| 00:12.08 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 00:14.15 | *** join/#asterisk alexoc (~IceChat77@187.64.254.47) |
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| 00:39.28 | *** part/#asterisk jerware (~root@www.kaylon.com) |
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| 01:08.56 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
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| 01:51.14 | saxa | [TK]D-Fender: ; nat = comedia ; Use rport if the remote side says to use it and perform comedia RTP handling. |
| 01:51.36 | saxa | I think you were talking about this, correct ? |
| 01:56.29 | [TK]D-Fender | saxa: I'm not solid on what the different new settings were as I'm not running 1.8 right now... |
| 01:57.10 | *** join/#asterisk rue_mohr (~rue@h24-207-19-104.cst.dccnet.com) |
| 01:58.18 | rue_mohr | there is a fault between the T1 channelbank I have and asterisk, the asterisk state machine dosn't recognize the 'hang up' signal from the channelbank, I have a line stuck again, how can I , via the asterisk prompt, for the call tieing up the line to be hung up |
| 02:01.05 | saxa | [TK]D-Fender: thats probably what i need to change nat=yes setting. |
| 02:01.10 | saxa | will try it now |
| 02:02.26 | [TK]D-Fender | rue_mohr: "soft hangup [channel]" |
| 02:02.40 | rue_mohr | idea on hwo to get the channel number? |
| 02:03.10 | rue_mohr | oh I only have one co channel :) |
| 02:03.47 | rue_mohr | phony2*CLI> soft hangup dahdi/7 dahdi/7 is not a known channel |
| 02:04.01 | rue_mohr | what did I forget? |
| 02:04.47 | [TK]D-Fender | to look at the channel list. |
| 02:05.07 | [TK]D-Fender | dahdi/7 is a devicename, not a channel name |
| 02:05.18 | rue_mohr | whats the channel name? |
| 02:05.47 | rue_mohr | http://ideone.com/uHhJS |
| 02:06.09 | rue_mohr | just 1 thru 8? |
| 02:06.17 | [TK]D-Fender | "show channels concise" |
| 02:06.24 | [TK]D-Fender | ASTERISK channels, not device specific |
| 02:07.18 | rue_mohr | No such command 'show channels concise' |
| 02:07.59 | [TK]D-Fender | show channels |
| 02:10.13 | *** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk) |
| 02:10.17 | rue_mohr | No such command 'show channels' |
| 02:10.53 | rue_mohr | checks the version of his [TK]D-Fender |
| 02:11.03 | rue_mohr | 1.6.0.22 |
| 02:11.15 | rue_mohr | ? |
| 02:11.22 | [TK]D-Fender | I was betting on you remaining completely decrepit given your history ;) |
| 02:11.37 | [TK]D-Fender | "core show channels concise" |
| 02:12.01 | rue_mohr | nothing, ..... hah, its hung up now... |
| 02:12.41 | rue_mohr | ok, now I know tho, and I set a max message length of 300 seconds, no more 2.5 hour messages while the channelbank waits for asterisk to figure it all out |
| 02:13.21 | rue_mohr | I'd love to know how to fix the state machine for the t1 signalling |
| 02:14.14 | [TK]D-Fender | What are you using on that, wink-start? |
| 02:14.33 | rue_mohr | no, its polarity reverse |
| 02:14.55 | [TK]D-Fender | You are a glutton for punishment.... |
| 02:15.01 | rue_mohr | all I know is asterisk often says "unknown state" from the t1 channelbank |
| 02:15.12 | carrar | change it to a PRI |
| 02:15.14 | rue_mohr | I dont know how to work out what the state is, or how to code it in |
| 02:15.23 | rue_mohr | carrar, for my 1 line house!? |
| 02:15.30 | rue_mohr | thats $1300/mo here |
| 02:15.31 | carrar | YES!! :) |
| 02:15.45 | carrar | ISDN? |
| 02:15.57 | rue_mohr | isdn not available, our co's are too old |
| 02:16.06 | [TK]D-Fender | rue_mohr: You're using a channel-bank and T1 card .... for 1 analog line. |
| 02:16.06 | carrar | go SIP |
| 02:16.18 | rue_mohr | [TK]D-Fender, AND 7 PHONES! |
| 02:16.25 | [TK]D-Fender | You'd be better off with a X100P cheap-ass POS for $15 |
| 02:16.27 | carrar | heh |
| 02:16.35 | rue_mohr | carrar, you want me to tell you about our jittery droppy isp? |
| 02:16.48 | carrar | get a different isp |
| 02:16.58 | rue_mohr | [TK]D-Fender, no I'm not, the channelbank does local bridging, so I dont get any echo or nuthin |
| 02:17.03 | carrar | get some qos |
| 02:17.06 | rue_mohr | carrar, none available |
| 02:17.16 | rue_mohr | not for less than $500/mo |
| 02:17.16 | carrar | oh well |
| 02:17.24 | carrar | like TK says, X100p |
| 02:17.29 | rue_mohr | no |
| 02:17.33 | carrar | or a 400 |
| 02:17.38 | rue_mohr | x100p would be echo hell |
| 02:17.49 | rue_mohr | echo hell, I'v worked with them before |
| 02:18.17 | [TK]D-Fender | rue_mohr: It's you..... technology hates you. Time to take up basket-weaving... |
| 02:18.34 | carrar | Get a 400 series card |
| 02:18.44 | rue_mohr | for a 7 phone house? |
| 02:19.07 | carrar | yes |
| 02:19.09 | rue_mohr | analog bridging rocks, even if its within a t1 channelbank |
| 02:19.19 | carrar | get the echo cancellation modules with it |
| 02:19.27 | carrar | since you think you have bad echo |
| 02:19.28 | rue_mohr | now your talking $800 |
| 02:19.37 | carrar | ok, then keep bitching |
| 02:19.57 | rue_mohr | to fix the fact that * has a fault in its t1 state table that *sometimes* causes me to get 3.5 hr voicemails |
| 02:20.13 | [TK]D-Fender | rue_mohr: I'd sooner blame your channel bank. |
| 02:20.23 | [TK]D-Fender | rue_mohr: then again... what T1 card do you have that on? |
| 02:20.31 | rue_mohr | its a newbridge ... |
| 02:20.31 | [TK]D-Fender | rue_mohr: And you know all of 1.6 is EOL... right? |
| 02:21.06 | rue_mohr | newbridge mainstreet 3624 |
| 02:21.30 | rue_mohr | if I installed * on the pbx yesterday whatever version it was would be obsolete today... |
| 02:22.02 | rue_mohr | I hold off so I'm not having to *continiously* learn new commands, I get to learn a lot of stuff all at once |
| 02:22.39 | [TK]D-Fender | rue_mohr: You're 4 branches behind now. |
| 02:22.41 | rue_mohr | if you look up the mainstreet 3624 on voipinfo, I think you will find me and kb1 wrote the book |
