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| 02:33.26 | mmourani | hi all |
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| 02:34.27 | mmourani | I need to compare the callerid of two incoming calls at the same time |
| 02:34.42 | mmourani | SIP calls |
| 02:35.03 | mmourani | how do I do that on asterisk 11 |
| 02:35.10 | Penguin | Compare them to what, each other? |
| 02:35.15 | mmourani | yes |
| 02:35.49 | Penguin | One will have to exist before you can compare the other one to it. |
| 02:35.54 | Penguin | So "at the same time" could be a problem. |
| 02:36.05 | mmourani | I mean I receive one call |
| 02:36.12 | mmourani | and then another one right after |
| 02:36.25 | mmourani | and I need to compare the value |
| 02:37.30 | Penguin | That shouldn't be too hard, then. Save the value of the first call into either a global variable or into the astdb. |
| 02:37.38 | Penguin | Then compare the second call to the stored data. |
| 02:37.59 | mmourani | but i don't want to answer the call |
| 02:38.17 | mmourani | I want to compare first and then decide if I cancel them or I dial |
| 02:38.17 | Penguin | Caller ID exists with or without an ANSWER. |
| 02:38.31 | mmourani | ok |
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| 08:39.48 | Tokeiito | hello |
| 08:41.08 | Tokeiito | when i'm configuring Hylafax with iaxmodem (as i understand this one just emulates fax interface), i need to provide fax number. not sure what type of number it needs. should it be DID number used by asterix inbound route or it can be number created by extention? |
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| 08:56.11 | nettie | Hi guys, I'm using a Sangoma B700 BRI card with asterisk 1.6.x and the latest dahdi/libpri/wanpipe. I'm experiecing consistent one way choppy audio. During an external call originated from asterisk to a third party, the caller hear perfectly but the called party hears a very choppy/scattered voice. The same issue is present while faxing, receiving faxes is 100% OK but sending them is impossible. It's a very strange issue I never encounter, I'm wondering if any |
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| 09:05.58 | Chainsaw | nettie: The description of the audio artifacts sounds like the G722 bitrate mismatch. |
| 09:06.32 | Chainsaw | nettie: Have you tried limiting the audio path to just ulaw & alaw? |
| 09:07.15 | nettie | Chainsaw if that's the case I'm wondering why it affects just a single way then? if it's a codec issue it will affect both ways no? |
| 09:07.18 | nettie | all is set to alaw |
| 09:07.33 | nettie | but that's on the sip endpoint |
| 09:07.42 | Chainsaw | nettie: If all is set to alaw and no transcoding is occurring, my theory is invalidated. |
| 09:07.50 | nettie | the issue is more related to the Sangoma B700 / wanpipe board |
| 09:08.06 | Chainsaw | nettie: If you're already sure what the issue is, I don't see what I can add. Good luck. |
| 09:08.24 | nettie | sorry I left an "I think" |
| 09:08.25 | Chainsaw | returns to the reception desk |
| 09:08.28 | nettie | :) |
| 09:08.50 | nettie | I'm not 100% sure, I have a ticket open with Sangoma as well |
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| 09:27.11 | defswork | anyone see any problems with wanting to do the following - automated dialler (record answered state), present a recorded message with some options (press 1, 2 etc..), record somehow the options chosen (one path is to rate out of 5 something) and then allow a message to be recorded if needed |
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| 09:37.02 | ipalmer | morning all, I'm running asterisk 11.5 and when I try to dial a SIP address using Dial(SIP/ipaddress/exten) I receive the following message exceptionally long voice queue length queueing to Local/603@directdial. Anyone know what this means please? |
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| 10:27.21 | snadge | can a single asterisk server listen on 2 seperate network interfaces and ip addresses? |
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| 10:31.12 | SunTsu | snadge: AFAIK it's "all addresses" aka 0.0.0.0/0 or a single ip address |
| 10:31.45 | snadge | then i dont see why an asterisk server couldn't listen on both a private subnet, which is used for communicating via RNAT to a citrix netscaler |
| 10:32.03 | SunTsu | snadge: I might not be uptodate, though, and you really should read your version's docs regarding bindassr |
| 10:32.06 | SunTsu | bindaddr |
| 10:32.06 | ipalmer | morning all, I'm running asterisk 11.5 and when I try to dial a SIP address using Dial(SIP/ipaddress/exten) I receive the following message exceptionally long voice queue length queueing to Local/603@directdial. Anyone know what this means please? |
| 10:32.08 | snadge | and the existing network interface that it uses currently |
| 10:32.43 | snadge | the other sysadmin i've been having discussions with.. seems to think that asterisk doesn't like doing that |
| 10:32.53 | snadge | multiple interfaces/ips |
| 10:33.05 | snadge | but from what i've found by googling.. others do have success with it |
| 10:33.44 | SunTsu | snadge: I know that there is/was an issue with multiple ip addresses on a single interface, but that, too, might be no longer the case |
| 10:34.01 | snadge | ahh, that might be what he was talking about.. thanks |
| 10:34.11 | snadge | there is light at the end of the tunnel ;) |
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| 11:03.15 | phix | fakhir likes haemodrine |
| 11:03.58 | fakhir | ? |
| 11:04.26 | WIMPy | Sounds like medicine. |
| 11:04.52 | phix | Depends if you think Marijuane is also medicine |
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| 11:05.06 | phix | Marijuana*| |
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| 12:38.27 | snadge | anyone else here successfully using L7 load balancing with RNAT in front of asterisk? |
| 12:38.51 | snadge | or is that a taboo subject |
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| 12:42.28 | snadge | because if you were.. presumably that would be because of potentially gigabit traffic |
| 12:43.46 | snadge | even 100mbit or so is a challenge |
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| 12:45.32 | snadge | other people are turning to things like opensip for scaleability.. but opensip != asterisk, clearly |
| 12:47.48 | snadge | im going to try sticking netscaler in front of asterisk and see what happens.. a pretty scientific approach |
