| 00:00.18 | tehrabbitt-1 | any ideas/ |
| 00:01.26 | Maliuta | misconfigured phones |
| 00:01.46 | tehrabbitt-1 | Maliuta: phones are supposed to pull the CID from the server |
| 00:01.47 | tehrabbitt-1 | hm |
| 00:01.53 | tehrabbitt-1 | one is using a PAP2, maybe that has a date setting |
| 00:02.03 | p3nguin | I'm not sure if time is part of CID or not. |
| 00:02.10 | p3nguin | I'm thinking not. |
| 00:02.35 | *** join/#asterisk Diffen (~diffen@c-f477e555.042-17-73746f11.cust.bredbandsbolaget.se) |
| 00:03.01 | tehrabbitt-1 | p3nguin: yea i know the SCCP phone shows 1 hour ahead |
| 00:03.03 | p3nguin | The ATA should have a time and date setting. |
| 00:03.11 | tehrabbitt-1 | the PAP2 phones show like several hours off |
| 00:03.24 | p3nguin | It should have an offset. |
| 00:04.18 | tehrabbitt-1 | p3nguin: Maybe it's set to the wrong offset in the PAP2 |
| 00:04.20 | tehrabbitt-1 | hm |
| 00:04.29 | tehrabbitt-1 | i'm gonna check once i figure out the IP of it heh |
| 00:05.15 | p3nguin | Pick up the attached phone and press **** |
| 00:05.28 | p3nguin | (I think) |
| 00:05.48 | tehrabbitt-1 | lol or just use nmap :-D |
| 00:06.16 | tehrabbitt-1 | X_X theres a timezone function |
| 00:06.21 | tehrabbitt-1 | it's set to GMT -8:00 |
| 00:06.23 | tehrabbitt-1 | that would be why |
| 00:06.25 | tehrabbitt-1 | Heh |
| 00:06.30 | tehrabbitt-1 | should be -5:00 for EST |
| 00:06.49 | p3nguin | Should be -4 for EDT |
| 00:06.50 | Maliuta | no, EST is GMT+10 |
| 00:07.13 | p3nguin | Eastern Standard Time (USA) |
| 00:07.16 | Maliuta | someone just hit one of my pet peeves |
| 00:07.21 | tehrabbitt-1 | Ah |
| 00:07.22 | tehrabbitt-1 | lol |
| 00:07.38 | tehrabbitt-1 | p3nguin: so -4 not -5? |
| 00:07.45 | Maliuta | Eastern Standard Time depends on the country you are in. The US is not the centre of the world |
| 00:08.12 | Maliuta | I am in the EST zone, and it's GMT+10 :P |
| 00:08.58 | p3nguin | So what do you think EST is? |
| 00:09.23 | tehrabbitt-1 | Much better... -4 worked for the PAP2 phones |
| 00:09.31 | tehrabbitt-1 | but the SCCP phone still shows 1 hour ahead X_X |
| 00:09.39 | tehrabbitt-1 | i'm guessing it's in either the XML file or the sccp.conf file |
| 00:09.49 | Maliuta | Technically it should be USEST, and I'm in AEST |
| 00:10.06 | p3nguin | That's better. EST is in the US. |
| 00:10.18 | Maliuta | no it's not |
| 00:10.23 | p3nguin | yeah, it really is. |
| 00:10.36 | Maliuta | only if you're an US imperial dog |
| 00:10.42 | p3nguin | Go check timeanddate.com |
| 00:10.42 | *** join/#asterisk matiasjrossi (~matias@host207.190-138-186.telecom.net.ar) |
| 00:11.40 | Maliuta | p3nguin: go visit a country that isn't the US and figure out why when they say EST they don't mean east coast US time |
| 00:12.04 | tehrabbitt-1 | nothing in the sccp.conf in regard to timezones |
| 00:12.05 | tehrabbitt-1 | hm |
| 00:12.50 | p3nguin | Why are you being such an asshole over our country using using the same timezone as your country? Is it really THAT huge of an issue for you? |
| 00:13.33 | p3nguin | I mean, I'm not getting all pissed off because you want to use EST in Australia. I couldn't care less. |
| 00:26.19 | *** join/#asterisk shmaltz (~chatzilla@mail.dmaven.com) |
| 00:26.42 | shmaltz | hi everyone |
| 00:26.55 | tehrabbitt-1 | p3nguin: have you tried out the newer chansccp modules yet? |
| 00:26.59 | tehrabbitt-1 | V3.0 |
| 00:27.02 | tehrabbitt-1 | just released like this month |
| 00:27.15 | shmaltz | any recommendations on an ITSP that can give me NY LI based DID with multiple inbound channels? |
| 00:27.20 | p3nguin | I'm using whatever was on svn two days ago. |
| 00:27.28 | tehrabbitt-1 | lol thats the one |
| 00:27.50 | shmaltz | price is not an issue, quality is |
| 00:27.59 | p3nguin | Release: 3.0.2 STABLE - 2687 |
| 00:28.35 | carrar | Who uses version numbers |
| 00:28.41 | carrar | thats so last year |
| 00:29.03 | p3nguin | Oh, I guess I built that yesterday, not two days ago. |
| 00:29.13 | p3nguin | 2011-08-17 14:21:03 UTC |
| 00:29.40 | carrar | (making fun of firefox) |
| 00:29.47 | carrar | Asa |
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| 00:34.04 | tehrabbitt-1 | sweet, upgrade to the latest sccp-b was simple |
| 00:34.04 | tehrabbitt-1 | :-D |
| 00:34.12 | tehrabbitt-1 | doesn't fix the time thing though X_X |
| 00:34.13 | tehrabbitt-1 | lol |
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| 00:48.07 | DrDigital | anyone use vitelity here? |
| 00:48.24 | p3nguin | yep |
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| 01:08.18 | michael-i | One last try before giving up today: anyone know why Bridge and Unlink events would be firing over the AMI on DTMF key presses? Asterisk 10 beta, SIP only environment, dtmfmode=rfc2833, directmedia=no |
| 01:11.29 | *** part/#asterisk trhodes (tom@cardinal.lizella.net) |
| 01:12.51 | *** join/#asterisk kaushal (~kaushal@14.97.122.46) |
| 01:12.59 | kaushal | WIMPy: hi |
| 01:13.41 | kaushal | Just to give an update about the issue being reported few days back |
| 01:13.51 | kaushal | the issue stands as it is |
| 01:14.52 | kaushal | Have implemented the suggestion and the error "got hangup, cause 101" is seen in the debug logs |
| 01:16.25 | kaushal | Do you need more information ? |
| 01:20.13 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 01:25.39 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 01:30.13 | kaushal | tonsofpcs: Hi |
| 01:47.26 | *** join/#asterisk lkthomas (~lkthomas@59.152.236.158) |
| 01:47.29 | lkthomas | hey guys |
| 01:50.04 | lkthomas | it might be out of topic |
| 01:50.14 | lkthomas | but yet we can't get receive fax working |
| 01:50.31 | p3nguin | Are you using fax for asterisk? |
| 01:50.38 | lkthomas | again, the logic is like this: PSTN <> asterisk <> ATA <> Fax machine |
| 01:50.46 | lkthomas | we could hear fax tone |
| 01:50.54 | lkthomas | when we dail using normal phone I mean |
| 01:51.00 | lkthomas | but fax just error and die out |
| 01:52.39 | *** join/#asterisk Hanumaan (~Hanumaan@dslb-092-075-145-204.pools.arcor-ip.net) |
| 01:56.06 | ChannelZ | What is the PSTN<->Asterisk link? |
| 02:00.32 | *** join/#asterisk xpot (~xpot@166-70-100-198.ip.xmission.com) |
| 02:07.41 | *** join/#asterisk jwiggins (~James@gateway/tor-sasl/jwiggins) |
| 02:08.09 | lkthomas | T1 |
| 02:08.13 | lkthomas | as I mention yesterday :) |
| 02:10.49 | ChannelZ | I'd ask some more questions but I guess I should aleady know the answers then. |
| 02:11.26 | p3nguin | You're either supposed to remember from the last time, or scroll back a full day. |
| 02:11.31 | p3nguin | Or be psychic. |
| 02:12.03 | p3nguin | I can't even remember what people talked about this morning when I came on. |
| 02:13.18 | *** join/#asterisk jayson_r (~jayson@cpe-071-076-046-081.sc.res.rr.com) |
| 02:19.29 | kaushal | Can some one please help me with my issue ? |
| 02:20.21 | kaushal | Its being going on for quite some time |
| 02:22.47 | *** join/#asterisk pzn (~pzn@pdpc/supporter/active/pzn) |
| 02:37.18 | dlynes | lkthomas, asterisk->ata->fax machine could be just about anything, from the phase of the moon, to your microwave oven for why it won't work |
| 02:38.31 | dlynes | lkthomas, try setting your fax machine to only work at 9600 baud, to see if it fixes the problem...afaik, 14400 is patented, and not supported (except for maybe in FFA) |
| 02:54.24 | kaushal | Where do i seek help for this issue ? |
| 03:02.25 | lkthomas | wait |
| 03:03.57 | ChannelZ | What is the issue |