| 02:23.01 | rue_mohr | ok, less proper support, whats the new version get me? |
| 02:23.09 | rue_mohr | :) |
| 02:23.29 | rue_mohr | I need to reinstall anyhow, the computer its running on has now killed.... |
| 02:24.11 | [TK]D-Fender | rue_mohr: Your branch was up to 1.6.0.28 which is 6 behind even what you should have minimum. |
| 02:24.11 | rue_mohr | 6 hard drives, and with all the times I'v recovered and transfered the data, a few things are ...odd, it duplicates every number it plays |
| 02:24.18 | carrar | why not get a Adtran and convert your T1 to a PRI T1? |
| 02:24.29 | coppice | oooh, newbridge networks. a name from the dim distant past |
| 02:24.32 | rue_mohr | message 1 1 message 2 2 message 3 3 |
| 02:24.42 | [TK]D-Fender | rue_mohr: ChangeLog-1.6.0.2202-Feb-2010 16:022.6M |
| 02:24.55 | [TK]D-Fender | rue_mohr: Congratulations, exactly 2 years old. Happy Anniversary! |
| 02:25.06 | rue_mohr | :) see, its an appliance! |
| 02:25.17 | [TK]D-Fender | I SMELL BURNT TOAST |
| 02:25.48 | rue_mohr | I have a box of voippack VOG4000's |
| 02:26.05 | rue_mohr | but again, latency delay and echo |
| 02:26.15 | carrar | move to a city |
| 02:26.19 | carrar | get out of BFE |
| 02:26.37 | rue_mohr | I dont know what ninny came up with voip packat timing but they knew nothing about telco audio |
| 02:26.59 | rue_mohr | carrar, chill chill, Its only intermittently a problem |
| 02:27.21 | rue_mohr | I want to fix it, I'm a coder, I just dont know * code enough to be able to do it |
| 02:27.33 | rue_mohr | I know there is a state machine on the D channel of the T1 |
| 02:27.47 | rue_mohr | I know its getting a sequence it cant interpert |
| 02:27.54 | rue_mohr | I dont knwo where the state machine code is |
| 02:28.11 | carrar | D channel of the T1 |
| 02:28.14 | carrar | So then it's a PRI? |
| 02:28.23 | rue_mohr | I dont know what sequence its getting... I also apparently forgot to stop pressing return every 8 letters |
| 02:28.25 | [TK]D-Fender | rue_mohr: You're speaking PRI to your CB? Thought you said "polarity reversal" which would indicate RBS CAS |
| 02:29.02 | rue_mohr | pots co<---> channelbank<----t1--->asterisk with t1 card |
| 02:29.23 | rue_mohr | channelbank<---> pots phones |
| 02:30.26 | rue_mohr | its true, the unsolicoted callers cant dial, so they hit a menu that times out and hangs up, they are blocked |
| 02:30.41 | rue_mohr | in 7 years I'v recieved 1 call |
| 02:31.33 | rue_mohr | the cabevision/isp provider, he was really exited when I asked for voip service and everything, till he found out he couldn't sell it to me here |
| 02:32.41 | rue_mohr | so there, its just a t1 between the channelbank and *, everything else is analog |
| 02:33.02 | rue_mohr | tho, I'd really like to try one of the isdn cards in the channelbank and see if I can get a nortel digital set working |
| 02:33.31 | rue_mohr | I think kb1 got some nortel phones going with *, but iirc they were ip phones |
| 02:34.39 | rue_mohr | I was going to switch the channelbank out for an mics with a t1 card, but its been sitting too long without power and its key expired |
| 02:35.59 | rue_mohr | I thought the station cards might use an internel t1 bus, but upon prying it all open, all the chips are proprietory and I cant work out anything |
| 02:36.31 | rue_mohr | if I could only find a nortel engineer that designed the stuff |
| 02:39.19 | rue_mohr | I have some old toshiba geaer I think might be easier to hack |
| 02:40.50 | rue_mohr | ok thanks, gnight |
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| 06:50.06 | meest | Can anyone tell me what the purpose of the swritechunk fild in the dahdi channel struct is for? |
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| 07:01.18 | Mawkee | Has any insomniac asterisk guru seen this error before: Ring/Off-hook in strange state 6 on channel 1? |
| 07:01.29 | Mawkee | Can't find too much useful info on this on google |
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| 08:51.50 | ajithp | Hi all |
| 08:51.51 | ajithp | how to divert call in asterisk while busy in line |
| 08:52.27 | kaldemar | define "line" |
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| 08:54.00 | ajithp | Line in the sense am calling a number '5001' and '5001' busy in answering another call. |
| 09:02.05 | ajithp | Can i divert a call while busy in asterisk ?? |
| 09:04.18 | kaldemar | 5001 is just a number. it is crucial to know what is behind it. otherwise the answer is a general "check its status and function accordingly in your dialplan". |
| 09:05.26 | kaldemar | are you talking about a phone? if so, what kind of a phone? analog, SIP, IAX2, something else? a line as in a voip connection to a provider? POTS? ISDN? |
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| 09:22.21 | ajithp | kaldemar: I am using an IP phone. Am calling to this IP phone from another IP phone |
| 09:23.08 | ajithp | Its is a SIP connection |
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| 09:25.23 | ajithp | While calling an Sip connection if it is bust talking to another IP phone using SIP, can i divert the call to another SIP connection through Asterisk configuration |
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| 09:34.16 | kaldemar | ajithp: use DEVICE_STATE function to check its status before dialing. if it is busy, do your divert. "core show function DEVICE_STATE" in CLI will show examples. |
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| 09:36.13 | ajithp | Thak you. i will try that |
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| 10:20.43 | *** join/#asterisk Mandla (~Mandla@168.167.180.161) |
| 10:21.07 | Mandla | Hello. |
| 10:21.46 | Mandla | Is there a way on configuring the use of individual pin code for using the phone in Asterisk? |
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| 10:25.23 | kaldemar | Mandla: sure, for example with the Authenticate application in dialplan. |