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| 13:48.12 | rox | hello, I was using Asterisk 1.2 and 1.8 for a while, and now I am trying out asterisk 11 for the first time. How do I get Verbose() messages from my dialplan to go to the messages log file? in Asterisk 1.2 and 1.8 Verbose() messages went messages, while in asterisk 11 I can't get them to appear there |
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| 13:50.02 | rox | Do I have to replace the Verbose() call with something else? |
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| 13:57.16 | [TK]D-Fender | Should simply have to set your logger.conf accordingly |
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| 13:59.29 | rox | [TK]D-Fender: well, I have tried, but in order to get contents of Verbose() calls, i need to have verbose level in messges file, which is far too much |
| 13:59.53 | rox | [TK]D-Fender: I replaced Verbose() calls with Log(NOTICE,...) and it does the trick |
| 13:59.59 | [TK]D-Fender | rox: You pick which files it logs to |
| 14:00.24 | rox | [TK]D-Fender: right, i am refering to the astlogdir/messages file |
| 14:01.13 | [TK]D-Fender | rox: that depends on our install. You don't have to use that file if you don't want to, and cn split them up however you like |
| 14:01.17 | rox | in logger, if my log level is set like this: messages => notice,warning,error then I don't get the Varning messages |
| 14:01.30 | rox | but if console is confuigured the same way, I do get them |
| 14:02.12 | rox | console => notice,warning,error caputers the Verbose(...) messages, but messages => notice,warning,error does not |
| 14:02.24 | [TK]D-Fender | hrm |
| 14:03.19 | Greenlight | That's just the default settings on the console, isn't it, they can get chaged by a "core set verbose xxxx" |
| 14:03.27 | rox | did something change in the semantics of Verbose command? The documentation doesn't mention anything. |
| 14:04.26 | rox | Greenlight: right, but I did not change them, or even with "core set verbose 0" console captures Verbose(..) messages from Dialplan |
| 14:04.47 | rox | the weird thing for me is that messages file does not, I was used to them both displaying those messages |
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| 14:08.32 | rox | thank you very much |
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| 14:25.33 | rbd_ | hey guys...is it possible to use Monitor() or Record() while MixMonitor is running on a channel (i.e. I am recording the whole call for any necessary error analysis, but we are doing prompt-level similarity matching, and need to record out specific parts of the call to different files so that we can compare them to other files we have) |
| 14:26.37 | mjordan | rbd_: Should be, yes. MixMonitor puts a frame hook on the channel and just intercepts and records the audio. |
| 14:26.48 | Greenlight | Sure, you'd be better using two mixmonitors though |
| 14:27.43 | Greenlight | Eg, a first MixMonitor for the whole call, and then a second or third MixMonitor for the snippets you want separately |
| 14:27.57 | rbd_ | thanks guys...well, for the first call, having it stop on hangup is perfect...but with the second Monitor/Record/MixMonitor call I want to be able to start and stop it.... |
| 14:28.20 | rbd_ | in that case, Monitor and StopMonitor seem right?? |
| 14:28.44 | rbd_ | or will StopMonitor stop monitoring my first MixMonitor invocation as well? |
| 14:29.02 | Greenlight | This from AMI >? |
| 14:29.08 | rbd_ | no, from extensions.conf |
| 14:29.33 | Greenlight | How do you intend to trigger the StopMonitor mid call ? |
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| 14:30.11 | rbd_ | well... start monitoring, wait a certain length of time, then stop monitoring somehow and save off/process the file |
| 14:30.46 | rbd_ | maybe Record() then with a max duration and no beep? |
| 14:31.18 | Greenlight | BUt you'll be "inside" a Dial presumably |
| 14:31.35 | rbd_ | I'm using call files, sorry |
| 14:32.00 | rbd_ | which take the dialed call to a handler in extensions.conf |
| 14:32.17 | Greenlight | Okay, and what do you "do" with that call once it his there? |
| 14:32.25 | Greenlight | Just iirc, Record() "blocks" |
| 14:33.40 | rbd_ | start mix monitor...then process commands (the app basically tests IVRs to make sure they function correctly by acting like a human interacting with the IVR). I think record blocking in this case is fine because I just want to record what the IVR is saying at a specific time, and then fuzzy compare it to a prompt recording I have for a similarity score... looking at alternatives to straight |
| 14:33.40 | rbd_ | ASR |
| 14:34.09 | Greenlight | Okay, in that case Record will work yea :) |
| 14:34.10 | rbd_ | we don't have to do anything while the record() call is going on in this case... we will process it quickly after we have the length of the recording we need |
| 14:34.29 | rbd_ | ok, great, thank you!! |
| 14:35.12 | Greenlight | You might be able to use the "silence" argument to trim it more neatly as well |
| 14:41.16 | rbd_ | ok, I'll check that out |
| 14:43.17 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-bnhvwzbuqppjkasf) |
| 14:43.17 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 14:54.39 | *** join/#asterisk ve6ts (nj@S01060010181c5856.cg.shawcable.net) |
| 14:56.06 | ve6ts | does anyone know if there is a way to read how many channels are active on a sip peer, asterisk must have this number because it has call-limit, is this variable accessible? |
| 14:56.32 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 14:56.41 | ve6ts | currently i'm parseing the output of core show channels to find out, but there has got to be a better way |
| 15:00.18 | *** join/#asterisk _Corey_ (~chatzilla@50-200-184-54-static.hfc.comcastbusiness.net) |
| 15:01.56 | *** part/#asterisk ve6ts (nj@S01060010181c5856.cg.shawcable.net) |
| 15:05.53 | Penguin | I'm not sure if that's something you can query or not. If not, you could use the CHANNEL() function to make your own counter. |
| 15:13.23 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 15:16.58 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 15:20.38 | *** join/#asterisk fatemeblue (~fateme@85.15.14.88) |
| 15:21.12 | fatemeblue | how to set permit in sip.conf asterisk? |
| 15:21.51 | kaldemar | SIPPEER(curcalls) has that if call-limit is set. but ve6ts was too hasty to get that information. and the note that call-limit is deprecated. |
| 15:22.20 | newtonr | fatemeblue, "permit=1.2.3.4/255.255.255.255" |
| 15:22.21 | kaldemar | fatemeblue: what do you want to permit and what have you tried so far? |
| 15:22.28 | Penguin | fatemeblue: It is often best to deny everything and then permit only the one address you want to allow. deny=0.0.0.0/0.0.0.0 permit=1.2.3.4/255.255.240.0 |