| 03:04.40 | lkthomas | how could I show specific number log ? |
| 03:04.55 | lkthomas | problem is that when I do asterisk -rvvv it shows tons of unrelated things |
| 03:06.31 | lkthomas | ChannelZ, any idea ? |
| 03:06.37 | lkthomas | we have a busy phone system |
| 03:06.41 | lkthomas | so tons of call flowing around |
| 03:06.52 | lkthomas | not sure how could I track down specific number log |
| 03:08.41 | p3nguin | What is a number log? |
| 03:09.36 | kaushal | ChannelZ: Are you referring to me ? |
| 03:11.28 | *** join/#asterisk Defraz (~Defraz@184-155-137-27.cpe.cableone.net) |
| 03:11.48 | lkthomas | for example, I want to track down dial out call from number 500 |
| 03:12.05 | ChannelZ | CDR |
| 03:12.23 | lkthomas | sorry ? |
| 03:12.36 | ChannelZ | /var/log/asterisk/cdr-csv |
| 03:12.57 | p3nguin | Or, in the case of a poorly designed SIP peer entry with the name 500, you can use sip debugging. |
| 03:13.19 | lkthomas | let's try CDR first |
| 03:13.23 | lkthomas | hang on |
| 03:13.48 | p3nguin | If you're just wanting to see the detail about the call, CDR is where you want to go. |
| 03:14.32 | p3nguin | If you care about what happens inside the call, sip debug might be what you're interested in. I'm not sure what exactly you're trying to find out. |
| 03:24.52 | *** join/#asterisk radic (~radic@dslb-094-216-249-057.pools.arcor-ip.net) |
| 03:25.37 | ChannelZ | kaushal: and yes I was referring to you |
| 03:25.44 | kaushal | ChannelZ: sure |
| 03:26.01 | kaushal | ChannelZ: Let me explain the issue currently i am facing |
| 03:27.24 | kaushal | so when i initiate campaign for Outbound Dialling using Asterisk on 8 PRI Sangoma Card with 25 channels on each PRI line, i get Cause: Message not compatible with call state (101) |
| 03:27.46 | kaushal | whereas when i try with just 4 PRI Lines it works perfectly fine |
| 03:28.44 | kaushal | So i have my own Dialer written in java which uses Asterisk Library and does this campaign |
| 03:29.34 | kaushal | so the campaign consits of 200 Phone numbers and attached sound file |
| 03:29.47 | kaushal | consists* |
| 03:30.13 | *** join/#asterisk corretico (~luis@201.201.44.82) |
| 03:30.20 | kaushal | ChannelZ: Do you need more information ? |
| 03:30.48 | kaushal | I am using mpg123 player on CentOS 5.6 |
| 03:33.12 | lkthomas | very very interesting |
| 03:33.24 | lkthomas | when I call using a phone, I could hear fax ring tone |
| 03:33.46 | lkthomas | but then when I use a real fax machine to send fax to that problem site, it shows following: |
| 03:33.50 | ChannelZ | Hmm sorry I'm not a PRI expert (or even novice) |
| 03:34.02 | lkthomas | ,"default","""FAX Out"" <2222>","Zap/30-1","Zap/1-1","NoOp","Thay hangs up!123456","2011-08-19 11:30:01",,"2011-08-19 11:30:59",58,0,"NO ANSWER","DOCUMENTATION" |
| 03:34.11 | lkthomas | no answer ?! |
| 03:36.09 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 03:40.52 | ChannelZ | lkthomas: Your ATA uses ulaw? |
| 03:42.10 | lkthomas | Mlaw |
| 03:42.22 | p3nguin | mlaw? I don't think so. |
| 03:42.42 | lkthomas | sorry ulaw |
| 03:44.23 | lkthomas | yes, using ulaw |
| 03:45.55 | ChannelZ | And what is the ATA? |
| 03:48.29 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-170.cablep.bezeqint.net) |
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| 03:53.08 | kaushal | Is there a utility available in Asterisk for Outbound Dialling for more than 200 Phone Numbers ? |
| 03:53.30 | kaushal | I mean generating 200 Outbound Calls concurrently |
| 03:58.19 | p3nguin | If you have resources for that many calls and you have that many people to call, I don't see why it can't be done with ease. |
| 03:58.36 | ChannelZ | How were you doing it before your java thing? Call files or.. |
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| 04:18.03 | kaushal | ChannelZ: sorry was away |
| 04:18.33 | kaushal | ChannelZ: i use channel originate DAHDI/g0/xxxxxxxxxx Application MP3Player /home/kaushal/obd-demo.mp3 |
| 04:18.42 | kaushal | similarly for g1-7 |
| 04:19.07 | kaushal | but this is just individual called number |
| 04:19.43 | p3nguin | So do it several hundred times with different numbers. |
| 04:19.52 | kaushal | p3nguin: ok |
| 04:19.53 | p3nguin | I use a bash script for that. |
| 04:20.06 | kaushal | p3nguin: can it be done 200 calls concurrently ? |
| 04:20.24 | p3nguin | The bash script reads from a file which has a list of numbers. You could also use a database with numbers. |
| 04:20.49 | p3nguin | If you have 200 channels available, and if you have enough system resources, sure it will. |
| 04:21.00 | kaushal | p3nguin: yes i do have |
| 04:21.06 | *** join/#asterisk MrOli (46bb8733@gateway/web/freenode/ip.70.187.135.51) |
| 04:21.17 | MrOli | hello all |
| 04:21.28 | MrOli | any Polycom gurus in the house ? |
| 04:23.37 | kaushal | p3nguin: Any sample bash script ? |
| 04:23.38 | ChannelZ | kaushal: are you doing that via AMI or using asterisk -x ? |
| 04:23.47 | kaushal | AMI ? |
| 04:23.53 | ChannelZ | I guess no then. |
| 04:24.26 | p3nguin | I don't know how to write programs 'n shit, so I use asterisk -rx 'originate ... ... ...' in my script. |
| 04:25.34 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-vigqnuetbyxaxzpq) |
| 04:25.56 | kaushal | p3nguin: ok |
| 04:26.23 | kaushal | p3nguin: can you please share me the snippet of the bash script |
| 04:26.43 | lkthomas | ChannelZ, Grandstream HT386 |
| 04:28.35 | ChannelZ | Does it do t.38 or have any sort of built-in fax detection that is maybe enabled and hosing things up? |
| 04:28.51 | lkthomas | we tested t38 and it does not work too |
| 04:28.56 | lkthomas | it does have fax detection feature |
| 04:29.14 | kaldemar | MrOli: just ask what you intend to ask. you're more likely to get an answer. |
| 04:30.30 | ChannelZ | lkthomas: what country are you in |
| 04:30.38 | kaldemar | last time i tried a HT386, it was hosed up in pretty much every possible way. |
| 04:30.50 | lkthomas | ChannelZ,Hong Kong, why |
| 04:30.56 | ChannelZ | oof nevermind |
| 04:31.25 | ChannelZ | I was going to offer to send one from here via FFA and see if it provided any insight as to why it won't connect, but sorry I'm not going to call hong kong :) |
| 04:31.43 | lkthomas | what's FFA ? |
| 04:32.03 | lkthomas | ChannelZ, you could tell me how to do and debug here |
| 04:32.44 | ChannelZ | Fax For Asterisk |
| 04:32.57 | lkthomas | well, we don't have that setup here |
| 04:33.15 | ChannelZ | or.. do you have guest SIP setup? |
| 04:33.26 | lkthomas | what do you mean by guest ? |
| 04:33.38 | ChannelZ | anonymous |
| 04:33.43 | lkthomas | where |
| 04:33.44 | ChannelZ | Can I call you via SIP |
| 04:33.50 | lkthomas | no |
| 04:34.07 | lkthomas | you want to talk ? |
| 04:34.21 | ChannelZ | No.. to try faxing without calling international |
| 04:34.37 | lkthomas | our asterisk is on private network |
| 04:34.38 | lkthomas | sorry |
| 04:34.54 | lkthomas | ok |
| 04:35.02 | lkthomas | we tested two fax machines there |
| 04:35.16 | lkthomas | one, without phone plugin to the fax machine, the fax ring tone will not appear |
| 04:38.22 | ChannelZ | hmm actually I think I can call hong kong for the same price as local calls.. interesting.. |
| 04:38.53 | *** join/#asterisk cerberus_za (~coert@41-134-110-162.dsl.mweb.co.za) |
| 04:50.48 | MrOli | Here's the question: the default behavior of a Polycom SoundPoint IP phone when a SECOND call comes in, is to beep in your ear and blink the second line key... Is there a way to change this to make the actual phone RING just like when it is a first call ? |
