00 | newasterx | X-Asterisk-HangupCause: User busy |
| 17:39.53 | pabelanger | newasterx: Because your User is busy? |
| 17:40.13 | pabelanger | we'd need to see a debug of your call |
| 17:41.02 | newasterx | should i copy paste to here... |
| 17:41.07 | p3nguin | ~pb |
| 17:41.08 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 17:41.14 | p3nguin | newasterx: ^^^^^^^^^^^^^^^^ |
| 17:41.18 | newasterx | yes |
| 17:41.19 | newasterx | sorry |
| 17:46.44 | *** join/#asterisk silvestre_id (~silvestre@200.175.198.95) |
| 17:50.24 | silvestre_id | Someone could tellme how reset queue stats in asterisk 1.4.30? I can't update to newer version now... |
| 17:50.55 | *** join/#asterisk jly2680 (~jly@94.96.122.173) |
| 17:51.25 | chazzam | Doesn't restarting Asterisk clear stats? |
| 17:51.30 | jly2680 | aastra 4422 dhcp problems |
| 17:52.13 | pabelanger | silvestre_id: *CLI> queue reload? |
| 17:52.45 | silvestre_id | I need restart stats without restart asterisk. Older versions works with module reload app_queue.so, but in this version i cant find a way to do this |
| 17:53.20 | chazzam | I think I remember something stating about just that |
| 17:53.28 | chazzam | that behavior changed somewhere |
| 17:53.33 | pabelanger | https://issues.asterisk.org/view.php?id=17535 |
| 17:54.10 | silvestre_id | I just could "queue [add|remove|show]" |
| 17:54.27 | p3nguin | Odd. When I reload the queue module, my agent stats are reset. |
| 17:55.00 | p3nguin | "has taken no calls yet" |
| 17:55.42 | silvestre_id | pabelanger: this is for 1.6.*. I need backport this to 1.4? |
| 17:55.51 | pabelanger | p3nguin: Yes, there has been some recent discussion about it. |
| 17:56.27 | p3nguin | Maybe I misunderstood. I thought you all just said that reloading the module didn't reset the stats. |
| 17:57.28 | pabelanger | p3nguin: 1.4 only |
| 17:57.29 | pabelanger | https://issues.asterisk.org/view.php?id=13790 |
| 17:57.43 | pabelanger | silvestre_id: Yes, you would need to backport logic |
| 17:58.06 | p3nguin | I'm using 1.4.35 and my stats are reset when I reload the module. |
| 17:58.26 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 17:58.58 | pabelanger | Hmm, looks like its working again |
| 17:59.29 | p3nguin | It has always worked that way for me through all the 1.4 versions I have used. |
| 17:59.40 | silvestre_id | 1 Yeas ago, I use 1.4.21.2 and module reload app_queue.so reset the stats... but in 1.4.30 doesnt work... pabelanger I will try this backport in the weekend. Thanks |
| 18:00.01 | p3nguin | I always hated that the stats got reset. That's how I know it always reset them. |
| 18:00.51 | pabelanger | I'm understanding, is that is a bug and will be fixed shortly. Might want to get into the -dev discussion about it. |
| 18:00.52 | pabelanger | http://lists.digium.com/pipermail/asterisk-dev/2010-August/045792.html |
| 18:03.50 | silvestre_id | If in 1.4.30 doesnt reset ant for 1.4.35 reset... maybe just need to acept an additional flag to reset stats or not. |
| 18:04.13 | *** join/#asterisk neurosys (~neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
| 18:04.24 | p3nguin | I'm pretty certain that it reset stats for me in 1.4.30, too. |
| 18:04.29 | *** join/#asterisk tris (tristan@camel.ethereal.net) |
| 18:04.48 | p3nguin | Reloading the queue module has reset queue stats in every 1.4 version I have used. |
| 18:05.28 | *** join/#asterisk jasonwert (~w3rt@97-83-98-83.dhcp.trcy.mi.charter.com) |
| 18:06.43 | silvestre_id | The reset stats is good to do with logrotate. I do this all day at midnight. |
| 18:06.58 | newasterx | hai |
| 18:07.11 | p3nguin | all day... at midnight |