| 15:23.20 | fatemeblue | i make a sip account but it does not register , i think i need to set permit but i dont know how |
| 15:23.45 | Penguin | If you did not deny anything, you do not need to explicitly permit anything. Everything is already permitted. |
| 15:24.07 | fatemeblue | so what should i do? |
| 15:24.41 | fatemeblue | i receive 408 error |
| 15:25.53 | Penguin | Sounds like a firewall problem to me. Forward port UDP:5060 to your asterisk IP address and also the RTP UDP range. |
| 15:26.27 | fatemeblue | i turn id off completly |
| 15:26.44 | kaldemar | receive 408 where? |
| 15:26.46 | newtonr | fatemeblue, you probably want to look at Asterisk logs to see if the REGISTER is hitting your Asterisk system and what Asterisk says about it |
| 15:27.05 | *** join/#asterisk vandyk (~vandyk@189.59.4.180) |
| 15:27.13 | newtonr | fatemeblue, https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information to get you started with the logs |
| 15:28.06 | fatemeblue | ok |
| 15:28.08 | fatemeblue | thanks |
| 15:28.17 | fatemeblue | i will look at it |
| 15:29.14 | newtonr | fatemeblue, its hard to say without looking at debug.. it could be a firewall/NAT/networking issue, it could be misconfiguration of the SIP peer or your client, etc |
| 15:29.55 | newtonr | fatemeblue, if you post your debug in pastebin where we can see it, we may be able to help out as well |
| 15:32.12 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 15:35.02 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 15:35.02 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 15:56.25 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:226:21b:63ff:fe31:8426) |
| 15:59.32 | fatemeblue | how can i enable debug logger |
| 15:59.34 | fatemeblue | ? |
| 15:59.42 | Penguin | see logger.conf |
| 15:59.54 | fatemeblue | how can i open it |
| 15:59.57 | *** join/#asterisk serafie (~erin@nat/digium/x-wygceaetpgarcwmr) |
| 15:59.58 | Penguin | vim |
| 16:00.05 | fatemeblue | its emtey |
| 16:00.17 | Penguin | Find the sample file. |
| 16:01.51 | fatemeblue | what do you mean by sample file |
| 16:01.52 | fatemeblue | ? |
| 16:02.05 | Penguin | Samples of all the config files are provided with asterisk. |
| 16:02.30 | *** join/#asterisk SunTsu (miyamoto@unaffiliated/suntsu) |
| 16:05.13 | *** join/#asterisk saxa (~sasa@2-234-206-126.ip224.fastwebnet.it) |
| 16:07.10 | *** join/#asterisk phonebuff (~chatzilla@173-15-220-185-BusName-Atlanta.hfc.comcastbusiness.net) |
| 16:09.29 | *** join/#asterisk ledoktre (~ledoktre@216.51.155.182) |
| 16:11.58 | ledoktre | greetings fellas. a quick question on MWI, BLF indicators. If I have a small VPS setup with 3 users, one behind a net gear router, two behind a drink router, the drink users are receiving MWI and BLF just fine, but the net gear one is not, can it be the net gear? The phone appears to work fine but the indicators seem to never even attempt to register with Asterisk |
| 16:13.00 | [TK]D-Fender | drink? |
| 16:13.39 | [TK]D-Fender | Also show us the device registering and subscribing.... |
| 16:13.47 | ledoktre | sorry, the auto spelling DLINK not drink |
| 16:17.50 | *** join/#asterisk _Corey_ (~chatzilla@50-200-184-54-static.hfc.comcastbusiness.net) |
| 16:30.04 | ledoktre | [TK]D-Fender: I seem to figure out part of the problem. By plugging the phone in directly to the internet, everything registered fine. Problem appears to be in the router (ALG perhaps?) |
| 16:30.06 | *** join/#asterisk gops (b869e022@gateway/web/freenode/ip.184.105.224.34) |
| 16:30.55 | [TK]D-Fender | If there is an ALG... you should see it and it should not be a guess. |
| 16:31.23 | ledoktre | yes, in the router there was a ALG setting. I just wanted to verify the issue was not my configuration. It is in the router somehow |
| 16:32.14 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 16:33.23 | gops | on heavy channel utilization chan_sip got freezed... and I got a error like this - Oct 17 14:25:00] WARNING[28866][C-0001a05a] chan_sip.c: Unable to cancel schedule ID 1714357. This is probably a bug (chan_sip.c: stop_session_timer, line 28785). |
| 16:33.30 | gops | is this due to cpu load? |
| 16:37.48 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 16:38.04 | *** join/#asterisk v0lZy (~Thunderbi@84-255-194-41.static.t-2.net) |
| 16:40.15 | *** join/#asterisk k610 (~K610@cable-78.29.241.186.coditel.net) |
| 16:41.36 | *** join/#asterisk [sr] (~kvirc@193-126-22-224.static.optimus.net.pt) |
| 16:41.44 | [sr] | hi |
| 16:42.19 | [sr] | i have some ATA's, with FXS ports (of course), but analog phones that can do call transfer, on the ATA's can't, what's the name of this feature? |
| 16:42.22 | [sr] | any hint? |
| 16:43.14 | [sr] | WIMPy: any idea? |
| 16:43.15 | [sr] | :) |
| 16:44.48 | [TK]D-Fender | [sr]: Your wording is confusing, please rephrase |
| 16:45.33 | [sr] | hum ok let me try |
| 16:45.48 | [sr] | i have an Voip ATA, cisco SPA8000 in case |
| 16:46.15 | [sr] | everythings fine, analog phones connected work ok, but if i try to transfer a call, doesn't work |
| 16:46.36 | [sr] | i know this is isn't really asterisk related, its something on the ATA |
| 16:47.33 | *** join/#asterisk cmendes0101 (~cmendes01@office.phone.com) |
| 16:47.51 | WIMPy | Analog phones can't do call transfers. Whatever they are connected to might be able to. |
| 16:50.23 | [sr] | WIMPy: so when i hit the "R" key, the TA must be configured for some feature code or so right? |
| 16:50.37 | WIMPy | And analog has not just opne standard for "sending commands". |
| 16:51.08 | WIMPy | R keys have different functions. They kan be ground keys or various lengths of flash signals. |
| 16:51.27 | WIMPy | You need to make sure both ends are compatible. |
| 16:52.09 | *** join/#asterisk navaismo (~navaismo@189.191.237.126) |
| 16:52.29 | [sr] | think so |
| 16:57.53 | [sr] | have to dig on this better |
| 17:05.11 | [TK]D-Fender | SPA8000 supports hook-flash transfers |
| 17:05.51 | gops | will cpu load freeze the chan_sip module.. I got this error "Unable to cancel schedule ID 1714357. This is probably a bug (chan_sip.c: stop_session_timer, line 28785)" and all the 240 channels got stuck |
| 17:06.34 | paulc | [sr] What kind of phone? I know flash durations are different in Europe (100ms) versus in North America (300ms+) |
| 17:07.56 | [sr] | paulc: its a siemens analog, desk phone |
| 17:08.21 | paulc | [sr] Where in the world? ;-) |
| 17:08.34 | WIMPy | Yaou can probably program the funcion of the R key. |
| 17:09.22 | [sr] | paulc: portugal |