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| 04:58.32 | p3nguin | kaushal: I'm not sure you'd understand what I'm doing in the script if you saw it. |
| 05:03.53 | *** join/#asterisk singler (~singler@81-7-123-162.static.zebra.lt) |
| 05:06.31 | kaushal | p3nguin: if you can share i would have fair amount of idea |
| 05:06.40 | kaushal | p3nguin: Thanks anyways |
| 05:06.49 | p3nguin | Okay. http://pastebin.com/G95Fqam9 |
| 05:07.18 | p3nguin | I have a text file with my phone numbers in it. I use a - on a line to break from calling. |
| 05:07.31 | kaushal | p3nguin: sure and Thanks a lot |
| 05:07.41 | kaushal | I would update you once i am done reading it |
| 05:07.47 | kaushal | p3nguin: Thanks |
| 05:07.49 | p3nguin | I typically won't call more than 10 numbers at a time, so the 11th line is a - by itself. |
| 05:07.54 | kaushal | ok |
| 05:08.02 | kaushal | p3nguin: much appreciated |
| 05:08.14 | p3nguin | I couldn't think of a better way to do what I was trying to do, so that's why I have what I have. |
| 05:08.22 | p3nguin | <--- not a programmer |
| 05:08.24 | kaushal | p3nguin: will you be available for sometime ? |
| 05:08.43 | p3nguin | I'll be here for a bit longer, then gone until later. |
| 05:08.52 | kaushal | US Timezone ? |
| 05:09.06 | kaushal | I have just started my day here |
| 05:09.08 | kaushal | IST |
| 05:09.10 | p3nguin | Yes, I'm in the US. The time is 00.09 now. |
| 05:09.14 | kaushal | :( |
| 05:09.19 | kaushal | midnight |
| 05:09.37 | kaushal | p3nguin: sorry for keeping you awake |
| 05:09.55 | p3nguin | It's okay. I'm just wrapping up for the day, then to bed I will go. |
| 05:10.48 | kaushal | p3nguin: can i contact you if i get into issues ? |
| 05:10.53 | p3nguin | Yes. |
| 05:11.03 | kaushal | p3nguin: here or email ? |
| 05:11.11 | p3nguin | here would be good for me. |
| 05:11.14 | kaushal | ok |
| 05:11.24 | kaushal | p3nguin: can i pvt message you ? |
| 05:12.08 | p3nguin | Do you have some secret information that you don't want to share here? |
| 05:12.12 | kaushal | yeah |
| 05:12.21 | kaushal | hence the concern |
| 05:12.30 | p3nguin | Go ahead with the privates. |
| 05:12.35 | kaushal | p3nguin: thanks |
| 05:15.30 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
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| 05:33.09 | ChannelZ | And here you were looking down on my jerk circle earlier. |
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| 05:38.59 | *** join/#asterisk catphish (~catphish@gateway.office.atechmedia.net) |
| 05:39.36 | p3nguin | Just because I saw his privates doesn't mean I liked it. |
| 05:39.55 | joako | What the hell is going on here? |
| 05:40.00 | p3nguin | haha |
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| 05:48.50 | ChannelZ | joako: Did you ever see The Joe Schmoe Show |
| 05:48.52 | ChannelZ | ? |
| 05:52.59 | joako | no |
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| 06:18.04 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 06:18.09 | schmidts | good morning |
| 06:19.19 | james_zhu | :) |
| 06:22.58 | awk | zap it in a zeebee can. |
| 06:28.28 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 06:28.32 | ChannelZ | shakes his booty |
| 06:34.22 | *** join/#asterisk oej (~olle@office.ipvision.dk) |
| 06:39.17 | *** join/#asterisk jacc0 (~jacc0@D522448D.static.ziggozakelijk.nl) |
| 06:39.50 | jacc0 | morning |
| 06:40.05 | jacc0 | found another memory leak |
| 06:40.15 | jacc0 | don't think it's thed same as I found last time |
| 06:40.37 | jacc0 | s/thed/the |
| 06:41.09 | schmidts | jacc0 there is everyday something new to discover :D |
| 06:41.33 | jacc0 | ;) true |
| 06:42.04 | jacc0 | everything has been running stable for oevr 2 weeks :) I'm already happy about that |
| 06:42.31 | jacc0 | and it's still running ; only asterisk is taking up about 1gb of mem |
| 06:42.55 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
| 06:43.07 | schmidts | which version? |
| 06:45.10 | jacc0 | 1.8.5 |
| 06:45.42 | jacc0 | if i'm interperting my gdb output right ; buf = '\000' <repeats 6980 times> |
| 06:45.51 | jacc0 | seems to be a wast of mem |
| 06:45.56 | jacc0 | I'm i right? |
| 06:46.07 | jacc0 | <PROTECTED> |
| 06:47.06 | schmidts | buf='\0' is just an empty string terminated at the beginning and with waste in it |
| 06:47.13 | schmidts | but i will take a look at the code |
| 06:47.20 | *** join/#asterisk zkn (~zkn@195.222.14.202) |
| 06:49.29 | schmidts | char buf[8192] this is a string with a fixed length so i cant see a waste here |
| 06:50.35 | jacc0 | I'll post gdb |
| 06:50.38 | jacc0 | one monet |
| 06:50.40 | schmidts | ok |
| 06:50.41 | jacc0 | *moment |
| 06:51.03 | schmidts | dont know how long monet used to take for an picture but i think it was more than a month :D |
| 06:52.24 | jacc0 | http://pastebin.com/GmwyJrXv |
| 06:52.29 | jacc0 | about this long :p |
| 06:53.08 | jacc0 | core show locks doesn't show anything and asterisk seems idle |
| 06:53.31 | *** join/#asterisk KNERD (~KNERD@adsl-99-72-119-220.dsl.hrlntx.sbcglobal.net) |
| 06:55.01 | jacc0 | asterisk is peeking to 99% cpu useage right now |
| 06:55.42 | jacc0 | and taking up 832mb ram |
| 06:56.02 | jacc0 | cpu usage is back to normal |
| 06:56.09 | jacc0 | 0.3 % |
| 06:56.36 | schmidts | jacc0 could you do a core show memory (or something equal to this ;) |
| 06:56.41 | schmidts | the gdb output seams normal |
| 06:56.42 | *** join/#asterisk stix (~stix@193.89.191.209) |
| 06:57.10 | zkn | Hi, regarding dial plan syntax of ExecIf application, is it possible to use more than one conditional option that corresponds to the same true and false arguments without writing separate ExecIf condition test for each option? |
| 06:57.17 | jacc0 | let's wait for asterisk to return to idle state; I have an incomming call |
| 06:57.33 | jacc0 | @zkn: yes |
| 06:57.36 | schmidts | jacc0 the show memory should do anything |
| 06:58.16 | jacc0 | execIf($[$["${CALLERID(number)}"="SU202"] & $["${EXTEN}"="103"] ]?hangup) |
| 06:58.22 | schmidts | zkn if i understand you right you are looking for something like this: exten => _X!,n,ExecIf($[$[1=1]&$[2=2]]? ... |
| 06:58.26 | jacc0 | use & or | |
| 06:58.42 | schmidts | and a $[ ] about each term ;) |
| 06:59.16 | schmidts | jacc0 i meant schould not do anything bad to your system ;) |
| 06:59.19 | zkn | would also that work? ExecIf($["${DIALSTATUS}" = "CHANUNAVAIL" || "NOANSWER"]?foo:bar |
| 06:59.48 | jacc0 | it does nothing at all :0 |
| 07:00.08 | jacc0 | i need to set some compiler flags i guess |
| 07:00.11 | schmidts | ExecIf($[ $["${DIALSTATUS}" = "CHANUNAVAIL"] | $[ "${DIALSTATUS}" = "NOANSWER" ]?]?foo:bar |
| 07:00.17 | schmidts | jacc0 AFAIK no |
| 07:00.20 | schmidts | let me have a look |
| 07:00.44 | zkn | okay, got it, thanks! |
| 07:02.25 | schmidts | jacc0 hmm ok maybe it really depends on a compiler flag :( |
| 07:03.55 | schmidts | jacc0 you was right its the MALLOC_DEBUG option, sorry my fault |
| 07:04.51 | jacc0 | Okay, I will recompile (using 1.8.6-rc1), but we have to wait for the event to happen again |
| 07:06.14 | schmidts | jacc0 how many peers do you have on this system, and how many channels? cause 1 GB doesnt scare me |
| 07:06.54 | jacc0 | I have one trunk with a pbx |
| 07:07.00 | jacc0 | nothing more |
| 07:07.08 | schmidts | OK thats not a lot, |
| 07:07.10 | jacc0 | as asterisk was idle; no active channels |
| 07:07.16 | schmidts | many sip messages? |
| 07:07.27 | schmidts | maybe you can take a look at sip show channels |