| 18:07.23 | newasterx | how to contact a issue reporter in issue.asterisk.org ? |
| 18:07.34 | *** join/#asterisk jasonwert (~w3rt@97-83-98-83.dhcp.trcy.mi.charter.com) |
| 18:09.02 | Corydon76-dig | p3nguin: language barriers |
| 18:09.50 | p3nguin | corydon76-dig: Yeah, I have that problem every day on Mondays. |
| 18:11.22 | Corydon76-dig | You can have any color car you want, as long as it's black |
| 18:11.44 | Corydon76-dig | (Henry Ford) |
| 18:16.08 | *** join/#asterisk kuku (~kuku@c-24-13-139-34.hsd1.il.comcast.net) |
| 18:16.25 | kuku | Anyone aware of RPM's for Centos of 1.4.23 and up ( not 1.6.x) |
| 18:17.13 | p3nguin | of course |
| 18:17.29 | Intel`` | asterisk RPM? |
| 18:17.34 | kuku | I can't seem to find any, I harly use RPM |
| 18:17.36 | kuku | yes. |
| 18:17.50 | Intel`` | hmm actaully i got mine from asterisknow yum repo |
| 18:17.59 | Intel`` | so its a matter of yum install asterisk14 |
| 18:18.09 | p3nguin | That's where I would have looked. |
| 18:18.14 | kuku | let me check |
| 18:18.20 | Intel`` | it will not install the gui |
| 18:18.31 | Intel`` | unless you do yum install freepbx |
| 18:19.15 | *** join/#asterisk lyetz (~lyetz@me.lyetz.me) |
| 18:19.22 | Intel`` | i can give you the .repo file if you want |
| 18:19.24 | kuku | I have asterisk installed |
| 18:19.29 | kuku | yes please |
| 18:19.42 | kuku | I just need to update from 1.4.22 to 1.4.23 |
| 18:19.53 | kuku | need AUDIOHOOK_INHERIT that was backported to 1.4 in 1.4.23 |
| 18:20.01 | p3nguin | Why such an old version? Current is 1.4.35. |
| 18:20.29 | kuku | The system is running well, don't want to introduce too many changes. |
| 18:21.18 | Intel`` | how did you install the old one? |
| 18:22.00 | Intel`` | not sure what will happen if you upgrade. just make sure you do it in a test environment first |
| 18:22.08 | Corydon76-dig | kuku: what about all the bugfixes to the backport since that time? |
| 18:22.23 | kuku | Intel``: trixbox... ( i know ) |
| 18:22.55 | kuku | Corydon76-dig: I'm worried trixbox have issues if you introduce too much. |
| 18:23.19 | kuku | Corydon76-dig: If that doesn't help, I'll wipe it clean, and get it working on 1.6 |
| 18:23.48 | Corydon76-dig | kuku: seriously, if you want to introduce just a few changes, you need to track the entire tree and review every patch that was committed since to see if there is anything that you need to have |
| 18:24.22 | Corydon76-dig | Maybe not the entire tree, but track the entire branch |
| 18:24.46 | Corydon76-dig | trixbox has issues, even if you introduce nothing at all |
| 18:25.01 | Naikrovek | yup |
| 18:25.22 | Intel`` | so trixbox or *now? =D |
| 18:25.25 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 18:25.40 | Naikrovek | *now |
| 18:25.42 | Corydon76-dig | I wouldn't recommend any GUI at all |
| 18:25.46 | Naikrovek | or that |
| 18:25.53 | kuku | Corydon76-dig: I understand. |
| 18:26.21 | kuku | Its not easy to admin 50 phones without gui. Especially if you need the client to review CDR's, recordings, reports, etc. |
| 18:26.31 | WIMPy | newasterx: Try to locate them in here. Otherwise, I don't know. |
| 18:26.34 | Intel`` | i believe these gui-based asterisk is great if its behind a firewall but not good if you will show it directly to the public |
| 18:26.45 | Corydon76-dig | A GUI can be helpful for rapid user add/change/delete, but anything else, you really need to understand the underlying mechanisms |
| 18:27.36 | kuku | I've been setting up asterisk since 2005, compiling. trixbox is a nice choice for some clients. |
| 18:27.49 | Corydon76-dig | You don't compile trixbox |
| 18:28.09 | kuku | I know. |
| 18:28.26 | Intel`` | yes you just do yum update to update everything |