| 10:26.17 | Mandla | Yah, but all users should enter a certain pin code for them to call. |
| 10:26.32 | kaldemar | or a custom prompt with Playback and reading the pin with Read. |
| 10:26.35 | kaldemar | and? |
| 10:27.43 | kaldemar | phones or users having their own pin codes is no problem. just a matter of deciding how to handle them. |
| 10:27.53 | Mandla | kaldemar, i want to know who used the phone when, where. |
| 10:29.59 | Mandla | kaldemar, i want to keep track of phone usage by individuals. |
| 10:30.28 | kaldemar | no problem with that. |
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| 10:30.47 | Mandla | How do i do that? |
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| 10:31.04 | kaldemar | after checking the PIN, you could for example put it to CDR(accountcode) and use CDR fot tracking or do something custom. |
| 10:31.45 | Mandla | kaldemar, any written article on the web that i can follow? |
| 10:32.56 | kaldemar | not that i know of. google away and maybe you'll find something. |
| 10:34.12 | Mandla | kaldemar, thanx |
| 10:35.10 | kaldemar | http://ofps.oreilly.com/titles/9781449303822/DialplanFundamentals.html#AuthenticatingCallers |
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| 11:40.41 | cjk | hi, i updated from 1.4 to 1.8. now some sip clients can not register when they send the packets to my fqdn (sipregister.mydomain.com) but when they enter the IP Address, then it works. Any idea? |
| 11:41.59 | bitwize | cjk: Any warnings or errors in your output? |
| 11:42.44 | cjk | bitwize, nothing in the logs |
| 11:42.54 | cjk | asterisk sends a 401 Unauthorized |
| 11:42.55 | cjk | thats all |
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| 11:45.30 | dym | Hello there! (: |
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| 11:46.05 | bitwize | cjk: ok, have you tried the fromdomain attribute in configuration? |
| 11:46.28 | cjk | bitwize, yes, but cant it be linked to the realm |
| 11:46.30 | dym | Im having a problem with pattern matching of phone numbers of different lenghts. I want to match 6 and 7 digit numbers with this extension: exten => _12345[5-9][0-9],1,Goto(test,1,1) for some reason this wont work, only if the number is dialled faster |
| 11:47.58 | cjk | is there a way to ignore the realm that the client sends? |
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| 11:50.16 | kaldemar | cjk: that pattern will not match 6 digits. it only matches 7. |
| 11:50.48 | kaldemar | cjk: sorry, not you. |
| 11:50.54 | kaldemar | dym: see above |
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| 11:54.10 | cjk | is there a way do see why asterisk rejects and incoming sip registration? |
| 11:54.12 | cjk | some debug level? |
| 11:54.57 | kaldemar | enable verbosity and sip debug |
| 11:55.45 | dym | kaldemar: sorry - here is the current setting: http://pastebin.com/vPNEHCA1 it still doesnt do "match as you go" and strikes on the 6 digit numbers with "not found". |
| 11:57.14 | kaldemar | dym: can you show a CLI output of such a call? |
| 11:57.22 | dym | ofcourse. |
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| 11:59.56 | dym | kaldemar: -- Span 1: Extension 123456@dahdi-in does not exist. Rejecting call from 'x'. |
| 12:00.10 | dym | this wwad dialled with an additional 7th digit but fell through |
| 12:00.13 | dym | was |
| 12:00.36 | dym | this falsifying is annoying :D let me repost with proper numbers. |
| 12:02.02 | kaldemar | use pastebin |
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| 12:05.22 | dym | kaldemar: http://stuff.patrick-geschke.de/ast1.txt |
| 12:05.40 | dym | lol |
| 12:05.43 | dym | i see one error :D |
| 12:05.44 | dym | sec |
| 12:06.48 | dym | forget everything :D |
| 12:06.52 | dym | It was layer 8 based :) |
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| 12:46.26 | jofry | hi folks |
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| 13:18.06 | Srini | Hi Room |
| 13:18.46 | Srini | While configuring digium card, are there only two files involved? system.conf and chan_dahdi.conf? |
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| 13:25.22 | kaldemar | Srini: if you use the init script that comes with dahdi-tools, the module for your card needs to be listed in /etc/dahdi/modules. |
| 13:26.11 | Srini | kaldemar, Yes the module is visible |
| 13:26.38 | Srini | kaldemar, wct4xxp |
| 13:27.23 | Srini | kaldemar, dahdi_hardware also is showing the wct4xxp |
| 13:28.17 | Srini | But the alarm is not turning green - so trying to understand what could have configured wrongly |
| 13:29.13 | kaldemar | Srini: which alarm? |
| 13:30.34 | Srini | I mean the light on the card next to the pri port is showing up red light. The dahdi_tool is showing BLU/YEL/RED |
| 13:31.16 | kaldemar | constant red? |
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| 13:49.43 | Srini | kaldemar, it is blinking not constant |
| 13:50.03 | Srini | on the pri modem side the light status is green |
| 13:51.43 | kaldemar | if its blinking, then the driver module for the card is not loaded. |
| 13:52.17 | Srini | kaldemar, Thanks for that lead! I will check the driver modules again....! |
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| 13:58.41 | eZz | hi |
| 13:59.33 | eZz | is it ok if a remote leg sends a multiple 183 with a different contact field path ? |
| 13:59.52 | eZz | meaning sdp's |
| 14:04.33 | Srini | lspci is showing wct4xxp - does that mean the module is loaded? |
| 14:05.02 | Srini | I mean lsmod |
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| 14:07.45 | kaldemar | Srini: yes. the mode or config might be wrong however. do you have a T1 or an E1? |
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| 14:10.14 | [TK]D-Fender | kaldemar, dahdi_hardware also is showing the wct4xxp |
| 14:10.37 | [TK]D-Fender | oops. |