| 07:08.40 | schmidts | for example i have this machine: |
| 07:08.42 | schmidts | rasterisk -x"core show uptime" |
| 07:08.42 | schmidts | System uptime: 4 weeks, 3 days, 8 hours, 20 minutes, 24 seconds |
| 07:08.50 | schmidts | root 376 0.2 2.6 724352 64580 ? Sl 2010 1527:49 \_ /usr/sbin/asterisk -f -v -p -U root -vvgt |
| 07:09.01 | schmidts | you can see its using 724mb of ram |
| 07:09.07 | schmidts | but this is with 1.6.2.18 |
| 07:09.46 | jacc0 | I have a script on local host sending an option package every 30 seconds to check if asterisk is still responding |
| 07:10.07 | schmidts | jacc0 :D ok thats like nothing ;) |
| 07:10.26 | jacc0 | and besides that about 2 call/sip packages an hour comming in from the trunk |
| 07:10.30 | schmidts | maybe you hit this performance problem with the hangup cause they discussed on the dev list lately |
| 07:10.39 | schmidts | ok thats still nothing |
| 07:10.47 | jacc0 | could be |
| 07:11.00 | jacc0 | can you point me to the discussion about it? |
| 07:11.25 | schmidts | the titel was : Asterisk 1.8 SIP_CAUSE performance regression |
| 07:11.27 | jacc0 | because I use originate, bridge and softhangup; it could cause a strang hanguop cause I guess |
| 07:11.34 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 07:11.46 | jacc0 | *strange |
| 07:12.52 | schmidts | but this could only have some impact on a heavy loaded system, not like yours |
| 07:13.00 | schmidts | how long does your asterisk runs? |
| 07:13.38 | jacc0 | 1 week, 22 hours, 58 minutes, 18 seconds |
| 07:13.44 | jacc0 | it's a new record!! |
| 07:13.48 | schmidts | :D |
| 07:13.55 | jacc0 | :P |
| 07:14.35 | schmidts | i am now looking at my only 1.8 server and this has a uptime for only 3 hours but allready use 213mb of ram and there was not a single call today |
| 07:14.47 | schmidts | but its a 1.8.3.2 and only used for meetme |
| 07:14.49 | *** join/#asterisk asterisk-Tester (~RAMYT@193.227.170.247) |
| 07:15.17 | jacc0 | 213 mb seems fair |
| 07:15.56 | schmidts | not really, cause this server has really nothing to do, only 2 sip peers with qualify and not really any calls |
| 07:17.00 | zkn | ehh..btw, ExecIf($["${DIALSTATUS}" = "CHANUNAVAIL" | "NOANSWER"]?foo:bar) worked just fine |
| 07:17.38 | schmidts | zkn are you sure? cause for me this looks like "NOANSWER" is allways true |
| 07:18.21 | zkn | what do you mean always true? |
| 07:18.51 | schmidts | jacc0 just to compare another heavy loaded system "only" has 634mb of ram, but 4055 sip peers on it, 50 concurrent calls and 150 sip channels |
| 07:19.35 | jacc0 | @zkn "NOANSWER" != 0 = true |
| 07:19.46 | schmidts | jacc0 thanks ;) |
| 07:20.19 | jacc0 | correct syntax should be : |
| 07:20.33 | *** join/#asterisk xnfinite (~xnfinite@62.82.191.170.static.user.ono.com) |
| 07:20.45 | jacc0 | ExecIf($["${DIALSTATUS}" = "CHANUNAVAIL" | "${DIALSTATUS}" = "NOANSWER"]?foo:bar) |
| 07:21.38 | jacc0 | @zkn: everything that is not 0 will return true |
| 07:22.25 | zkn | ok, i'll try and analyze also your proposition, jacc0 |
| 07:23.23 | jacc0 | execIF($["thiswillalwaysbetrue"]?foo:bar) |
| 07:25.15 | zkn | yep, jacc0 & schmidts, just tested it, you are both correct! |
| 07:26.35 | zkn | ExecIf($["${DIALSTATUS}" != "CHANUNAVAIL" | "NOANSWER"]?foo:bar) did not give be "bar" when DIALSTATUS in fact was NOANSWE |
| 07:26.40 | zkn | NOANSWER |
| 07:26.47 | zkn | me* |
| 07:30.02 | *** join/#asterisk irroot (~irroot@41.122.157.165) |
| 07:45.21 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:9556:1436:fdb:eb11) |
| 07:48.26 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 08:05.19 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 08:07.44 | jacc0 | Okay, I recompiled asterisk 1.8.6-rc1 with MALLOC_DEBUG compiler flag |
| 08:08.29 | jacc0 | asterisk is taking up 165mb ram |
| 08:08.37 | jacc0 | now we wait :) |
| 08:09.29 | schmidts | ok ;) |
| 08:11.25 | irroot | sitting in meeting at HP |
| 08:13.56 | ChannelZ | Tell them they can suck it |
| 08:14.15 | ChannelZ | Thanks a lot for burying webOS, jerks |
| 08:15.15 | jacc0 | and stick a big pole there where the sun never shines :P |
| 08:15.53 | jacc0 | (in the ground ofcource) |
| 08:16.50 | jacc0 | schmidts: any news from cisco? |
| 08:19.04 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-nehmiqtdpkvsdzdq) |
| 08:23.41 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-ewewsjdbxypqcmwu) |
| 08:29.28 | jacc0 | schmidts: after compiling with MALLOC_DEBUG "core show memory" is still not available |
| 08:31.57 | *** join/#asterisk singler (~singler@88-119-250-190.static.zebra.lt) |
| 08:35.11 | jacc0 | do I need to run it in valgrind for that? |
| 08:37.56 | jacc0 | I need to get back into the field |
| 08:37.58 | jacc0 | ttyal |
| 08:38.00 | jacc0 | bye |
| 08:38.18 | schmidts | jacc0 valgrind would sure be a good idea |
| 08:39.05 | irroot | ChannelZ the sales guys did not know they sold off the PC's / closing WebOS |
| 08:39.26 | ChannelZ | Well he'll figure it out when he gets his pink slip |
| 08:39.57 | irroot | lol |
| 08:40.09 | irroot | the email they got did not have all the details |
| 08:40.47 | irroot | they opening 12 new branches in africa this is why we talking to them using asterisk to mesh it all together and access the data |
| 08:43.29 | *** join/#asterisk Rico29 (~rico@sar-b123.olm.fr) |
| 08:43.32 | Rico29 | hello all |
| 08:43.40 | irroot | every time they mentioned "PC" asked if they will get one from dell |
| 08:43.40 | Chainsaw | Morning Rico. |
| 08:44.01 | Rico29 | I have a problem with "channel request hangup" which doesn't work |
| 08:44.10 | Rico29 | it doesn't hangup my channel |
| 08:44.13 | irroot | what it do ?? |
| 08:44.17 | Chainsaw | Does it give a reason? |
| 08:44.20 | Chainsaw | Asterisk generally does. |
| 08:44.23 | Rico29 | no |
| 08:44.25 | irroot | stay in channel list ?? |
| 08:44.28 | Rico29 | it just doesn(t hangup |
| 08:44.30 | Rico29 | yes |
| 08:44.33 | irroot | set verbose/debug |
| 08:44.41 | irroot | "show channel ....." |
| 08:45.00 | irroot | look if it is blocking |
| 08:45.12 | irroot | core show threads does it have a thread ?? |
| 08:45.18 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 08:45.51 | Rico29 | show channel ... works |
| 08:45.53 | irroot | seen this after a pickup thought i had nailed it down does seem better but not 100% |
| 08:47.03 | Rico29 | show threads shows me a lot of stuff |
| 08:48.39 | Rico29 | is this a bug ? |
| 08:49.46 | Rico29 | last message is "BYE" in sip show channels |
| 08:50.27 | ChannelZ | works here |
| 08:50.36 | ChannelZ | Oh, SIP channel.. lemme try that |
| 08:51.58 | irroot | if the channel in question is orpahned there is no way to hang it up and clear it |
| 08:51.58 | Rico29 | ?? |
| 08:53.08 | ChannelZ | hmm that worked too. Are you trying to hang up active channels or are you saying you have zombies you're trying to clear? |
| 08:53.33 | Rico29 | I don't know if it'sd a zombie |
| 08:53.45 | Rico29 | in hints, the phone appears "'in use" |
| 08:53.54 | *** join/#asterisk maxagaz (~maxagaz@220.231.37.106) |
| 08:54.00 | ChannelZ | and does it *never* work or it stops working after a time? |
| 08:54.05 | Rico29 | and what can I do if the channel is orphaned ? |
| 08:54.45 | Rico29 | it works with other channels |
| 08:54.53 | Rico29 | just this one |
| 08:55.03 | Rico29 | which doesn't want to Haraken ngup |
| 08:55.08 | Rico29 | tu hangup, sorry |
| 08:55.11 | Rico29 | to |
| 08:55.31 | ChannelZ | That's OK it sounds like fake cuss words the other way, it's fun |