| 17:09.50 | [sr] | WIMPy: that's something i'd dont want, since there's alot of different phone types, tought this was something on the ATA side |
| 17:09.59 | paulc | Ok, so it's probably 100ms timed line break. There's a setting in the SPA8000 that has min and max durations I think.. |
| 17:10.05 | [sr] | when i press "R" i hear something, but nothing happens |
| 17:10.18 | paulc | And I'll bet the default is higher than 100ms.. so your R press doesn't register.. |
| 17:10.22 | paulc | let me go look at my ATA112 here |
| 17:10.25 | [sr] | paulc: with which name ? |
| 17:11.22 | paulc | [sr] Give me 2 secs - I'm just looking at mine here |
| 17:11.34 | [sr] | paulc: i have also an ATA112 here! |
| 17:12.52 | paulc | In the ATA112 it's called "Hook Flash Timer Min" and "Hook Flash Timer Max" - with mine set to 0.1 for Min and 0.9 for Max. So if your R button is 100ms (common in Europe, provided it's timed line break (TBR) and not earth recall (you DO hear a click, right?)) then I'd suggest changing those numbers |
| 17:13.05 | paulc | maybe 0.08 and 0.12 |
| 17:13.28 | paulc | the max can be longer and not an issue.. it's really the minimum that I think is causing your troubles |
| 17:14.06 | [sr] | paulc: where are you seing that? on the line1/2 menu ? |
| 17:14.15 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 17:14.30 | paulc | [sr] sorry - I should have said.. it's in Voice --> Regional |
| 17:14.31 | navaismo | no im not longer |
| 17:14.40 | paulc | under "Control Timer Values (secs)" |
| 17:15.05 | [sr] | ah yes, it has 0.1 |
| 17:15.18 | [sr] | for min and 0.9 for max |
| 17:15.42 | gops | will cpu load freeze the chan_sip module.. I got this error "Unable to cancel schedule ID 1714357. This is probably a bug (chan_sip.c: stop_session_timer, line 28785)" and all the 240 channels got freezed...! |
| 17:15.52 | gops | will cpu load freeze the chan_sip module..? I got this error "Unable to cancel schedule ID 1714357. This is probably a bug (chan_sip.c: stop_session_timer, line 28785)" and all the 240 channels got freezed...! |
| 17:18.01 | WIMPy | The default is probably 90ms, not 100. |
| 17:18.44 | WIMPy | Might even be 80. |
| 17:18.57 | [sr] | paulc: i love you! |
| 17:19.07 | paulc | <3 :) |
| 17:19.36 | paulc | WIMPy: True that.. certain "cheaper" PBXs would support rotary 1 the same as R/timed line break |
| 17:19.48 | [sr] | paulc: other thing is, when i dial a number on the SPA112 like yours, after the number dial, there's a delay of about 5 secs, is that tunneable also? |
| 17:19.52 | paulc | Begs the question - how can you measure the duration of the TBR easily.. |
| 17:20.30 | WIMPy | I found an old news post that suggests the save values to avvept are 50-310ms. |
| 17:20.40 | paulc | [sr] YES! That's your dialplan.. If you know the pattern of standard numbers, you can make it match as soon as it's got the right number of digits. You can lower the timeout too, but 5 is fairly reasonable (when the length isn't known, like 00+ international calls for example) |
| 17:21.21 | paulc | [sr] If Portugal's dialplan is anything like the UK, there are probably some simple/standard rules for local, long distance, and international dialing.. makes the dialplan fairly easy.. you just need to know the dialing rules for where you are :-) |
| 17:21.35 | [sr] | paulc: thats not that, when i dial a number, no matter if an internal ou external, there's a delay since it start dialling |
| 17:23.03 | [sr] | ex, i pickup the fone, i get dial tone, dial a number, after the last number, there's a delay till it start ringing |
| 17:24.53 | paulc | [sr] Yes, that's your dialplan for sure. If the ATA knows how many digits to dial, it can send the SIP INVITE message off as soon as it knows "Ah, I've got enough digits now - let's make the call!" |
| 17:25.17 | paulc | Right now it's doing more "Have I got enough digits? have I got enough digits? oh, they haven't pressed any for a few seconds.. must be the end of the number.. " |
| 17:32.08 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 17:32.26 | *** join/#asterisk ChannelZ-Wk (~bobm@spark.idolum.com) |
| 17:33.59 | WIMPy | Yes, a proven ideal way to annoy your users on every single call. |
| 17:37.07 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 17:37.07 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 17:48.18 | paulc | We took it private to have a chat about the power of the dialplan on ATAs :-) |
| 17:49.09 | paulc | Dialplan POWERRRRRRR |
| 17:50.11 | paulc | I recently had an issue where I wanted to use *2663 as an extension number and couldn't.. I'm like WTF? then realised the dialplan allowed *xx and **xx. so it became **2663. Easier than changing provisioning for all the phones. |
| 17:51.27 | WIMPy | prefers to use phones that don;t require such a horrible hack. |
| 17:52.22 | paulc | Such as..? (these were SPA508Gs, and there are a couple of *xx codes used internally, so I didn't want to conflict) |
| 17:52.42 | [TK]D-Fender | change the phones... |
| 17:52.51 | [TK]D-Fender | (codes that is) |
| 17:53.43 | paulc | Yeah.. doable.. but we ended up with a DSS/BLF for DND and CFWD.. works really well.. shows you if the feature is on or off, and pressing the button takes you to a simple IVR to change settings. |
| 17:54.22 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 17:55.21 | *** part/#asterisk ledoktre (~ledoktre@216.51.155.182) |
| 17:56.43 | *** part/#asterisk fatemeblue (~fateme@85.15.14.88) |
| 17:56.48 | *** join/#asterisk fatemeblue (~fateme@85.15.14.88) |
| 17:58.36 | fatemeblue | can any body provide a document for asterisk configuration |
| 17:58.49 | pabelanger | ~book |
| 17:58.49 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 17:58.53 | pabelanger | fatemeblue, ^ |
| 18:00.44 | fatemeblue | thanks |
| 18:03.38 | fatemeblue | these are not free |
| 18:07.17 | anonymouz666 | LOL |
| 18:07.27 | anonymouz666 | why people always require things to be free? |
| 18:07.35 | [TK]D-Fender | It is free |
| 18:07.37 | [TK]D-Fender | He can't read |
| 18:07.58 | anonymouz666 | I know it's free. I had my copy, even with that, I bought from the site. |
| 18:11.08 | *** join/#asterisk wasanzy (~wasanzy@197.159.129.10) |
| 18:11.10 | wasanzy | hi |
| 18:11.34 | wasanzy | pls what is the different between asterisk version 1.8 and 11? |
| 18:11.42 | wasanzy | this is confusing me |
| 18:14.17 | [TK]D-Fender | wasanzy: upgrade.txt , changelog.txt |
| 18:14.19 | [TK]D-Fender | ^ |
| 18:15.29 | Tekz | so i've got a site with 3 phone users. Digium reseller wants to sell me a $3200 base system the AA80. Includes phones + recurring subscription fee. Anyone think that's overkill? |