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| 14:37.23 | *** join/#asterisk AaronCharpentier (~IceChat77@pool-71-162-114-14.bstnma.fios.verizon.net) |
| 14:38.36 | AaronCharpentier | Hey guys, we currently use CBeyond for our VOIP handler, and it runs through an asterisk box that we have local. The connection is a T1 line (1.5) but it's too slow for the amount of employees we have. We have a 35 up 35 down fios line, is there any way to connect the pbx/asterisk box up through this? Or does it have to be through a channeled T1/T3, etc. |
| 14:40.06 | [TK]D-Fender | AaronCharpentier, VoIP is IP. it goes over whatever IP routing tech your server can talk to |
| 14:40.37 | [TK]D-Fender | AaronCharpentier, CB sells you the upstream link so they can gurantee the quality (QoS end to end), and gouge you on the price |
| 14:40.53 | AaronCharpentier | Thought so, the server is connected through both the T1 and the Fios line, but configured to run through the wildcard T1 handler. |
| 14:41.06 | AaronCharpentier | Yeah, we can't even cancel it but we just want to stop using the T1, it really sucks. |
| 14:41.43 | AaronCharpentier | Could we in theory just remove the T1 Card, and force the machine to route through the fios connection? The updates etc all run through fios when sshing in, never had to work with an asterisk box before. |
| 14:42.22 | [TK]D-Fender | AaronCharpentier, make sure your server's default route points to the proper gateway to go out your FIOS, make sure your NAT setting (if applicable) are appropriate to the new link, forwarding, etc and go for it |
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| 14:42.48 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:43.05 | [TK]D-Fender | AaronCharpentier, You saying you installed a T1 card direct in your server to talk to CB? |
| 14:43.25 | [TK]D-Fender | AaronCharpentier, I've heard of them using a router on your side and SIP from there to your box, but that all |
| 14:43.31 | AaronCharpentier | Awesome thanks a ton, one quick question - until now I've only had to deal with extensions.conf voicemail.conf and users.conf, I'm very familiar with linux - my question is where in asterisk can I define the server routing. |
| 14:43.56 | AaronCharpentier | Wildcard TE121, Card 1 - Port undefined (span_1) OKESF/B8ZS23/24PRI - CPE |
| 14:44.02 | [TK]D-Fender | routing is OS level, not * level |
| 14:44.05 | AaronCharpentier | That's the card they installed in our server. |
| 14:44.22 | [TK]D-Fender | AaronCharpentier, OH, you had a straight PRI with CB? well.. then that's jsut a regular PSTN link |
| 14:44.45 | [TK]D-Fender | AaronCharpentier, And I guess that's out the window. |
| 14:44.54 | AaronCharpentier | So how does that change it for me being able to go into Fios? Impossible now? |
| 14:45.11 | [TK]D-Fender | AaronCharpentier, You can leave it in as-is, handle calls that still come in however you want, and setup the new stuff on the side. |
| 14:45.31 | [TK]D-Fender | AaronCharpentier, Changes nothing at all... makes it easier. Means you didn't have 2 IP interfaces on your server |
| 14:45.41 | [TK]D-Fender | (that we would have guessed at) |
| 14:45.53 | [TK]D-Fender | Nor outing to change then if it already goes out FIOS |
| 14:45.59 | [TK]D-Fender | ~sipnat |
| 14:46.00 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, canreinvite, externhost or externip, and localnet. |
| 14:46.01 | [TK]D-Fender | ^^^ |
| 14:46.12 | [TK]D-Fender | Guide if your server is indeed on a private IP behind NAT |
| 14:46.36 | AaronCharpentier | That's the thing the traffic is going through the T1, but the server is *also* connected to fios through it's regular lan port, the T1 line goes in at the wildcard |
| 14:47.09 | AaronCharpentier | I'll read that guide and see if it helps out with my confusion, I feel bad pestering you guys. |
| 14:47.48 | *** join/#asterisk Praise- (~Fat@unaffiliated/praise) |
| 14:48.41 | [TK]D-Fender | AaronCharpentier, No, your T1 is "voice", not "data". Don't think of it as "routing" or "network" at all. Your server sounds to have just a single NIC and networking-wise is going through your FIOS like any other PC there |
| 14:50.54 | AaronCharpentier | I see, I guess what I was asking is if we can route the voice through the fios, since we're constantly encountering issues with the voice (choppy, dropped calls) |
| 14:51.44 | AaronCharpentier | Bah, I can't get out of the whole "Data is data" even if it's voice mindset. |
| 14:52.32 | AaronCharpentier | CBeyond tells us the T1 looks fine, when they do their little remote check. But the quality of service for the system has been nothing but crap. |
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| 14:59.17 | [TK]D-Fender | AaronCharpentier, Yeah, forget QoS, that's a network thing.. you're straight PRI. So you can have BOTH on your server at the same time, no issue |
| 15:00.21 | AaronCharpentier | Thanks, any ideas as to where we should look as far as call quality is concerned then? We're using grandstream GXP280's...they're honestly kind of crappy. |
| 15:01.10 | [TK]D-Fender | AaronCharpentier, Phones are on the local LAN right? |
| 15:01.17 | AaronCharpentier | Yeah |
| 15:01.35 | [TK]D-Fender | AaronCharpentier, Unless your LAN has some serious traffic, then it's just between your server and CB |
| 15:01.36 | AaronCharpentier | Configured within to connect to the SIP server (the asterisk box) which also has an ip on our local lan |
| 15:02.51 | AaronCharpentier | How much traffic are we talking before we call it serious? We have about 25 phones connected, 30 users on the internet, a zimbra server and an ftp server. |
| 15:03.41 | [TK]D-Fender | AaronCharpentier, basically LAN flooding. SIP phones take up nothing so unless you've got them inline with PC sucking off a file server something fierce the odds are low that it's between your server & phones. So that means the link to CB is the flakey end |