| 08:56.00 | Rico29 | will Y have to restart asterisk ? |
| 08:56.04 | Rico29 | oops |
| 08:56.08 | Rico29 | will I have to restart asterisk ? |
| 08:57.04 | Rico29 | or is there another way to hangup this channel ? |
| 08:58.01 | ChannelZ | Dunno. |
| 08:58.25 | ChannelZ | Interesting, sip show channels actually shows 4 BYE messages to the same peer that aren't going away |
| 09:00.21 | Rico29 | i'm usign svn version |
| 09:01.03 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 09:05.42 | Rico29 | what can I do ? |
| 09:06.05 | schmidts | rico29 do you only have sip channels or also normal channels? |
| 09:06.14 | schmidts | channel request hangup could help maybe |
| 09:06.29 | Rico29 | no it doesn't work |
| 09:06.37 | Rico29 | alredy tried |
| 09:06.44 | Rico29 | i also have normal channels |
| 09:06.44 | *** join/#asterisk ickmund (~ickmund@cli-5b7e85f2.bcn.adamo.es) |
| 09:06.51 | schmidts | maybe its deadlocked |
| 09:07.02 | Rico29 | and channel request hangup works on normal channels |
| 09:07.27 | schmidts | i know ;) thats why i asked if you also have normal channels or only the sip channels |
| 09:07.36 | schmidts | could you show me the output of sip show sched |
| 09:07.49 | ChannelZ | Do you have physical access to the device? |
| 09:07.50 | Rico29 | SRV-IPOP1*CLI> sip show sched |
| 09:07.50 | Rico29 | <PROTECTED> |
| 09:07.50 | Rico29 | <PROTECTED> |
| 09:07.50 | Rico29 | <PROTECTED> |
| 09:07.50 | Rico29 | <PROTECTED> |
| 09:07.50 | Rico29 | <PROTECTED> |
| 09:07.52 | Rico29 | <PROTECTED> |
| 09:07.54 | Rico29 | <PROTECTED> |
| 09:07.56 | Rico29 | <PROTECTED> |
| 09:07.58 | Rico29 | <PROTECTED> |
| 09:08.00 | Rico29 | <PROTECTED> |
| 09:08.04 | Rico29 | <PROTECTED> |
| 09:08.06 | Rico29 | <PROTECTED> |
| 09:08.08 | ChannelZ | ~pb |
| 09:08.08 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.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. |
| 09:08.08 | Rico29 | which device ? the phone ? |
| 09:08.20 | ChannelZ | yes |
| 09:08.22 | Rico29 | sorry, will use pastebin next time |
| 09:08.30 | Rico29 | Y have remote access to it, yes |
| 09:08.39 | ChannelZ | reboot it :) |
| 09:08.41 | Rico29 | but I already rebooted it |
| 09:08.42 | Rico29 | :) |
| 09:09.06 | Rico29 | no change, phons comes "in use" when it registers on the server |
| 09:09.14 | schmidts | the output of sip sched seams normal, atleast you dont have a lot of retransmits in there |
| 09:09.29 | ChannelZ | so it's preventing you from calling it or otherwise using it? |
| 09:09.44 | Rico29 | twice |
| 09:09.58 | Rico29 | I can't do anything with the phone |
| 09:10.22 | ChannelZ | these are dialplan hints you're talking about? |
| 09:10.51 | Rico29 | yes |
| 09:11.29 | Rico29 | asterisk -rx "core show hints" | grep jarry > 101@blf_jarry : SIP/101.jarry State:InUse Watchers 2 |
| 09:11.35 | Rico29 | always in use |
| 09:13.15 | schmidts | Rico29 what do you see with sip show inuse? |
| 09:14.11 | Rico29 | http://pastebin.com/aVHAwJ5X |
| 09:14.14 | Rico29 | schmidts, http://pastebin.com/aVHAwJ5X |
| 09:15.56 | *** join/#asterisk usrbinfoobar (~none@89.201.163.12) |
| 09:15.57 | schmidts | looks strange |
| 09:16.08 | usrbinfoobar | hi guys |
| 09:16.26 | usrbinfoobar | is there any way to increase volume on a linear channel driver? or any other - in asterisk 1.4 |
| 09:19.06 | *** join/#asterisk orn (~orn@rtr1.sh23.sip.is) |
| 09:21.06 | Rico29 | schmidts, any idea ? |
| 09:21.32 | schmidts | Rico29 could you show me a sip show channels |
| 09:22.18 | Rico29 | http://pastebin.com/rTrMfzPy |
| 09:24.32 | schmidts | and also a sip show channel beac9e11f442b71 |
| 09:25.37 | Rico29 | schmidts, http://pastebin.com/9gZWyyq4 |
| 09:26.53 | schmidts | Rico29 which asterisk version are you using? |
| 09:27.00 | Rico29 | 1.8 svn |
| 09:27.11 | Rico29 | sterisk SVN-branch-1.8-r329529 |
| 09:27.11 | schmidts | ah ok, then please the output of sip show objects |
| 09:27.24 | schmidts | it looks like you have a dead dialog in there: |
| 09:27.27 | schmidts | <PROTECTED> |
| 09:27.33 | schmidts | Last Message: Rx: BYE |
| 09:27.49 | schmidts | if the last message was a bye, the dialog should have the need destroy flag set |
| 09:28.00 | Rico29 | yes |
| 09:29.04 | schmidts | i guess in the sip show objects you will see a dialog with a very high ref counter |
| 09:29.13 | Rico29 | schmidts, sip show objects > http://pastebin.com/qYhik4bR |
| 09:30.46 | schmidts | Rico29 atleast it looks ok, but i guess there is something which hold a lock for this dialog, or added a refcounter value but didnt removed it |
| 09:30.47 | schmidts | name: beac9e11f442b719 |
| 09:30.48 | schmidts | type: dialog |
| 09:30.48 | schmidts | objflags: 0 |
| 09:30.48 | schmidts | refcount: 2 |
| 09:31.35 | Rico29 | this one ? |
| 09:32.08 | Rico29 | and how can I resolve it ? |
| 09:32.15 | *** join/#asterisk ollii (~risker@2001:470:1f15:1384:62eb:69ff:fe31:b4) |
| 09:33.01 | ollii | hey..i have a simple question about dahdi_genconf (2.5) and its option parsing...could someone give me an example how to set options? in detail i would like to set another echocanceller in system.conf |
| 09:34.17 | *** join/#asterisk Katty (~Katty@mail.copi-rite.com) |
| 09:34.26 | Rico29 | mmh |
| 09:34.38 | Rico29 | and I have the same problem with another phone.... |
| 09:36.01 | Rico29 | which his refcount is 7 |
| 09:36.28 | Rico29 | is this an asterisk bug ? |
| 09:37.30 | kaldemar | ollii: have you noticed /etc/dahdi/genconf_parameters ? |
| 09:38.49 | schmidts | Rico29 it could be a problem with asterisk but also with the phones, for example if the other side sents a bye, asterisk response with an 403 unauthorized and the phone didnt sent the bye again with proper authentication this can also happen |
| 09:39.08 | schmidts | maybe you should try a sip debug to see if the phone does a normal hangup |
| 09:39.19 | Rico29 | yes |
| 09:39.28 | Rico29 | but it would do the same thing with each phone ? |
| 09:39.43 | ollii | yeah...but i dont know how dahdi_genconf parser works.... :X |
| 09:39.52 | Rico29 | we only had this problem twice, and have more than 40 phones like this... |
| 09:43.16 | ollii | thats why i need an example ;) |
| 09:43.28 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-vhloerajewzvhryp) |
| 09:43.44 | *** join/#asterisk irroot (~irroot@pbx.distrotech.co.za) |
| 09:48.43 | *** join/#asterisk iggy_work (~iggy@FW01.AS42383.NET) |
| 09:49.06 | iggy_work | Hi there. Does anyone know of a wireless SIP conference phone? |
| 09:50.05 | irroot | iggy_work get a decent one like a polycom 6000 and put a wifi bridge on it |
| 09:50.09 | Chainsaw | iggy_work: Normally I advise a DECT handset with a SIP base. But I'm not aware of DECT conference phones. |
| 09:50.38 | Chainsaw | iggy_work: But I support irroot's suggestion of a Polycom device. Wire it in if you can. WiFi latency is unpredictable. |
| 09:50.49 | iggy_work | Cool. thanks |
| 09:56.48 | *** join/#asterisk x1user (~x1user@host-212-75-8-69.bbccable.net) |
| 09:57.29 | x1user | [Aug 19 12:55:46] WARNING[12133]: res_config_mysql.c:326 realtime_mysql: MySQL RealTime: Invalid database specified: mya2billing (check res_mysql.conf) |
| 09:57.45 | x1user | but ihave mya2billing in res_mysql which is existing database |
| 09:58.25 | x1user | http://pastebin.com/hWm05Dt9 this is res_mysql.conf |
| 09:59.45 | irroot | x1user and this database can be accessed from 127.0.0.1 |
| 10:00.12 | irroot | ie not localhost the 2 are different |
| 10:00.15 | x1user | yes i am sure |