| 18:15.52 | [TK]D-Fender | Tekz: My watch can handle that .. and it's ANALOG. |
| 18:16.10 | Tekz | I should rephrase... anyone *ELSE* think it's overkill? Because it seems like complete overkill to me. |
| 18:16.29 | Tekz | Yea I agree [TK]D-Fender |
| 18:16.45 | wasanzy | [TK]D-Fender: please I don't understand |
| 18:17.12 | wasanzy | the version 1.8 is found in repositories but 11 is not there |
| 18:17.15 | [TK]D-Fender | wasanzy: What part do you not understand? There are documents included with Asterisk that clearly list what's changed. Go read them |
| 18:18.31 | Tekz | Trying to convince my boss that I could take one of our old workstations, throw an FXO card into it and do what the $3200 dollar system could do for easily half the price. |
| 18:18.32 | wasanzy | [TK]D-Fender: I didn't download all the versions that is why am asking |
| 18:18.53 | [TK]D-Fender | waDownload 11. it has those files for previous versions |
| 18:18.58 | [TK]D-Fender | wasanzy: Download 11. it has those files for previous versions |
| 18:23.35 | wasanzy | anyway which one will you advise for an IVR system? |
| 18:24.16 | [TK]D-Fender | wasanzy: I see no reason to go to an older branch |
| 18:24.27 | [TK]D-Fender | wasanzy: 11 is LTS and will be supported longer. |
| 18:25.50 | anonymouz666 | 11 is the right way to go |
| 18:25.57 | *** join/#asterisk bipul (~bipul@unaffiliated/bipul) |
| 18:28.04 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 18:28.33 | *** join/#asterisk Vann (~manny@71-14-7-106.static.stbr.ga.charter.com) |
| 18:29.09 | wasanzy | ok I am facing problem with starting up the 11.5.0 I installed. when I start like this: service asterisk start and check status service asterisk status, I get this output always: |
| 18:30.21 | wasanzy | asterisk dead but subsys locked |
| 18:30.47 | wasanzy | I do run asterisk -vvvvvvvvvvvvvvvvc |
| 18:30.50 | gops | how to identify caller disconnection and agent disconnection? |
| 18:31.02 | [TK]D-Fender | gops: where/when? |
| 18:31.10 | gops | for incoming calls |
| 18:31.14 | wasanzy | and no tangible error |
| 18:31.28 | [TK]D-Fender | gops: No, Where, and when are you looking to have this knowledge? |
| 18:31.36 | [TK]D-Fender | wasanzy: start * manually |
| 18:32.18 | wasanzy | [TK]D-Fender: you mean asterisk -cvvvvvvv ? |
| 18:32.20 | gops | [TK]D-Fender: Basically am looking to have it in CDR or CEL when there is a incoming call and routed to queue... |
| 18:32.45 | [TK]D-Fender | gops: If it's a queue call then it's in the queue log |
| 18:33.22 | gops | [TK]D-Fender: in the queue log all i have is only COMPLETECALLER.. only for outgoing call I have COMPLETEAGENT |
| 18:33.50 | [TK]D-Fender | Outgoing is another matter... as queues are not "outgoing |
| 18:34.02 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.166) |
| 18:34.37 | [TK]D-Fender | queues will set COMPLETECALLER / COMPLETEAGENT based on who ended the call... |
| 18:35.04 | gops | [TK]D-Fender: actually am using Queuemetrics, and agents receive the call, they might put the break or pause and make outgoing call via softphone or hardphone |
| 18:35.25 | wasanzy | [TK]D-Fender: when installing from source, do you advise I do make install sample? I didn't do that |
| 18:35.40 | [TK]D-Fender | gops: that other call is always 2100% separate to the other call and has no way to tie them together |
| 18:35.44 | [TK]D-Fender | 100% |
| 18:35.45 | wasanzy | I only copied some needed confs |
| 18:36.00 | [TK]D-Fender | wasanzy: Is it complaining to you about missing configs? |
| 18:36.15 | Penguin | I install my samples into a samples directory. |
| 18:36.29 | wasanzy | yes something like that |
| 18:37.02 | *** join/#asterisk mitchrodrigues (~mitchrodr@c-98-244-30-255.hsd1.ca.comcast.net) |
| 18:38.00 | wasanzy | am reinstalling |
| 18:38.19 | Penguin | This isn't Windows. |
| 18:38.34 | wasanzy | Penguin: me? |
| 18:38.50 | Penguin | you |
| 18:39.04 | wasanzy | am using CentOS |
| 18:39.16 | wasanzy | am installing version 11.5.1 |
| 18:44.05 | gops | yesterday I faced a warning like this "chan_sip.c: Unable to cancel schedule ID 1714357. This is probably a bug" and all my SIP channels freezed....what could be the reason? |
| 18:44.20 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 18:44.39 | [TK]D-Fender | gops: What are you running? |
| 18:45.08 | gops | [TK]D-Fender: Asterisk 11.2.1 in Cent OS 5.8 64 bit with Sangoma transcoding card |
| 18:45.29 | [TK]D-Fender | gops: Upgrade to current and retest |
| 18:46.17 | gops | [TK]D-Fender: right now I can't upgrade since its a production machine.... and also this I faced only yesterday... last 10 days there is no issue and even today also its running |
| 18:46.31 | *** join/#asterisk sarobat (~saroth@24.244.29.121) |
| 18:46.36 | gops | [TK]D-Fender: just would like to know in what kind of scenario this might happen.... |
| 18:46.51 | drmessano | If I upgrade the asterisk and asterisk-dahdi packages (using the digium RPMs) and don't reboot to load the new kernel and kernel modules, should I assume some sort of instability? |
| 18:46.53 | gops | [TK]D-Fender: any cpu load issue... or session reinvite.. issue? |
| 18:47.07 | [TK]D-Fender | gops: Take your chances as long as you feel like like it and upgrade as soon as you find the downtime it causes to be worthwhile to you |
| 18:47.19 | gops | [TK]D-Fender: ok |
| 18:48.53 | wasanzy | [TK]D-Fender: please how do I start asterisk manually? |
| 18:48.59 | drmessano | asterisk |
| 18:49.03 | wasanzy | as you mentioned before |
| 18:49.24 | [TK]D-Fender | wasanzy: asterisk -gvvvvvvc |
| 18:49.34 | wasanzy | ok |
| 18:49.35 | [TK]D-Fender | wasanzy: You should spend some quality time with the book.... |
| 18:50.30 | *** part/#asterisk sarobat (~saroth@24.244.29.121) |
| 18:50.45 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.166) |
| 18:56.34 | drmessano | hmmm |
| 18:56.45 | *** join/#asterisk cmendes0101 (~cmendes01@office.phone.com) |
| 19:00.12 | wasanzy | Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
| 19:00.19 | wasanzy | asterisk is running |
| 19:00.38 | wasanzy | but asterisk -r is giving me that error, even when I run it as root |
| 19:02.02 | newtonr | wasanzy, what user is running Asterisk? |
| 19:06.03 | *** join/#asterisk cmendes0101 (~cmendes01@office.phone.com) |
| 19:10.40 | wasanzy | newtonr: root |
| 19:11.00 | wasanzy | I didn't change anything in the /etc/asterisk/asterisk.conf |
| 19:11.40 | newtonr | so, "ls -l /var/run/asterisk/" ? what is there, who owns it, what are the permissions? |
| 19:11.49 | [TK]D-Fender | [15:00]wasanzyasterisk is running <- show us |