| 15:04.09 | [TK]D-Fender | AaronCharpentier, Especially if everyone is getting hit |
| 15:04.47 | AaronCharpentier | Some days are better than others, some people have consistently bad calls - others do not. I'll call CB today, it must be like you're saying, some kind of flakey connection between us and the server. |
| 15:08.34 | [TK]D-Fender | No, your server to CB |
| 15:09.03 | AaronCharpentier | Sorry I know, I mean us as in our asterisk box, and server as in CB |
| 15:09.12 | [TK]D-Fender | phone>server = OK (do you ever have issues calling the guy next to you?), phone > outside world = bad |
| 15:09.56 | AaronCharpentier | Aye, just a misunderstanding in the wording that's all. I knew what you meant =) |
| 15:10.41 | AaronCharpentier | Hey thanks so much for your time you've been more than helpful, I'm sorry I had to ask so many questions but I fully understand now, I'm sure I'll be back at some point with a new problem ;-) |
| 15:10.55 | dym | When dialing out with asterisk - is there any way to "catch" a congested line within the dialplan? Say i have an automated call and i wanted to jump into a certain context on the line beeing congested - possible? |
| 15:11.24 | [TK]D-Fender | dym, Depends what you're dialing out of and if it returns a distinguishable state |
| 15:12.03 | dym | [TK]D-Fender: SendFax via DAHDI |
| 15:12.36 | [TK]D-Fender | Sendfax doesn't place calls, Dial does. |
| 15:12.42 | [TK]D-Fender | and DAHDI isn't a specific tech. |
| 15:13.10 | dym | well i dialout via a dahdi card |
| 15:13.15 | dym | and then connect sendfax |
| 15:13.22 | [TK]D-Fender | ...yes, and what KIND of card is important |
| 15:13.35 | dym | TE |
| 15:13.42 | [TK]D-Fender | What signalling? |
| 15:15.09 | dym | E1 |
| 15:15.44 | *** join/#asterisk e7e5 (~rudenko@188.134.2.33) |
| 15:16.29 | [TK]D-Fender | PRI? |
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| 15:20.44 | mattsqz | anyone out there using a nortel ip 1535 or lg lvp-2800? |
| 15:23.45 | *** join/#asterisk ddickenson (~android@67-198-0-5.static.grandenetworks.net) |
| 15:30.55 | ddickenson | is there a way to use variables in the dialplan to do something like: exten => s,n,${DB(family/key)} and have the variable fill in the rest of the statement? I keep getting No Application errors even though the variable is set in AstDB with the proper info to finish statement |
| 15:31.47 | ddickenson | say the variable had this set "NoOp(This is a test)" |
| 15:32.16 | [TK]D-Fender | ddickenson, You can't use a variable in place of an appname like that |
| 15:32.31 | ddickenson | bummer... |
| 15:32.40 | [TK]D-Fender | ddickenson, You could use an ExecIf and reference it in there though. |
| 15:34.15 | *** join/#asterisk Jimmy890 (~chatzilla@79.175.98.33) |
| 15:35.58 | ddickenson | I had a theory on how I could allow users to set a variable by calling an extension and basically picking either "1" or "2" and that variable if placed prior to my time of day routing information would either enter a NoOp statement that the dialplan would just fallthrough or a GoTo statement redirecting to another context |
| 15:37.46 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-unvfbujlizihwzjf) |
| 15:37.59 | ddickenson | Does that sound like ExecIf would do the trick? Ill have to do some reading as im not yet familiar with that app |
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| 15:51.05 | eZz | will repeat a question, |
| 15:51.06 | eZz | is it ok if a remote leg sends a multiple 183 with sdp with a different contact field path ? |
| 15:51.10 | eZz | my voip company sends 183 with persistent contact with sdp only in case if their uac can reach an user. In other case - sends a different contact path within one call-id on 183 (only in case the remote uac can't reach an user for some reasons) |
| 16:02.22 | dym | [TK]D-Fender: Sorry - yes PRI |
| 16:06.44 | *** join/#asterisk puzzled (~patrick@2001:980:5e31:1:6ef0:49ff:fe50:659d) |
| 16:08.49 | [TK]D-Fender | dym, then yes you should get a usable DIALSTATUS and HANGUPCAUSE to track this |
| 16:09.32 | dym | [TK]D-Fender: So say I send a fax via the PRI and the Dial hits a congested line - will it fall through to some extension? |
| 16:09.50 | [TK]D-Fender | dym, your dialplan moves on like normal. |
| 16:11.58 | *** join/#asterisk nathan1981 (~nay-fan@93.95.122.2) |
| 16:19.54 | nathan1981 | Hi all, I am seeing a strange problem with one of my server which passes traffic between our users and our provider. On some calls the server is not sending the RTP through to the client after forwarding the 183. Anoyingly it seems pretty intermitant. Most calls seem to work OK and if the client responds with RTP the call seems to always work if they not there is no audio until we recive and |
| 16:19.54 | nathan1981 | then pass the 200 OK on to the client and then they pass audio. example call ladder => http://coolioso.org/junk/callladder.PNG |
| 16:20.16 | nathan1981 | asterisk 1.4.3 no 'r' in the dial plan |
| 16:20.54 | eZz | nathan1981: I have the same issue |
| 16:21.44 | nathan1981 | i have check the SDP in the invites and 183's for working and non working and they all look the same |
| 16:21.49 | [TK]D-Fender | 1.4.3? Seriously? LIke 40 updtes behind a branch that is 5 entire branches behind? |
| 16:22.27 | nathan1981 | yeah pretty old |
| 16:23.45 | p3nguin | I don't know if it's really fair to say a particular branch is 5 branches behind current when they were being developed or at least maintained concurrently. |
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| 16:25.37 | eZz | nathan1981: does your provider is bandwith ? |
| 16:25.38 | p3nguin | It is, however, really friggin' old. |
| 16:25.42 | [TK]D-Fender | p3nguin, Everything pre 1.8 is security only IIRC |
| 16:26.01 | eZz | 1.8 has the same btw |