| 18:28.28 | Corydon76-dig | Sorry, I missed the period |
| 18:28.39 | kuku | :) |
| 18:28.45 | p3nguin | corydon76-dig: You're pregnant! |
| 18:29.01 | Corydon76-dig | p3nguin: My bf will be so happy! |
| 18:30.10 | Intel`` | wait i thought you were talink about punctuations :D |
| 18:30.16 | Intel`` | *talking |
| 18:30.20 | Naikrovek | kuku: AsteriskNOW > Trixbox |
| 18:30.22 | Naikrovek | by far |
| 18:30.25 | Corydon76-dig | Intel``: I was. |
| 18:30.29 | Naikrovek | avoid trixbox if you can |
| 18:31.16 | Corydon76-dig | Intel``: You probably think I'm female, now, too |
| 18:31.24 | Intel`` | =)) |
| 18:31.41 | Intel`` | just a hint |
| 18:31.56 | kuku | Naikrovek: how so ? |
| 18:32.18 | Corydon76-dig | Intel``: 0 for 2 |
| 18:32.56 | *** join/#asterisk diegomad (~mad@190.147.221.78) |
| 18:33.47 | Naikrovek | kuku: yum update works. fewer things to break, it's FreePBX and not some fork of FreePBX like Trixbox uses, zero fonality customizations. |
| 18:34.45 | kuku | Back when I tested it, it was in beta, and had half the functions of trixbox |
| 18:35.12 | Intel`` | kuku for me the reason i switch to asterisknow is its more logical because digium and asterisk are partners and they ensure their products compatible with it. |
| 18:35.15 | Naikrovek | are you really going to base your opinion of the current version on a beta you tested years ago |
| 18:35.26 | Naikrovek | "digium and asterisk are partners" lol |
| 18:35.35 | kuku | here goes nothing: yum --skip-broken update asterisk.i386 |
| 18:36.09 | kuku | Intel``: based on my understanding, digium and asterisk are not partners... more like husband&wife |
| 18:36.24 | p3nguin | "digium and asterisk are partners" ??? What? |
| 18:36.27 | Naikrovek | digium is a software & hardware company. Asterisk is software. Digium produces Asterisk. Digium produces AsteriskNOW. |
| 18:36.58 | Naikrovek | and a lot of other stuff |
| 18:37.01 | p3nguin | That's like Toyota and Camry being partners. |
| 18:37.11 | Naikrovek | yep |
| 18:37.21 | Intel`` | ok i admit got the wrong word for that +D |
| 18:37.26 | Intel`` | XD |
| 18:37.45 | russ | more like canonical and ubuntu maybe? |
| 18:38.26 | Corydon76-dig | kuku: husband and wife? Nah, lesbian life partners |
| 18:38.42 | Intel`` | =)) |
| 18:39.08 | Corydon76-dig | Asterisk brought the U-haul on the first date |
| 18:39.38 | troy42 | nice |
| 18:41.09 | kuku | ok, so the asterisknow repos didnt help |
| 18:42.27 | Intel`` | i believe trixbox added some conf files not standard to asterisknow |
| 18:43.33 | chazzam | mv /etc/asterisk{,-trixbox} && mkdir /etc/asterisk ? |
| 18:43.45 | chazzam | then install the asterisk and freepbx packages from yum? |
| 18:44.43 | kuku | I just need the core asterisk to be 1.4.23 from 1.4.22 |
| 18:45.12 | p3nguin | You could always roll your own. |
| 18:46.04 | kuku | but I still need trixbox to work :0 |
| 18:47.21 | Naikrovek | back up the machine |
| 18:47.22 | Naikrovek | yum update |
| 18:47.27 | Naikrovek | and see if it works |
| 18:47.46 | Naikrovek | if it doesn't (it won't) restore from backup, try another method |
| 18:47.52 | kuku | I'm working off of a cloned drive now. yum update, wants to kernel i686 as a dependancy, I have 386 kernel |
| 18:49.04 | p3nguin | hmm? |
| 18:49.12 | p3nguin | kernel.i686? |
| 18:49.33 | kuku | yes. |
| 18:50.27 | kuku | nvm |
| 18:53.01 | kuku | http://pastebin.ca/1922181 |
| 19:07.17 | *** join/#asterisk Intel`` (~DND@86.99.229.224) |
| 19:15.07 | *** join/#asterisk brendansterne (~brendanst@cpe-70-124-61-17.austin.res.rr.com) |
| 19:16.52 | brendansterne | Greetings. I have patched my asterisk with a new feature - the ability to configure (via contacthost in sip.conf) the SIP Contact header host part. |