| 10:00.28 | x1user | different? |
| 10:00.40 | irroot | yes for access in mysql |
| 10:00.59 | irroot | you need to allow access from localhost and or 127.0.0.1 or all |
| 10:01.20 | irroot | its caught me out before not a heavy my user though |
| 10:02.04 | *** join/#asterisk bn-7bc (~bjarne-im@pdpc/supporter/active/bn-7bc) |
| 10:02.43 | x1user | Hm, you mean in the mysql? |
| 10:02.50 | x1user | it is not an asterisk issue |
| 10:02.59 | x1user | i've tried with localhost but it is the same |
| 10:22.27 | *** join/#asterisk jkroon (~jkroon@41-133-100-241.dsl.mweb.co.za) |
| 10:25.23 | x1user | [19/08/2011 13:15:53]:[file:a2billing.php - line:620 - uniqueid:1313748950.0]:[CallerID:1000]:[CN:]:[[AUTHENTICATION FAILED (cia_res:-1)]] |
| 10:25.33 | x1user | this is from a2billing_agi.log ? what does it mean |
| 10:27.25 | *** join/#asterisk bn-7bc (~bjarne-im@pdpc/supporter/active/bn-7bc) |
| 10:27.58 | *** part/#asterisk ollii (~risker@2001:470:1f15:1384:62eb:69ff:fe31:b4) |
| 10:34.21 | *** join/#asterisk otwieracz (~gonet9@v6.gen2.org) |
| 10:34.22 | otwieracz | Hey. |
| 10:34.29 | *** part/#asterisk chrisjunkie (~Chris@2001:4428:22d:2:208:2ff:fe7e:6312) |
| 10:34.48 | otwieracz | I have problem with Asterisk-gui â when I'm trying to add SIP trunk, I can't click âSaveâ. |
| 10:34.52 | otwieracz | Nothing happens. |
| 10:38.47 | *** join/#asterisk xnfinite (~xnfinite@62.82.191.170.static.user.ono.com) |
| 10:44.00 | otwieracz | I tried in both Opera and chromium, so that's not browser bug, I think. |
| 10:54.07 | *** join/#asterisk jkroon (~jkroon@197.168.18.169) |
| 11:38.20 | *** part/#asterisk james_zhu (~Administr@183.16.211.132) |
| 11:40.08 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 11:46.50 | *** join/#asterisk datalay (~datalay@unaffiliated/datalay) |
| 11:47.27 | datalay | what s digium wildcard? why wildcards are so expensive? clearly we use them for that? |
| 11:52.17 | kaldemar | telephony hardware tends to be expensive. |
| 11:56.36 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 11:59.51 | *** join/#asterisk Nasga (~Nasga@186.212.10.93.rev.sfr.net) |
| 12:17.01 | datalay | when i need wildcard kaldemar |
| 12:22.16 | WIMPy | Wehn you want to connect your Asterisk directly to the PSTN or want to connect legacy phones directly to Asterisk. There are other verndors to choose from. |
| 12:22.18 | kaldemar | datalay: if you connect to PSTN via PRI or for a channel bank. |
| 12:23.14 | datalay | what is PRI means that? |
| 12:23.18 | datalay | paralel communication? |
| 12:23.36 | datalay | what is the main advantage of wildcard |
| 12:23.45 | datalay | techincally |
| 12:24.11 | eduzimrs | when * runs this exten "exten => _0XX[7-9]XXXXXXX,1,ChanIsAvail(Khomp/*ggsm)" the * crashes and stop. WHY ? version 1.4.34 |
| 12:24.12 | kaldemar | as opposed to what? |
| 12:24.53 | WIMPy | datalay: That's going to be a lengthy one. Better try to tell us what you want to do. then we can tell you how you can do it. |
| 12:25.53 | datalay | WIMPy, i just wonder, what people do with wildcard technically, i wonder it because it s too expensive |
| 12:26.09 | *** join/#asterisk irroot (~irroot@197.169.213.73) |
| 12:26.10 | datalay | i don't know what is PRI |
| 12:26.20 | WIMPy | Connect phone lines, phones or their legacy PBX. |
| 12:26.45 | WIMPy | A digital phone line for 23 or 30 simultaneous calls. |
| 12:29.04 | SunTsu | datalay: do you know isdn? |
| 12:29.18 | kaldemar | eduzimrs: https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace |
| 12:30.03 | *** part/#asterisk asterisk-Tester (~RAMYT@193.227.170.247) |
| 12:30.05 | SunTsu | datalay: if so, PRI is sort of isdn with up to 30 speech channels |
| 12:30.17 | datalay | ISDN is alternative digital structure to analog phone system |
| 12:30.45 | WIMPy | A PRI is sometimes called ISDN30. |
| 12:31.34 | *** join/#asterisk merlin8282 (~merlin828@AStrasbourg-554-1-200-219.w83-203.abo.wanadoo.fr) |
| 12:32.31 | *** join/#asterisk djuhl30 (~quassel@121.135.85.226) |
| 12:32.50 | irroot | or E1 PRI |
| 12:33.32 | *** join/#asterisk coppice (~chatzilla@116.92.38.165) |
| 12:33.55 | SunTsu | nothing like a consistant terminology ;) |
| 12:36.07 | *** join/#asterisk billmania (~bill@38.98.130.98) |
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| 12:45.06 | merlin8282 | If I set __DYNAMIC_FEATURES to something, will it be set globally (for everyone), or only to the calling or called channel ? |
| 12:46.20 | eduzimrs | kaldemar this app ChanIsAvail is supported by this version? |
| 12:46.53 | *** join/#asterisk bratner_ (~bratner@94.159.138.32) |
| 12:47.06 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 12:48.01 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-hmllborlytynsjix) |
| 12:48.39 | eduzimrs | kaldemar do u recommend to always start * with dumpcore=yes option? |
| 12:50.37 | singler | there is no use of core dump if debuging compile options are not enabled, and if you compiled with debug options where is no use of them if you do not core dump :) |
| 12:52.37 | eduzimrs | singler ok, but do know something abou my issue? |
| 12:53.21 | singler | sorry, but no.. did you try to search bug reports? |
| 12:53.51 | eduzimrs | singler yeap, but nothing about i found. |
| 12:54.31 | eduzimrs | singler very weird cause this app is to usable. |
| 12:55.05 | eduzimrs | singler i`ve never met this problem before its just begin today |
| 12:57.45 | *** join/#asterisk bchia (~Adium@24.42.227.160) |
| 13:00.56 | kaldemar | eduzimrs: yes, it has been around since long before 1.4. however, no idea if Khomp supports what it does. or any idea on what Khomp is for that matter. |
| 13:06.00 | *** join/#asterisk serafie (~erin@nat/digium/x-yckubhjasapipxal) |
| 13:07.47 | *** join/#asterisk kemix (~kemic@75-144-79-242-Michigan.hfc.comcastbusiness.net) |
| 13:08.05 | merlin8282 | I have following configuration: http://pastebin.archlinux.fr/433611 . When the caller does *3, it works without any problem. But when the callee does *3, the caller is put into the conference, and immediately the conference closes. See http://pastebin.archlinux.fr/433612 . |
| 13:08.40 | merlin8282 | What's the problem here ? I can't find any error or debug message that would help me :/ |
| 13:10.51 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-irovedlzblrfeydu) |
| 13:12.00 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 13:13.05 | kaldemar | merlin8282: is it intentional to have "caller" instead of "both" in the applicationmap? |
| 13:16.08 | kemix | I'm having some hair pulling troubles with recorded phone calls becoming out of sync. The callee has a 5-10 second delay compared to the caller. I've google'd for two days now attempted various fixes (updating SoX and modifying globals_custom.conf) and it seemed to fix for a while. but upon a spot check the next day i found that it's still a problem. we recently made the switch from trixbox |
| 13:16.08 | kemix | 2.6.x to pbxiaf purple (asterisk 1.4 to 1.8 ... zaptel to dahdi as well). This out of sync issue was not present in trixbox (asterisk 1.4). Any ideas how to remedy this problem? ... for good.. |
| 13:18.58 | merlin8282 | kaldemar: thanks pointing that out, maybe it's the problem. Asterisk 1.6 doesn't seem to allow the keyword "both". |
| 13:20.23 | merlin8282 | kaldemar: and if I set "callee" the macro is executed on the callee, that's not what we want. We want to send the other party (not the party that dials *3) to the conference. |
| 13:25.33 | Katty | ATTENTION |
| 13:25.39 | Katty | IT IS FRIDAY |