| 16:26.24 | p3nguin | "were being developed or at least maintained concurrently" |
| 16:26.35 | p3nguin | Not in development now. |
| 16:27.05 | [TK]D-Fender | Security-only means that if what they are looking at is a "bug" then it isn't going to get fixed thus would be a dead duck to them. |
| 16:27.34 | p3nguin | "were" |
| 16:27.40 | p3nguin | past tense |
| 16:27.46 | nathan1981 | I was more looking to see if anyone had seen a similar problem and had a work around |
| 16:28.25 | [TK]D-Fender | p3nguin, Yes, but they ARE asking NOW (current tense) :) |
| 16:28.28 | p3nguin | Regardless, 1.4.3 is still extremely old and may even be considered a fossil by some. |
| 16:29.00 | [TK]D-Fender | I believe the word you're looking for would be antediluvian :p |
| 16:29.09 | p3nguin | I wasn't addressing if it was old or not. I was addressing the fairness of the statement. |
| 16:29.19 | [TK]D-Fender | p3nguin, And I countered on the same :0 |
| 16:29.47 | [TK]D-Fender | I'm all about "fair"... |
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| 16:30.07 | anonymouz666 | [TK]D-Fender: out of curiosity, what version you are using in production? |
| 16:30.45 | [TK]D-Fender | anonymouz666, My office is on a really old version itself which I am in the process of ripping out entirely. At home, 1.6.2, and pening an upgrade |
| 16:31.17 | [TK]D-Fender | anonymouz666, Can't do as much at the office because I'm ripping out one GUI and migrating the whole base to new hardware, OS, * ver and GUI |
| 16:31.47 | anonymouz666 | hmmm here I am using the latest 1.4 in pbxs, latest 1.8 in distributed call centers |
| 16:32.57 | p3nguin | I don't have any better stability using 1.8 than I had in 1.4. |
| 16:33.10 | p3nguin | I only gained a few features by moving up. |
| 16:33.22 | *** join/#asterisk Tim_Toady (~fuzzy@77.49.104.166.dsl.dyn.forthnet.gr) |
| 16:33.37 | SuperNull | anyone know of a place that has the text that is said by each sound file in the sound pack ? |
| 16:33.57 | p3nguin | Yes. It is included with the sounds. |
| 16:34.07 | SuperNull | o rly. |
| 16:34.10 | anonymouz666 | p3nguin: true |
| 16:34.14 | p3nguin | core-sounds-en.txt and extra-sounds-en.txt |
| 16:34.19 | SuperNull | derp. thank you. |
| 16:35.56 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 16:36.49 | nathan1981 | As its stand my problem only happens with some types of client hardware and only some voip providers (e.g. global crossing), there is nothing in logs relating to an RTP problems. It only seems to effect less than 1% of customers and they all seem to have Siemens systems. |
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| 16:46.47 | *** join/#asterisk w32blaster (~w32blaste@46.214.202.20) |
| 16:47.29 | w32blaster | hi |
| 16:47.43 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v030-242.mobile.uci.edu) |
| 16:48.55 | w32blaster | is this the correct place where can I ask about module development? (I am new in in asterisk module dev)? |
| 16:49.01 | *** join/#asterisk vitaliyt (a5c40017@gateway/web/freenode/ip.165.196.0.23) |
| 16:49.09 | p3nguin | #asterisk-dev |
| 16:49.10 | vitaliyt | hello |
| 16:49.24 | w32blaster | ok, thanks |
| 16:49.37 | vitaliyt | can someone help me with a music on hold issue? |
| 16:50.27 | p3nguin | ~ask |
| 16:50.28 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 16:51.56 | vitaliyt | Ive setup MOH on asterisk 1.8 and its setup so that if i call ext 1001 it auto answers and plays music, the issue is that it can only handle one user at a time, to the second caller I just get a fast busy... |
| 16:52.18 | p3nguin | Show me the entire extension 1001. |
| 16:52.20 | p3nguin | ~pb |
| 16:52.21 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 16:52.23 | p3nguin | Pastebin it. |
| 16:52.41 | vitaliyt | one moment |
| 16:53.23 | vitaliyt | http://pastebin.com/LcymAx2v |
| 16:54.37 | p3nguin | This is what it should look like: http://pastebin.com/tuQWgaCp |
| 16:54.42 | vitaliyt | I can paste the CLI output as well... |
| 16:55.24 | p3nguin | Not yet. Make the change and then try two or more calls again. |
| 16:55.37 | vitaliyt | is CHANNEL a defined value? |
| 16:55.51 | vitaliyt | or is it just channel? |
| 16:56.02 | [TK]D-Fender | p3nguin, if the dialplan was good for one, it's good for both |
| 16:56.28 | p3nguin | What I put in the pastebin is what you should copy verbatim. |
| 16:56.37 | vitaliyt | got it. |
| 16:56.55 | p3nguin | You're using deprecated and possibly broken apps in an old-style dialplan. |
| 17:00.08 | vitaliyt | p3nguin: ok, same issue. fast busy to 2nd caller. |
| 17:00.14 | p3nguin | core set verbose 3 |
| 17:00.35 | p3nguin | Make one call, then make the second call. Pastebin the output. |
| 17:02.16 | vitaliyt | http://pastebin.com/yLxGvKnn |
| 17:02.25 | vitaliyt | it looks like it doesnt even see a 2nd caller... |
| 17:03.05 | vitaliyt | ah crap |
| 17:03.07 | vitaliyt | i know why.... |
| 17:03.12 | vitaliyt | brb |
| 17:04.54 | p3nguin | Did you ever try call #1 from that other phone, or was it always call #2? |
| 17:04.54 | vitaliyt | ok, i fixed it. the phone system that im using know, i registered 1001 to point to the asterisk but by default i left the max trunk capacity to 1, dumb me. |
| 17:05.09 | p3nguin | Sounds misconfigured. |
| 17:05.25 | vitaliyt | yea, its working fine now with multiple callers. |
| 17:06.01 | p3nguin | And, as a bonus, you've updated that part of your dial plan to the current century. |
| 17:06.16 | vitaliyt | yep, thanks for that. |
| 17:10.16 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 17:10.22 | vitaliyt | while im at it, every time someone enters the MOH the music file starts from the begining, ive read that it should continue from where it left off, is that accurate? |
| 17:11.23 | vipkilla | what does "CHANUNAVAIL" mean in asterisk's CDRs? |