| 19:12.34 | wasanzy | srwxr-xr-x. 1 root root 0 Oct 18 19:01 asterisk.ctl |
| 19:12.34 | wasanzy | -rw-r--r--. 1 root root 5 Oct 18 19:01 asterisk.pid |
| 19:14.29 | ChannelZ-Wk | are you suuuuure it's running? |
| 19:14.36 | newtonr | wasanzy, double-check "ps aux | grep -i asterisk" |
| 19:15.39 | wasanzy | <PROTECTED> |
| 19:16.21 | Penguin | ps -C asterisk <--------- |
| 19:16.31 | wasanzy | when I press Ctr+C and check again, it seem asterisk stopped running |
| 19:16.52 | Penguin | or: pgrep asterisk |
| 19:16.56 | Penguin | or: pidof asterisk |
| 19:17.00 | wasanzy | <PROTECTED> |
| 19:17.00 | wasanzy | <PROTECTED> |
| 19:17.00 | wasanzy | <PROTECTED> |
| 19:17.07 | [TK]D-Fender | [15:16]wasanzywhen I press Ctr+C and check again, it seem asterisk stopped running <- correct |
| 19:17.19 | [TK]D-Fender | wasanzy: I told you to start it manually... when means when you exit, it DIES |
| 19:17.38 | Penguin | You shouldn't have more than one asterisk process. |
| 19:18.00 | wasanzy | let me kill all and start again |
| 19:18.08 | Penguin | pkill asterisk |
| 19:19.43 | wasanzy | [TK]D-Fender: when I start manually, it means I have to leave the CLI like that? |
| 19:20.10 | [TK]D-Fender | wasanzy:Yes. This was for testing. |
| 19:20.23 | [TK]D-Fender | try starting it as a daemon after if it was SUCCESSFUL before |
| 19:20.35 | wasanzy | ok, now I can asterisk -r |
| 19:20.46 | wasanzy | let me start it as daemon and see |
| 19:21.39 | Penguin | For normal operation, start it with "service asterisk start" |
| 19:22.43 | wasanzy | Penguin: that is where the problem is,when I start it that way and check the status, I get: asterisk dead but subsys locked |
| 19:24.30 | wasanzy | hmm I just start it "service asterisk start" and ps -C asterisk shows nothing |
| 19:24.30 | Penguin | Did you correctly configure asterisk.conf, /etc/sysconfig/asterisk (I think this is the right path), and also set the correct ownership on your asterisk files and directories? |
| 19:25.03 | *** join/#asterisk cmendes0101 (~cmendes01@office.phone.com) |
| 19:25.28 | wasanzy | I created asterisk user and set it as the owner of all asterisk config files |
| 19:25.48 | wasanzy | mine is /etc/asterisk/asterisk.conf |
| 19:27.24 | *** join/#asterisk cmendes0101| (~cmendes01@office.phone.com) |
| 19:27.31 | wasanzy | ;runuser = asterisk and ;rungroup = asterisk is commented out in the config |
| 19:27.50 | wasanzy | so basically, asterisk is running as root |
| 19:28.53 | Penguin | Go ahead and uncomment those. |
| 19:29.08 | Penguin | I'm logging onto a CentOS box now to help guide you. |
| 19:29.26 | wasanzy | Penguin: thank you so much |
| 19:29.38 | Penguin | Do you have anything set it /etc/sysconfig/asterisk? |
| 19:30.45 | wasanzy | Penguin: ah yes, something is there |
| 19:31.04 | wasanzy | # Startup configuration for the Asterisk daemon |
| 19:31.33 | Penguin | Set the following two values: AST_USER=asterisk and AST_GROUP=asterisk |
| 19:32.03 | wasanzy | this is very strange, infact this is my third time of setting up asterisk and have not noticed this before till today. hmm |
| 19:32.04 | wasanzy | o |
| 19:32.05 | wasanzy | ok |
| 19:32.53 | Penguin | The init.d script will read this file. |
| 19:33.20 | wasanzy | ok hv done that, going ahead to start it and see |
| 19:34.00 | *** join/#asterisk bipul (~bipul@unaffiliated/bipul) |
| 19:34.46 | wasanzy | Penguin: the same problem: asterisk dead but subsys locked |
| 19:36.03 | Penguin | I guess it's time to start checking the things you said you checked/changed. |
| 19:36.16 | Penguin | getent passwd asterisk |
| 19:36.23 | Penguin | getent group asterisk |
| 19:36.32 | Penguin | ls -dl /var/run/asterisk |
| 19:37.15 | wasanzy | Penguin: as in set password for asterisk? |
| 19:37.41 | Penguin | As in: copy what I just typed, paste it into your terminal, press enter, report to me what it says. |
| 19:38.03 | wasanzy | drwxr-xr-x. 2 asteriskpbx asteriskpbx 4096 Oct 18 19:37 /var/run/asterisk |
| 19:38.20 | Penguin | Are you trying to use freepbx? |
| 19:38.32 | wasanzy | no |
| 19:38.52 | wasanzy | I added the user asteriskpbx not asterisk |
| 19:38.53 | Penguin | Did you create user "asterisk" or user "asteriskpbx"? |
| 19:39.05 | wasanzy | asteriskpbx:x:500:500::/home/asteriskpbx:/bin/bash |
| 19:39.24 | wasanzy | so all the changes I did, I used asteriskpbx not asterisk |
| 19:39.37 | Penguin | You don't need a shell, so change that. |
| 19:39.58 | Penguin | getent group asteriskpbx |
| 19:40.23 | wasanzy | asteriskpbx:x:500: |
| 19:40.26 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.166) |
| 19:40.28 | Penguin | In asterisk.conf, you set the user and group to asteriskpbx? |
| 19:40.41 | wasanzy | Penguin: yes |
| 19:40.47 | Penguin | In /etc/sysconfic/asterisk, you set the user and group to asteriskpbx? |
| 19:40.54 | wasanzy | yes |
| 19:41.07 | *** join/#asterisk bipul (~bipul@unaffiliated/bipul) |
| 19:46.38 | Penguin | Let's try to start asterisk as your asteriskpbx user. su -c asterisk asteriskpbx |
| 19:48.54 | Penguin | Oh, do you have selinux enabled? |
| 19:49.12 | Penguin | sestatus |
| 19:49.38 | wasanzy | I disabled it but let me check again |
| 19:50.30 | wasanzy | aaaahhhhhh very interesting, is enabled hmm |
| 19:51.24 | wasanzy | do I have reboot after disabling it? |
| 19:51.32 | Penguin | I don't think so. |
| 19:53.01 | Penguin | Remember to set selinux to disabled in /etc/sysconfig/selinux. |
| 19:55.31 | wasanzy | Penguin: am doing that now |
| 19:56.44 | wasanzy | SELINUX=disabled so I don't know why the command is showing this: |
| 19:56.59 | wasanzy | SELinux status: enabled |
| 19:56.59 | wasanzy | SELinuxfs mount: /selinux |
| 19:57.00 | wasanzy | Current mode: enforcing |
| 19:57.00 | wasanzy | Mode from config file: disabled |
| 19:57.00 | wasanzy | Policy version: 24 |
| 19:57.00 | wasanzy | Policy from config file: targeted |
| 19:57.46 | wasanzy | am rebooting the system and see |
| 20:01.46 | Penguin | Next time, use a pastebin. |
| 20:03.23 | wasanzy | Penguin: ok sorry |
| 20:04.42 | wasanzy | ah I think the problem as I didn't reboot the system. after booting, I see that asterisk is running and I can asterisk -r too |
| 20:06.07 | wasanzy | Penguin: thank you for the great help I appreciate it a lot. |
| 20:09.26 | Penguin | No prob. |
| 20:14.45 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 20:14.47 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 20:15.11 | *** join/#asterisk BrokenArrow (~BrokenArr@unaffiliated/brokenarrow) |
| 20:29.31 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 21:00.13 | _Corey_ | Can someone help me find a CALLERID() field from a some PRI debugging...? http://pastebin.com/zr6MbZgG |