| 19:17.08 | brendansterne | I'm wondering if this would be helpful to others |
| 19:18.00 | brendansterne | And how I might go about progressing this feature into asterisk. |
| 19:20.03 | *** join/#asterisk matagou_ (57f8bf49@gateway/web/freenode/ip.87.248.191.73) |
| 19:20.13 | matagou_ | hello, asterisk users |
| 19:20.34 | matagou_ | have a problem regarding CLIP on pstn lines |
| 19:20.46 | matagou_ | connected to AEX800 card |
| 19:20.55 | drmessano | brendansterne: Versus in the register string? |
| 19:20.58 | matagou_ | asterisk 1.6.2.9 |
| 19:21.09 | matagou_ | dahdi 2.3.0.1 |
| 19:21.53 | matagou_ | PSTN provider told that CLIP-DTMF and CLIP-FSK is supported on their carreers |
| 19:21.53 | brendansterne | drmessano: Right now asterisk fills the SIP Contact header with <sip:user@hostip:port>, I needed to be able to set it to <sip:user@hostname:port> |
| 19:22.41 | brendansterne | instead of <sip:123@10.10.10.10> I needed <sip:123@myasterisk.mycompany.com> |
| 19:22.46 | kuku | brendansterne: maybe #asterisk-dev |
| 19:22.55 | brendansterne | Ahhh... ok |
| 19:22.59 | brendansterne | I'll check asterisk-dev |
| 19:24.54 | matagou_ | should i provide the chan_dahdi.conf? |
| 19:25.02 | b14ck | sup guys |
| 19:25.44 | *** join/#asterisk hopper75 (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 19:39.58 | Naikrovek | ~sipnat |
| 19:39.58 | infobot | rumour has it, sipnat is Quick guide on configuring Asterisk + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 19:40.04 | Naikrovek | brendansterne: ^^^^^^^^^^^^^ |
| 19:47.26 | Naikrovek | drmessano: i have you ignored but i'm sure you'll be happy to see me trolling you again |
| 19:53.54 | kuku | UPDATE: I compiled asterisk 1.4.23 from source, and it works so far with trixbox 2.6 without breaking it. |
| 19:54.14 | kuku | peace to all- I'm out. |
| 19:54.28 | path | Hey guys. Need help in here!. Ive just setup an ivr and when Background app finishes I cant do any DMTF. Though if I press before it finishes the sound file, I can go through the menu. |
| 19:54.45 | path | I already tried setting timeout response |
| 19:54.56 | chazzam | Try WaitExten(#) ? |
| 19:55.08 | chazzam | You have to have something listening... |
| 19:55.29 | path | http://pastebin.com/QUDvZji0 thats my voicemenu |
| 19:55.49 | path | eeh no I just wrote Wait to pause the dialplan |
| 19:56.22 | chazzam | Wait doesn't listen |
| 19:56.26 | chazzam | WaitExten does |
| 19:57.54 | path | worked like a charm! thanks chazzam |
| 19:58.04 | chazzam | np, take luck! |
| 19:58.13 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 19:58.15 | path | :-) |
| 19:59.09 | *** join/#asterisk lost_soul (~noymfb@cpe-67-249-141-227.twcny.res.rr.com) |
| 20:05.24 | *** join/#asterisk odenkos (~odenkos@ip-212-081-019-170.static.nextra.sk) |
| 20:16.44 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 20:16.58 | *** part/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 20:18.06 | EmleyMoor | has had a couple of REGISTER attacks launched against his Asterisk box tonight |
| 20:18.28 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 20:18.44 | *** part/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 20:23.25 | hardwire | strange. |
| 20:24.00 | hardwire | I think I'm experiencing some funky bridging |
| 20:24.05 | hardwire | even though I'm transcoding |
| 20:25.08 | hardwire | calls over IAX go to a remote PBX with a TDM400 in it.. the TDM400 has some FXS modules |
| 20:25.43 | hardwire | when the call is bridged I get choppy/choppered audio to the standard telephone |
| 20:25.46 | hardwire | from works fine |