| 17:11.28 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 17:12.10 | Katty | hello my asterisk does not work at all how to fix plz??? |
| 17:14.24 | vipkilla | anybody in here? |
| 17:14.32 | Katty | no. |
| 17:14.37 | Katty | i mean, yes. |
| 17:14.50 | vipkilla | does anybody know what "CHANUNAVAIL" mean in asterisk's CDRs? makes no sense to me |
| 17:15.12 | vipkilla | and google is turning up 0 results on my inquiry |
| 17:15.35 | Katty | chan unavailable..means.... |
| 17:15.51 | vitaliyt | vipkilla: really? you googled "asterisk chanunavail" and you got 0 results? |
| 17:15.54 | vitaliyt | are you serious? |
| 17:15.59 | Katty | the channel is...unavailable |
| 17:16.03 | vipkilla | yes it is strange |
| 17:16.24 | vipkilla | what is difference between BUSY and CHANUNAVAIL in asterisk's CDRs? |
| 17:16.26 | vitaliyt | http://lmgtfy.com/?q=asterisk+CHANUNAVAIL |
| 17:16.30 | Katty | busy means the line returned a busy |
| 17:16.37 | Katty | the chan unavailable means...it was not there to use |
| 17:16.45 | Katty | it's gone on vacation |
| 17:16.46 | vipkilla | i dont understand... what was returned? |
| 17:16.57 | Katty | HELLO DEAR LINE |
| 17:17.00 | Katty | LINE: GO AWAY I"M BUSY |
| 17:17.03 | Katty | HELLO DEAR LINE |
| 17:17.05 | Katty | (no answer) |
| 17:17.09 | Katty | DEAR LINE ARE YOU THERE |
| 17:17.10 | Katty | (no answer) |
| 17:17.13 | Katty | LINE WHERE ARE YOU |
| 17:17.14 | vitaliyt | lol |
| 17:17.15 | Katty | (crickets) |
| 17:17.21 | vitaliyt | nice |
| 17:17.23 | eZz | hah |
| 17:17.44 | Katty | that's the difference. |
| 17:18.01 | Katty | you could check your qualify= in sip.conf |
| 17:18.02 | vipkilla | so it means the call never connected to anyting? |
| 17:18.06 | Katty | if it's a sip chan that's unavailable. |
| 17:18.14 | Katty | yes. all the asterisk got back was crickets. |
| 17:18.17 | Katty | apparently it's gone on vacation |
| 17:18.23 | Katty | have you checked the bahamas? i hear it's lovely this time of year |
| 17:19.10 | vipkilla | thanks Katty |
| 17:19.17 | dym | Is there a way to "simulate" a congested channel in the dialplan? |
| 17:19.34 | Katty | yeah |
| 17:19.36 | *** join/#asterisk dxd828 (~dxd828@host81-133-31-249.in-addr.btopenworld.com) |
| 17:19.37 | Katty | unplug your t1 |
| 17:19.45 | Katty | oh wait, that's not a simulation |
| 17:19.46 | Katty | nevermind |
| 17:20.28 | dxd828 | Hey guys need some help, just shutdown and moved my Asterisk server (been up over 100 days) get to the web panel and it just show a white blank page at /admind/config.php |
| 17:20.31 | dxd828 | any ideas? |
| 17:20.48 | [TK]D-Fender | dxd828, #freepbx <-------- |
| 17:20.49 | eZz | php error |
| 17:20.56 | Katty | hi fender bender |
| 17:21.06 | dxd828 | thanks |
| 17:21.11 | [TK]D-Fender | Katty, Mew |
| 17:28.42 | p3nguin | dym: Congestion() |
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| 17:35.01 | dym | p3nguin: thanks got that already (: |
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| 18:27.06 | FLeiXiuS | How can I pass variables between contexts? Do I set a global or is there a fancy way to pass them through Dial? |
| 18:28.13 | p3nguin | They pass on the channel. Context has nothing to do with it. |
| 18:28.40 | p3nguin | But if you are spawning new channels off the parent channel, you'll need to think about inheritance. |
| 18:28.56 | FLeiXiuS | p3nguin, Thats what my issue is then... |
| 18:29.18 | FLeiXiuS | I'm passing variables from one context (created by AMI) to another and I'm loosing the var. |
| 18:29.35 | p3nguin | Again, context has nothing to do with it. |
| 18:29.45 | [TK]D-Fender | FLeiXiuS, Thre is no concept of scope for * variables. |
| 18:30.11 | p3nguin | It is limited only by the current channel. |
| 18:30.20 | [TK]D-Fender | correct |
| 18:30.53 | FLeiXiuS | OH! That explains everything. I was thinking vars were contained via contexts. |
| 18:30.55 | p3nguin | If you are creating new channels off the original, prepend your variable with one underscore when you set it. That will allow it to be inherited by one new level. |
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| 18:31.19 | p3nguin | Set(_myVar=data) |
| 18:31.51 | p3nguin | Contexts are only containers for extensions. |
| 18:31.59 | p3nguin | Extensions are where it all happens. |
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| 18:53.57 | mattsqz | woot! now the wifi phone on the boss' sailboat is on the office pbx. fantastic. i dont even want to know how much a pbx vendor would have charged to set that up. |
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| 19:12.23 | vitaliyt | exit |
| 19:12.30 | vitaliyt | woops |
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| 20:05.25 | *** join/#asterisk Hive (~Hive@173-165-205-1-jacksonville.hfc.comcastbusiness.net) |
| 20:08.27 | Hive | I've got this web interface I'm making that lets users edit their phone system. I'm trying to figure out how to handle reloading config files after they are changed by the users. In total, 6 config files could need to be updated whenever changes are made. I'm wondering if it would be better for me to execute a 'config reload whatever' for each file, or just 'core reload'. Any input? |
| 20:10.08 | [TK]D-Fender | Hive, Depends on which files. |
| 20:11.32 | Hive | extensions.conf, musiconhold.conf, voicemail.conf, sip.conf, extensions_additional.conf, another_extensions_additional.conf (those last 2 aren't the real names :P) |
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| 20:12.28 | [TK]D-Fender | IIRC voicemail and musiconhold have to have the actual module reloaded, not just the configs |
| 20:12.28 | Hive | im trying to figure out the best way to do this without bogging down the server |
| 20:13.02 | Hive | ahh ok good to know |
| 20:13.37 | Hive | I guess asterisk handles 'core reload' pretty quickly now anyways |
| 20:13.50 | Hive | but im just not sure which would consume less resources on the server |