| 21:00.37 | _Corey_ | I'm looking for the number that's on line 77: <38 30 34 37 37 35 38 30-30 35> - "8047758005" |
| 21:00.59 | _Corey_ | It's a forwarded number identifier so I expected to find it in CALLERID(rdnis) but it's not there |
| 21:02.27 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.221) |
| 21:13.02 | newtonr | _Corey_, I'm not a PRI guy (at least in depth), but I asked a PRI guy and he says the following: "libpri does not know how to interpret that facility ie. It only knows it is in ASN.1 format but does not know what the fields mean.", "Libpri knows very little about National ISDN specific ROSE messages.", this is also indicated in lines 104 and 106 |
| 21:14.47 | _Corey_ | Hmmm... Thanks Rusty. Not looking promising. |
| 21:15.09 | newtonr | if Libpri doesn't know know how to interpret it, it's pretty certain that you won't find it anywhere in Asterisk |
| 21:15.27 | _Corey_ | Doing an Avaya voicemail replacement. I'm trying to see if they can put the RDNIS in the right place now... |
| 21:37.00 | *** join/#asterisk Chotaire (chotaire@chotaire-home.vipri.net) |
| 21:37.38 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 21:42.32 | *** join/#asterisk tzafrir (~tzafrir@bzq-84-109-18-138.red.bezeqint.net) |
| 21:53.57 | *** join/#asterisk kresp0 (~kresp0@unaffiliated/kresp0) |
| 22:01.42 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:21.18 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:30.31 | *** join/#asterisk Chotaire (chotaire@chotaire-home.vipri.net) |
| 22:52.47 | *** join/#asterisk mmourani (~Adium@64.229.213.168) |
| 22:53.05 | mmourani | hi |
| 22:53.23 | *** join/#asterisk redotis (~redotis@nc-76-5-176-123.dhcp.embarqhsd.net) |
| 22:53.35 | redotis | What's going on in here? |
| 22:53.58 | redotis | Is D-Fender still being a big meany head? |
| 22:54.06 | WIMPy | The question what number goes where definitely needs some tidying up. Especially in the documentation, but the ANI part regularly produces unexpected results. |
| 22:54.15 | mmourani | lol |
| 22:54.41 | mmourani | i need to fix an issue of incoming calls on my asterisk box |
| 22:55.04 | mmourani | I need to compare all of the incoming calls during a 2 sec period between them |
| 22:55.32 | mmourani | what will be the best way to do this ? |
| 22:57.37 | WIMPy | An unlimited number of calls within 2s? |
| 23:01.20 | mmourani | yes |
| 23:01.33 | mmourani | large number of calls |
| 23:02.27 | WIMPy | Then I'd suggest writing an application that does it via AMI. |
| 23:02.30 | navaismo | sipp |
| 23:02.59 | mmourani | ok |
| 23:03.32 | mmourani | any other suggestion ? |
| 23:07.32 | [TK]D-Fender | mmourani: compare how? |
| 23:07.39 | redotis | enable a user/password for ami that just gets calls, telnet to the ami port, copy the data to a spreadsheet....compare |
| 23:07.39 | Nugget | telnet is eeeeeeevil! |
| 23:09.06 | *** join/#asterisk vinhdizzo (~vinh@cpe-98-154-220-20.socal.res.rr.com) |
| 23:11.52 | vinhdizzo | Hi. I used to have an auto-attendant set up on my google voice account via jabber/gtalk in asterisk 1.8 on Ubuntu. Trying to set the same up the same configuration on asterisk 11 on CentOS 6. I'm aware that instead of using jabber.conf and gtalk.conf, I now have to rely on xmpp.conf and motif.conf. However, although asterisk successfully goes online with my google voice account, the call attendant doesn't seem to work. Is there an |
| 23:14.32 | snadge | dammit.. my scrollback isnt large enough |
| 23:14.52 | snadge | i have 16GB of ram.. and x-chat defaults to like a 2 kilobyte buffer |
| 23:15.05 | snadge | fail :p |
| 23:15.30 | vinhdizzo | my extensions.conf starts with: |
| 23:15.30 | vinhdizzo | exten => s,1,Answer() |
| 23:15.30 | vinhdizzo | exten => s,n,Wait(1) |
| 23:15.30 | vinhdizzo | exten => s,n,SendDTMF(1) |
| 23:15.30 | vinhdizzo | exten => s,n,Background(/home/vinh/TNTTSP-Phone/welcome) |
| 23:15.31 | vinhdizzo | exten => s,n,Background(/home/vinh/TNTTSP-Phone/current) |
| 23:15.31 | vinhdizzo | exten => s,n,Goto(0,1) |
| 23:15.44 | ChannelZ | ~pb |
| 23:15.45 | infobot | 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/; or install pastebinit with yum or aptitude. |
| 23:15.58 | WIMPy | KB? I'm used to specify scrollback in lines. |
| 23:16.02 | snadge | its because of people like vinhdizzo ;) |
| 23:16.08 | snadge | yeah it says 500 |
| 23:16.13 | snadge | how do i make it unlimited? |
| 23:16.16 | ChannelZ | vinhdizzo: does the call even come into your dialplan? |
| 23:16.20 | WIMPy | There's auto-ignore for that. |
| 23:16.21 | redotis | you gotta have an exten for your number right |
| 23:16.43 | navaismo | yea exten s in that case |
| 23:16.51 | ChannelZ | snadge: turn on logging.. it will restore the history when you restart even. |
| 23:16.52 | snadge | any of you guys do any l7 routing for asterisk? |
| 23:17.17 | snadge | im just going to keep asking this question until one day.. somebody says yes ;) |
| 23:17.44 | vinhdizzo | redotis: navaismo: care to elaborate? what i had worked in 1.8, but doesn't appear to work in 11 |
| 23:17.58 | WIMPy | L7 routing as in shouting acrss the room: Tom, a call for you on line 2? |
| 23:18.14 | navaismo | vinhdizzo: to me your dianplan is ok, now you need to show us the cli output |
| 23:18.24 | snadge | WIMPy, that sounds about right.. but no.. more like L7 routing as in, a load balancing solution in front of a bunch of asterisk servers |
| 23:18.32 | redotis | exten => 14564573545,1,Goto(s,1) |
| 23:18.41 | redotis | something like that |
| 23:18.57 | redotis | I'm a newb though |
| 23:18.58 | redotis | so |
| 23:19.07 | redotis | don't trust what I say |
| 23:19.11 | vinhdizzo | navaismo: asterisk -rvvvvv? i didn't see anything log when i dial my google voice number |
| 23:19.25 | snadge | RNAT and L7 .. it requires a bit of knowledge of SIP internals |
| 23:19.35 | vinhdizzo | navaismo: or is there a different way to look at the log? |
| 23:19.38 | ChannelZ | then you have Some Other Issue probably.. with your xmpp/motif config |
| 23:19.48 | snadge | you can allegedly route packets via the unique callid field in the sip header |
| 23:20.15 | navaismo | vinhdizzo: pastebin your motif & xmpp conf and the complete dialplan for the context involved |
| 23:20.18 | [TK]D-Fender | vinhdizzo: If you see nothing then the call wouldn't appear to be making it into the dialplan at all |
| 23:20.18 | WIMPy | snadge: LVS has a persistance option. |
| 23:21.13 | navaismo | exactly ^^ |
| 23:21.33 | snadge | right, LVS uses that concept.. but is anyone actually using LVS to do that.. or anything else for that matter |