| 13:25.49 | Katty | DO YOU KNOW WHERE YOUR BOOGIE PANTS ARE?! |
| 13:26.15 | merlin8282 | kaldemar: sorry, I'm wrong. I forgot the /<ActivatedBy> |
| 13:28.57 | merlin8282 | kaldemar: thank you for having spotted the problem ! |
| 13:34.38 | *** join/#asterisk nighty^ (~nighty@TOROON12-1279662182.sdsl.bell.ca) |
| 13:35.03 | *** join/#asterisk hobodave (~hobodave@pdpc/supporter/professional/hobodave) |
| 13:35.21 | anonymouz666 | Katty: what is good about friday? |
| 13:35.24 | anonymouz666 | :P |
| 13:36.25 | Katty | it just is! |
| 13:36.29 | Katty | hugs anonymouz666 |
| 13:37.29 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 13:39.30 | anonymouz666 | hugs back |
| 13:50.07 | *** part/#asterisk xnfinite (~xnfinite@62.82.191.170.static.user.ono.com) |
| 13:56.44 | *** join/#asterisk Sylnai (~andi@andimiller.net) |
| 13:57.51 | *** join/#asterisk saxa (~sasa@189.26.255.43) |
| 14:03.21 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 14:04.51 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 14:16.03 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:16.03 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:18.25 | *** join/#asterisk ollii (~risker@2001:470:1f15:1384:f27b:cbff:fe9d:f3e4) |
| 14:18.40 | merlin8282 | kaldemar: mmm, in fact, that does not solve the problem. I set "self/both" in features.conf, and when say 22 calls 10, and 10 presses *3, then 22 goes for milliseconds in the conf and immediately the conf gets closed, and 22 hung up. 10 is hung up as expected. |
| 14:20.07 | merlin8282 | Note that I also changed exten => h,n,GotoIf($[${CFCNT}=1]?:end) to exten => h,n,GotoIf($[ "${CFCNT}" <= "1" ]?:end) |
| 14:20.20 | merlin8282 | in # |
| 14:20.21 | merlin8282 | [dynamic-nway-conf] |
| 14:22.50 | *** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk) |
| 14:24.10 | *** join/#asterisk hobodave (~hobodave@pdpc/supporter/professional/hobodave) |
| 14:25.47 | *** join/#asterisk dzalewski (53f4d842@gateway/web/freenode/ip.83.244.216.66) |
| 14:26.22 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 14:28.56 | *** join/#asterisk bchia (~Adium@nat/digium/x-rstazwrqzzmtgpdr) |
| 14:30.01 | dzalewski | Hi, I got an asterisk box connected to a NTT isdn box in Japan with signaling primary-ntt |
| 14:30.13 | dzalewski | does dahdi support primary-ntt singnaling ? |
| 14:33.51 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:35.04 | *** join/#asterisk Rico29 (~rico@sar-b123.olm.fr) |
| 14:35.07 | Rico29 | hi again |
| 14:35.47 | Rico29 | i have trouble using g729 codec. I've purchased licences, and allowed only this codec on my phone and the sip peer "allow" section. |
| 14:35.47 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 14:36.24 | Rico29 | But when I try to call my voicemail, it tells me that no compatible codecs were found |
| 14:36.31 | Rico29 | licences are activated |
| 14:36.35 | Rico29 | and not used |
| 14:36.49 | Rico29 | why doesn't asterisk do the transcoding ? |
| 14:37.38 | Qwell | Have you confirmed that the codec is loaded, and active/registered? |
| 14:38.14 | Qwell | http://downloads.digium.com/pub/telephony/codec_g729/README |
| 14:39.20 | dzalewski | Qwell: Hi, have you ever configured dahdi with Japaneese NTT isdn box ? signaling is primary-ntt |
| 14:39.35 | dzalewski | Qwell: does dahdi even support NTT signaling ? |
| 14:39.38 | Rico29 | Qwell, yes it is |
| 14:41.01 | Rico29 | Qwell, http://pastebin.com/yCHEKu4G |
| 14:42.37 | Qwell | Rico29: Yeah that definitely looks right. Show us logs of a failed call? |
| 14:43.27 | Rico29 | [Aug 19 16:43:11] NOTICE[32333]: chan_sip.c:8895 process_sdp: No compatible codecs, not accepting this offer! |
| 14:43.51 | Rico29 | in my sip.conf general settings, I've not allowed g729 |
| 14:43.55 | Rico29 | should I do it ? |
| 14:44.25 | Rico29 | and on the phone, only the g729 is allowed (same thing for the sip peer) |
| 14:46.20 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 14:49.37 | Rico29 | Qwell ? any idea ? |
| 14:49.47 | Qwell | Rico29: Not without logs of the call, no. |
| 14:49.57 | Qwell | sip debug, etc |
| 14:56.04 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 14:57.44 | Rico29 | Qwell, there was a problem on the phone |
| 14:57.48 | Rico29 | it works fine now |
| 14:58.21 | *** join/#asterisk pbati (~fozpaulo@189.58.102.230) |
| 14:59.36 | p3nguin | I wish I had a croissant, some eggs, some ham, and some cheese. |
| 14:59.59 | pbati | Hello! Personally, I have a problem, very serious, |
| 14:59.59 | pbati | [08.19.2011 11:59:06] WARNING [1488]: chan_sip.c: 6213 sip_write: Asked to transmit frame type alaw, while native formats is 0x4 (ulaw) read / write = 0x8 (alaw) / 0x4 (ulaw) |
| 15:00.00 | pbati | Can someone help me? |
| 15:01.40 | _Corey_ | p3nguin: Too close to lunch where i am to be talking food... torturous |
| 15:01.44 | *** join/#asterisk StaRetji (~BigAll@80.93.240.171) |
| 15:01.50 | *** join/#asterisk binbash_ (~peter@server.digitog.nl) |
| 15:01.57 | espro | got an hour till catered lunch gets here :S |
| 15:01.58 | StaRetji | Folks, can I have # in extensions? |
| 15:02.09 | atheos | StaRetji yes |
| 15:02.09 | p3nguin | I haven't had any breakfast, and lunch is still a couple hours away. |
| 15:02.17 | StaRetji | exten => #,1,Background(IVR14) |
| 15:02.29 | p3nguin | staretji: yes |
| 15:03.04 | *** join/#asterisk bchia (~Adium@nat/digium/x-ltmjviurmvsjjbxu) |
| 15:03.20 | StaRetji | atheos: p3nguin thx, however this works exten => 2,1,Background(IVR-03) and this not: exten => #,1,Background(IVR14) |
| 15:03.43 | StaRetji | Invalid extension '#', but no rule 'i' in context 'ivr777777' |
| 15:03.47 | StaRetji | got it |
| 15:04.02 | p3nguin | Try [#],1,BackGround(IVR14) |
| 15:04.13 | StaRetji | there is no rule, but I don;'t see rules for other either |
| 15:04.22 | StaRetji | p3nguin: thx, will do it now |
| 15:04.33 | p3nguin | Everything that starts with "exten =>" is a rule. |
| 15:04.47 | p3nguin | Extensions are rules. |
| 15:05.06 | StaRetji | <PROTECTED> |
| 15:05.26 | p3nguin | Are you remembering to run dialplan reload after you save changes to extensions.conf? |
| 15:05.32 | StaRetji | yes |
| 15:05.35 | StaRetji | reload |
| 15:05.39 | StaRetji | exten => [#],1,BackGround(IVR14) |
| 15:05.47 | p3nguin | I said "dialplan reload" |
| 15:06.07 | *** join/#asterisk hobodave (~hobodave@pdpc/supporter/professional/hobodave) |
| 15:06.18 | StaRetji | yes |
| 15:06.20 | StaRetji | sorry |
| 15:06.31 | StaRetji | I did |
| 15:07.09 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:08.37 | Katty | ATTENTION #ASTERISK |
| 15:08.39 | Katty | IT IS FRIDAY |
| 15:08.48 | Katty | that is all. |
| 15:08.56 | p3nguin | dialplan show #@ivr777777 |
| 15:09.07 | p3nguin | [ Context 'ivr777777' created by 'pbx_config' ] '#' => 1. BackGround(IVR14) [pbx_config] |
| 15:10.04 | StaRetji | yes |
| 15:10.09 | StaRetji | exactly like that |
| 15:10.27 | p3nguin | Yours shows the same thing, yes? |
| 15:10.43 | StaRetji | exactly the same |
| 15:11.06 | StaRetji | OMG |
| 15:11.08 | StaRetji | it works now |
| 15:11.15 | StaRetji | wtf |
| 15:11.20 | p3nguin | What did you do different? |
| 15:11.26 | StaRetji | I didn't touch anything |
| 15:11.34 | StaRetji | except several desperate |
| 15:11.36 | p3nguin | Also, I only put # in my dialplan, not [#]. |
| 15:11.36 | StaRetji | dialplan reload |
| 15:11.44 | Qwell | and by several, do you mean 1? |
| 15:11.47 | StaRetji | yes, I return baqck |
| 15:11.49 | p3nguin | haha |
| 15:11.57 | StaRetji | no, I did dialplan reload |