| 20:26.04 | hardwire | however if I call over IAX to SIP (transcoding along the way as well) it's fine |
| 20:26.16 | hardwire | and local calls between the TDM400 and the * server appear fine (echo test) |
| 20:26.23 | hardwire | echo cancel is completely disabled |
| 20:26.26 | hardwire | how.. strange |
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| 21:03.54 | jamko | I brought this up yesterday but would like to hear some more debate on it. What things work better in realtime, and what things better with static realtime. |
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| 21:26.54 | pabelanger | jamko: Depends on what you need to do. |
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| 21:43.53 | jamko | pabelanger: I would like for multiple * boxes to share a central DB, mainly for ease of swap in/swap out, and for a desire to have uniformity across all. Having to reload * is not really a huge deal at this point, but one less thing to do is always nice. |
| 21:46.08 | jamko | However, I am concerned a little about a lost connection to the db server with dynamic realtime resulting in all boxes going down.. |
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| 21:55.43 | Gary_B | what kind of setup would be required for a business critical asterisk pbx setup in a small company (2-6 employees/receptionists). Pika sounds good and cheap but i hear wouldnt be a reliable solution |
| 21:55.53 | Gary_B | are there any appliances? |
| 21:57.19 | p3nguin | What's wrong with Asterisk? |
| 21:58.25 | p3nguin | Let me rephrase. What's wrong with using Asterisk for that scenario? |
| 21:58.52 | Gary_B | asterisk is a given, its the hardware it would run on |
| 21:59.26 | p3nguin | Get a decent business grade desktop PC. |
| 22:00.52 | Gary_B | pabelanger from #askozia has mentioned going for a PC with dual power supplies, RAID 1/5 HDD |
| 22:02.01 | Gary_B | its for a taxi company, ie if the pbx fails, the business stops |
| 22:02.10 | p3nguin | I probably wouldn't bother with the dual power supply thing... that could prove to be a huge expense. I would, however, keep a spare PSU on hand just in case. It should only take a couple of minutes to swap a power supply. |
| 22:02.35 | Gary_B | yea but it would be unattended |
| 22:03.00 | Gary_B | with maybe a 12 hour service level for hardware replacement |
| 22:03.09 | p3nguin | I would probably consider RAID 1. |
| 22:03.35 | p3nguin | In an unattended location, the price of the dual power supply may be justified. |
| 22:04.17 | WIMPy | Setup two complete PCs and switch on the spare one if the first one fails. |
| 22:05.09 | Gary_B | Wimpy: that couldnt be done till i got there either |
| 22:05.56 | WIMPy | Why is it located where not even someone could press a button? |
| 22:06.24 | Gary_B | taxi offices |
| 22:06.41 | WIMPy | You'd need monitoring then. Also for a redundant PSU. |
| 22:06.46 | Gary_B | with just receptionists on 24hr shifts |
| 22:08.12 | WIMPy | Should be enough to press a button. But it could be automated. |
| 22:10.16 | Gary_B | ive put this idea on hold numerous times before. The typical office would just have say 4 lines and 2 phones/receptionists. Telephony side i just need to capture the CLI of calls answered, the problem part requiring a pbx is matching the clis to the receptionist who answers the call, a cli capture box cant do that. |
| 22:11.23 | Gary_B | i saw Askozia and though a simple embedded pbx device would be the answer, but thats not a reliabe solution |
| 22:11.40 | Gary_B | *thought |
| 22:11.41 | pabelanger | what is your budget? |
| 22:12.52 | pabelanger | That will dictate your 'reliability' for your system |