| 23:21.39 | vinhdizzo | one sec |
| 23:21.41 | snadge | such as F5, barracuda, netscaler, A10, etc |
| 23:21.57 | snadge | all the latter solutions cost big dollars |
| 23:22.12 | snadge | LVS is free.. but also its quality probably reflects that ;) |
| 23:23.27 | WIMPy | I haven't used it with sip, but otherwise it worked perfectely 13 years ago. |
| 23:24.35 | snadge | thats part of the problem though.. load balancing is a complicated subject in itself.. for http, https, you'd have to be a retard not to understand it |
| 23:24.45 | snadge | thats about as easy as it gets |
| 23:24.52 | redotis | Hey vinhdizzo did you redirect your google voice to your primary number? |
| 23:24.54 | redotis | in gmail? |
| 23:24.59 | snadge | then you have UDP.. a little bit more complicated.. then you have SIP on top of UDP, which just makes your brain explode |
| 23:25.37 | WIMPy | I don't see where it's more complicated if you enable persistence. |
| 23:25.41 | redotis | I think that's the way you tell google voice numbers to go to asterisk |
| 23:25.45 | snadge | and the official documentation for LVS says.. "sip is not supported" |
| 23:25.55 | snadge | it is out of date |
| 23:26.00 | snadge | http://lwn.net/Articles/399571/ |
| 23:26.07 | redotis | maybe not |
| 23:26.14 | snadge | one guy.. "simon horman" .. has written the persistence engine for sip |
| 23:26.22 | redotis | or set the context in motif.conf |
| 23:26.24 | snadge | but i've found zero evidence of anyone actually successfully using it |
| 23:27.06 | snadge | and thats the thing.. if some commercial entity is actually using that code to scale/balance asterisk servers |
| 23:27.15 | snadge | they're probably not going to write a blog and post on forums about it |
| 23:27.56 | snadge | im just kinda trying to get a feel for.. is it worth going down that path, and trying out simon's code.. or just paying money for citrix netscaler |
| 23:28.05 | vinhdizzo | navaismo: [TK]D-Fender: hmm, just deleted my conf files and re-entered them, think i'm getting somewhere now |
| 23:28.05 | WIMPy | I don't see why SIP would need special support. |
| 23:28.20 | WIMPy | But you will have to enable session-timers. |
| 23:28.37 | snadge | 1 because its udp.. 2 because if you want to do funky things with load balancing, you need to get inside the sip packet itself |
| 23:28.42 | vinhdizzo | [Oct 18 23:27:11] WARNING[2200][C-00000000]: file.c:701 ast_openstream_full: File /home/vinh/TNTTSP-Phone/current does not exist in any format |
| 23:29.00 | vinhdizzo | i have a wav file. am i missing something that it isn't playing it? |
| 23:29.08 | vinhdizzo | oh wait |
| 23:29.14 | vinhdizzo | [Oct 18 23:27:11] WARNING[2200][C-00000000]: file.c:1017 ast_streamfile: Unable to open /home/vinh/TNTTSP-Phone/current (format (ulaw)): Permission denied |
| 23:29.17 | Kobaz | is it readable by asterisk |
| 23:29.22 | Kobaz | there you go |
| 23:29.33 | vinhdizzo | do the files have to be owned by asterisk? |
| 23:29.37 | vinhdizzo | it's world-readable on the system |
| 23:29.37 | Kobaz | no |
| 23:29.42 | Kobaz | just readable |
| 23:29.56 | Kobaz | parent dirs must be +x |
| 23:30.25 | vinhdizzo | ok let me check |
| 23:30.31 | Kobaz | so if the owner isn't asterisk, and the group isn't asterisk. then you need o+x |
| 23:31.30 | vinhdizzo | drwxr-xr-x 5 vinh vinh 4.0K Jan 19 2013 TNTTSP-Phone |
| 23:31.40 | Kobaz | and /home/vinh |
| 23:31.43 | Kobaz | and /home |
| 23:32.26 | Kobaz | and / too, if you have a funky setup |
| 23:32.56 | [TK]D-Fender | vinhdizzo: I highly recommend you stop trying to put your files in places that Asterisk has no reason to have to be looking for them in. Put the un the base of where it already stores sounds with asterisk actually being the owner |
| 23:33.09 | [TK]D-Fender | under* |
| 23:33.18 | Kobaz | that too |
| 23:33.22 | vinhdizzo | [TK]D-Fender: where is a good place to store the files? |
| 23:33.23 | Kobaz | ^^^ what he said |
| 23:33.28 | Kobaz | the default directory? |
| 23:33.36 | [TK]D-Fender | vinhdizzo: Where Asterisk already looks by deffault |
| 23:33.38 | Kobaz | /var/lib/asterisk/sounds |
| 23:33.46 | Kobaz | i use /vae/lib/asterisk/sounds/custom |
| 23:33.49 | [TK]D-Fender | vind"sore show settings" <- |
| 23:33.51 | Kobaz | var |
| 23:34.14 | [TK]D-Fender | vinhdizzo: "core show settings" <- |
| 23:34.17 | [TK]D-Fender | gah |
| 23:35.11 | Kobaz | friday coding spree starts.....now! |
| 23:35.29 | vinhdizzo | ok |
| 23:35.30 | vinhdizzo | thanks |
| 23:36.22 | vinhdizzo | Kobaz: Can I create a custom directory in /var/lib/asterisk/sounds/? Say, ../my_sound/? |
| 23:36.29 | vinhdizzo | would it know to look for the sound in there? |
| 23:36.31 | Kobaz | sure |
| 23:36.34 | Kobaz | [10 18 19:33] <Kobaz> i use /vae/lib/asterisk/sounds/custom |
| 23:36.44 | vinhdizzo | oh oops |
| 23:36.46 | vinhdizzo | thanks |
| 23:36.46 | Kobaz | var rather |
| 23:36.49 | [TK]D-Fender | vinhdizzo: It looks relative when you don't start with a "/" |
| 23:36.54 | Kobaz | yeah |
| 23:37.00 | Kobaz | Playback(custom/mysound) |
| 23:37.11 | vinhdizzo | cool |
| 23:37.13 | vinhdizzo | thanks so much! |
| 23:37.30 | snadge | http://horms.org/gallery/me/a/4_6543_s.shtml#nav .. would you trust this guy to load balance your asterisk servers? ;) |
| 23:37.55 | snadge | probably.. that white suite does look spiffy |
| 23:37.56 | Kobaz | sure |
| 23:38.29 | snadge | but he has worked on about 50 projects since he did the persistence engine for sip |
| 23:38.32 | Kobaz | person in white tuxedo is always trusted to balance the load |
| 23:38.46 | snadge | http://horms.org/gallery/me/a/5_1280_s.shtml#nav |
| 23:38.52 | snadge | same man.. with thuggish beanie on |
| 23:39.03 | snadge | much less trustworthy in that picture :p |
| 23:43.16 | snadge | maybe i'll shoot him an email.. wont hurt |
| 23:43.44 | snadge | he might have an idea of the state of his code.. whether people use it or not |
| 23:50.23 | vinhdizzo | Hi again. I used to have an option in the dialplan to transfer the call to an outside number. it used to ring then an answering machine picks up. now, i dont hear the ring (although it still does because i get to the answering machine). is there a way to get a ring tone to play? here is message in the CLI: [Oct 18 23:48:45] NOTICE[2707][C-00000001]: chan_motif.c:1636 jingle_indicate: Don't know how to indicate condition '15' |
| 23:51.23 | iq | Hi |
| 23:59.24 | vinhdizzo | let me know if you have sugg |
| 23:59.33 | [TK]D-Fender | vinhdizzo: Show us the actual call. We haven't gotten to see what is happening yet |
| 23:59.50 | vinhdizzo | one sec |