| 20:13.57 | Hive | 6 individual reloads, or a core reload |
| 20:14.28 | *** join/#asterisk omani (~hasan@33.37.69.80.in-addr.net-lab.net) |
| 20:15.00 | [TK]D-Fender | individual shold because it's not reloading things you weren't specifically caring about... |
| 20:15.47 | omani | we have a analoge pbx. will change to voip. but what I dont understand is, how it is possible to have our numbers like (1234-131 and 1234-132) although we have only a numberblock from 00-29 |
| 20:15.55 | omani | am besten brauche ich jemanden, der deutsch kann. |
| 20:16.24 | Hive | seems like a logical answer, thanks D-Fender |
| 20:20.33 | p3nguin | moh reload and voicemail reload should be just fine for reloading their respective files. |
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| 20:31.55 | Hive | hmm |
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| 20:37.42 | p3nguin | I personally cannot stand "reload" and/or "core reload". It really gets under my skin for an unknown reason. |
| 20:38.19 | p3nguin | Therefore, you will never catch me doing it. |
| 20:43.28 | Hive | what do you use instead? |
| 20:44.40 | leifmadsen | module reload <something specific> |
| 20:45.08 | leifmadsen | or if the module has it's own thing, then you can use something like p3nguin pointed out with: moh reload, voicemail reload, sip reload, etc... |
| 20:45.20 | leifmadsen | reload what you need, not everything at once |
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| 20:45.32 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 20:45.53 | Hive | ohhhh i misunderstood p3nguin, i thought he meant any reload such as config reload <something specific> |
| 20:46.01 | p3nguin | or if the module has its own thing. |
| 20:46.02 | Hive | I guess you meant just "reload"? |
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| 20:46.46 | p3nguin | I meant, literally, reload. |
| 20:48.16 | Hive | Gotcha. |
| 20:48.22 | p3nguin | Keep in mind that from time to time you may run into a module which does not support reloading. For those, you need to unload and then load again. |
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| 21:01.29 | [TK]D-Fender | Checkout time, BBL |
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| 21:18.05 | karl370 | I have a question in regards to sms text messaging. Is there specific hardware that's needed? or does some protocol need to be supported on the line? I have a PRI line. |
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| 21:25.29 | leifmadsen | karl370: essentially, you can't do it with asterisk |
| 21:25.47 | leifmadsen | while app_sms exists, it's for a very specific situation that almost no one has |
| 21:25.59 | leifmadsen | (some particular european carrier) |
| 21:27.11 | karl370 | that sucks. |
| 21:28.42 | leifmadsen | might want to look at other software for SMS (if it exists) |
| 21:29.05 | Joel | the best way to SMS is to use an email to sms gateway |
| 21:29.35 | karl370 | That's fine for sending them, but how about receiving them? |
| 21:29.42 | Joel | same. |
| 21:31.41 | karl370 | ok, maybe I'm not following exactly then. I know the carriers allow you to email to something like 7145551234@verizon.com and then the text will go to the 7145551234 phone number. What happens when they reply to the txt? |
| 21:31.53 | Joel | google email sms gateway |
| 21:32.10 | karl370 | alright |
| 21:32.15 | Joel | that's an email address, not an email sms gateway. |
| 21:33.09 | Joel | any decent service will even provide you with a developer account where you get X amount of messages to test with |
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| 21:38.53 | Joel | http://www.bulksms.com/int/w/eapi-sms-gateway.htm for example |
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| 22:05.03 | min3r | I went to a local thrift store and got a 2wire ATA, internal wifi card, and ISA hardware modem, standard PC ac power adapter, and ethernet cable for $10 |
| 22:05.50 | min3r | Too bad they wanted $50 for a win95 box with no HDD. plus another $30 for the monitor, and $5 for the old keyboard |
| 22:05.57 | min3r | but i got all the above so cheap. |
| 22:06.09 | min3r | they have no idea how to price items |
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| 22:14.53 | leifmadsen | isa hardware modem? |
| 22:15.01 | leifmadsen | who even has a computer who can run that anymore ;) |
| 22:15.15 | leifmadsen | peers at the box of hardware he has that contains a USR 33.6 ISA modem |
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| 22:44.54 | rdegges | Hey all, quick question about using MeetMe + DAHDI in 1.8 latest-- |
| 22:45.23 | rdegges | I'm currently using DAHDI dummy--is it possible that I'd get better performance if I was using a hardware source for timing? Like a cheap digium fxo card or something? |
| 22:45.27 | rdegges | Or would that make no difference whatsoever? |
| 22:45.53 | rdegges | I'm having stability problems with asterisk =/ We crash once a week or so, and the only app we use is meetme. |
| 22:46.09 | rdegges | The logs don't give any useful information other than a generic 'frame' error. |
| 22:46.40 | rdegges | I'm trying to figure out whether I should possibly get some cards in our boxes to possibly improve stability or not :x |
| 22:47.08 | [TK]D-Fender | It's should crash on you... that sounds like something else.... |
| 22:47.48 | rdegges | You're suggesting it's another issue? :o |
| 22:49.24 | rdegges | I've been having this issues like __forever__. Upgrading Asterisk revisions hasn't helped at all =/ |
| 22:49.47 | rdegges | We're running ubuntu-server (64-bit), latest kernel revision, latest dahdi, latest asterisk stable. |
| 22:49.50 | rdegges | Still the same thing though =/ |
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| 23:58.16 | kessius | hi pabelange and friends, one day off, and miss irc |