| 22:13.44 | Gary_B | pabelanger: im unsaure of the budget. Rather its the "reliability" that will determine whether i go ahead or not! |
| 22:17.58 | pabelanger | Well, determine what is an acceptable downtime if a problem was to happen, 5 / 30mins, 1 / 4hours, 1day, etc. Then spec a system from there. You may just get 2 system, and cold swap the 2nd box if the first fails, however that requires somebody onsite. |
| 22:18.58 | pabelanger | For me, I've used Dell 2950 in the past when clients required redundancy. |
| 22:19.16 | pabelanger | in the range of $5000 a box |
| 22:24.00 | Gary_B | pabelanger: your right id need to look at the requirements again |
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| 22:32.00 | Gary_B | Wimpy: i see what you mean RE a redundant PC setup, http://www.voipon.co.uk/xorcom-twinstar-p-2284.html |
| 22:34.33 | Gary_B | Wimpy: that particular setup looks expensive, but interesting |
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| 22:36.12 | WIMPy | Gary_B: It all just depends on how much you want to spend. Do you have a reasonably big UPS in your budget, for example? |
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| 22:53.13 | Gary_B | WIMPy: that needs factored in as well, i was thinking of a managed service priced at x amount per month, the question also is how many months do you go without making money, would the monthly cost a small business is willing to pay be worth it |
| 22:57.47 | b14ck | starting an asterisk hosting service? oO |
| 23:03.15 | Gary_B | no. is there money to made from starting up a service like that anymore? |
| 23:05.23 | b14ck | I don't think so. Not with twilio / tropo, etc. |
| 23:05.57 | Gary_B | pbxes.com is a briliiant site, i found the quality of the service has been degrading recently thoug |
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| 23:08.26 | Mark22 | Gary_B: it is possible to make money with it, however the quality has to be great (and prices low) so making money will be hard if you don't have enough volume |
| 23:13.53 | jamko | anyone have any stats on how much bandwidth sip uses for signalling etc?... Not including any audio, just for sip. |
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| 23:42.25 | Gary_B | if i place a standard PSTN phone call from one computer to another, i could use dtmf codes to transmit decimal number, this would take quite a long time relatively, is there a quicker but reliable way to transmit a number, could i cut the time of each dtmf "noise" in half and it still be reliable? |
| 23:44.05 | Gary_B | in essance im talking about setting up an old modem to modem system, but i would be routing the call through asterisk at one end to generate the noise/tome using tap ins at the other end to listen to the tones |
| 23:44.57 | Gary_B | how quickly could i send a single decimal digit across such a setup? |
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| 23:54.06 | Gary_B | im basically looking for a more efficient alternative to dtmf |
| 23:54.59 | WIMPy | isdn |
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| 23:56.09 | Gary_B | i mean sending encoded information between computers over an open standard pstn call |
| 23:56.17 | Gary_B | or isdn call |
| 23:56.52 | WIMPy | uus? |
| 23:57.14 | Gary_B | ? |
| 23:57.21 | Gary_B | uus? |
| 23:58.38 | Gary_B | sorry, the use would be for computers to communicate some basic information between the 2 parties with the minumum interuption to the talking humans |
| 23:59.47 | Gary_B | ie send some information across at the start of the call, just a few digits, dtmf would be an option but that would be very noticable to the humans |