| 15:11.58 | StaRetji | nothing |
| 15:12.02 | StaRetji | then I did again |
| 15:12.12 | StaRetji | and then third time |
| 15:12.14 | StaRetji | and it work |
| 15:12.21 | StaRetji | maybe network congestion |
| 15:12.24 | StaRetji | it is remote server |
| 15:12.37 | StaRetji | or maybe server overloaded, not sure |
| 15:12.47 | StaRetji | I only know that you guys are great :D |
| 15:12.51 | StaRetji | thx so much |
| 15:13.11 | Katty | Qwell |
| 15:13.25 | Qwell | Katty |
| 15:13.31 | Chainsaw | Oh boy. Friday! |
| 15:13.46 | StaRetji | p3nguin: thx man, I really appreciate your help |
| 15:13.48 | Qwell | Which seat will I take!? |
| 15:13.50 | Katty | hugs on Qwell |
| 15:13.54 | Qwell | is hugged on |
| 15:14.01 | Katty | hugs on Chainsaw |
| 15:14.09 | p3nguin | Maybe you have a huge dial plan and low system resources, so it took seven minutes to reload the dialplan. |
| 15:14.11 | Katty | perk up! |
| 15:14.16 | Chainsaw | Hey there Katty *hug* |
| 15:14.17 | Katty | i have been chewed out twice already today! |
| 15:14.22 | Qwell | yay! |
| 15:14.23 | Katty | but it's not gonna get me down! |
| 15:14.24 | Qwell | or not yay. |
| 15:14.29 | Qwell | probably not yay |
| 15:14.33 | Katty | not so much, no |
| 15:14.37 | Katty | but definately yay for friday |
| 15:14.37 | Qwell | Whyfor? |
| 15:15.09 | Katty | well the first chewing out was the office manager scolding me for 'falsifying my time sheet' |
| 15:15.16 | Qwell | O.o |
| 15:15.21 | Katty | i started working this morning before i got to the office. |
| 15:15.25 | x1user | Anyone who have experience with a2billing? |
| 15:15.27 | Katty | she didn't bother to ask, she just chewed me out |
| 15:15.54 | Katty | and the second chewing out was by a remote sales rep who wanted to know why his new leads were not in sugarcrm yet |
| 15:16.07 | Katty | the leads that no one told me about, or sent to me... |
| 15:16.18 | Katty | but hey, FRIDAY! woo! |
| 15:16.22 | Qwell | woo! |
| 15:18.24 | kaii | boogy time! |
| 15:18.54 | kaii | o/ |
| 15:22.34 | p3nguin | katty: You should know better than to work before getting to work. How dare you try to do something beneficial for the company or anyone else?! |
| 15:23.53 | Chainsaw | p3nguin: It skews their statistics if worker bee #77787 logs into their phone and switches into "READY" before any of the lines open. |
| 15:23.59 | Chainsaw | has gotten told off for this in a previous job |
| 15:25.01 | *** join/#asterisk hobodave_ (~hobodave@pdpc/supporter/professional/hobodave) |
| 15:31.43 | *** join/#asterisk sourcode (~code@ppp-58-11-95-85.revip2.asianet.co.th) |
| 15:42.03 | *** join/#asterisk hobodave_ (~hobodave@pdpc/supporter/professional/hobodave) |
| 15:49.13 | *** join/#asterisk cyril_sneer (~alex@chello080108126057.9.11.vie.surfer.at) |
| 15:49.51 | cyril_sneer | i have a problem with asterisk and libss7: SAM messages are quit as "unknown" |
| 15:50.15 | cyril_sneer | anyone familiar here ? |
| 15:52.36 | *** join/#asterisk Polysics (~luca@host157-65-dynamic.4-87-r.retail.telecomitalia.it) |
| 15:52.38 | Polysics | hello |
| 15:52.45 | cyril_sneer | hi |
| 15:53.01 | Polysics | to dial a SIP address, does my softphone need to be connected to a server? |
| 15:53.12 | Polysics | i need to dial a local address |
| 15:53.29 | Polysics | sip:foo@127.0.0.1 |
| 15:53.35 | p3nguin | That depends on the soft phone. |
| 15:53.37 | cyril_sneer | anyone must be the server which handles the session |
| 15:54.05 | Polysics | p3nguin, i am on linux, so i do need a client that can do that, do you know of any, please? |
| 15:54.13 | p3nguin | Some softphones are silly, and require active registration to SOMETHING even to dial a SIP URI. |
| 15:54.26 | p3nguin | Try ekiga. |
| 15:54.46 | *** join/#asterisk Jasnejac (kvirc@81.91.107.236) |
| 15:55.01 | p3nguin | I'm not sure if that's one of the silly ones or not, since I generally do not use soft phones. |
| 15:55.23 | p3nguin | When I do use one, I use twinkle in conjunction with Asterisk. |
| 15:55.42 | p3nguin | But twinkle needs qt3, and most people complain about that. |
| 15:56.40 | *** part/#asterisk ollii (~risker@2001:470:1f15:1384:f27b:cbff:fe9d:f3e4) |
| 15:58.54 | *** join/#asterisk drift- (ae3001a4@gateway/web/freenode/ip.174.48.1.164) |
| 16:00.03 | *** join/#asterisk penguin (penguin@cpe-74-77-221-5.buffalo.res.rr.com) |
| 16:02.08 | Polysics | does Twinkle dial without a register server? |
| 16:02.32 | penguin | Polysics: no u will need to register via SIP |
| 16:03.26 | Polysics | could i just install Asterisk locally? |
| 16:04.54 | cyril_sneer | of course |
| 16:05.43 | *** join/#asterisk gmcharlt (~gmchart`@pdpc/support/active/gmcharlt) |
| 16:06.21 | *** join/#asterisk webguynow (~webguynow@c-24-1-222-204.hsd1.il.comcast.net) |
| 16:07.19 | *** part/#asterisk merlin8282 (~merlin828@AStrasbourg-554-1-200-219.w83-203.abo.wanadoo.fr) |
| 16:07.46 | *** join/#asterisk retentiveboy (~pdugas@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 16:08.19 | cyril_sneer | you need a server which deals the sip-conversation |
| 16:08.36 | cyril_sneer | it could be a public, or for your LAN just a workstation |
| 16:08.41 | cyril_sneer | or maybe the same machine |
| 16:11.28 | *** join/#asterisk SparFux (~raoul@rl2-1-pt.tunnel.tserv5.lon1.ipv6.he.net) |
| 16:17.21 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 16:18.20 | Polysics | are there public SIP servers around? |
| 16:20.25 | p3nguin | ekiga |
| 16:20.52 | p3nguin | I don't know if twinkle needs to be registered to make calls or not. Let me try real quick. |
| 16:21.41 | cyril_sneer | ekiga.net i think there a free public ones |
| 16:21.55 | cyril_sneer | https://www.ekiga.net/ |
| 16:22.49 | p3nguin | It seems like twinkle might need the registration, because when I tried to make the call, it sent the call through the proxy which is configured. |
| 16:22.55 | p3nguin | I didn't remove the proxy values and try that way, though. |
| 16:24.30 | *** join/#asterisk navaismo (~navaismo@fixed-203-96-202.iusacell.net) |
| 16:37.51 | *** join/#asterisk hobodave (~hobodave@pdpc/supporter/professional/hobodave) |
| 16:51.31 | *** part/#asterisk sorressean (~tyler@tds-solutions.net) |
| 16:56.27 | *** join/#asterisk nuken (~evandro@200.189.57.190) |
| 16:58.17 | nuken | hi guys |
| 16:58.23 | cyril_sneer | hi guy |
| 16:58.29 | navaismo | hi |
| 16:58.33 | nuken | i have one question with my asterisk |
| 16:58.44 | cyril_sneer | okay? |
| 16:58.46 | nuken | i think anybody can help me ! |
| 16:59.22 | *** join/#asterisk war9407 (war@c-71-62-63-105.hsd1.va.comcast.net) |
| 16:59.33 | nuken | when I type some numer in my IP Phone |
| 17:00.08 | nuken | it takes to start calling |
| 17:00.20 | nuken | take about 10 seconds |
| 17:00.42 | nuken | there is some option where I can fix this ? |
| 17:00.57 | cyril_sneer | is this related to asterisk? |
| 17:01.10 | cyril_sneer | then you have to be a little more detailed |
| 17:01.10 | navaismo | I think is related to IPPHONE |
| 17:01.13 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 17:01.23 | nuken | yes asterisk 1.8 |
| 17:01.29 | nuken | for example |
| 17:01.38 | cyril_sneer | what shows the asterisk cli? |
| 17:01.39 | nuken | when I pickup with *8 |
| 17:02.02 | nuken | it take 5 seconds to give me the call |
| 17:02.14 | cyril_sneer | what shows the asterisk cli at this time? |
| 17:02.21 | nuken | nothing |
| 17:02.34 | cyril_sneer | using SIP? |
| 17:02.36 | navaismo | if you dial a number and terminate the number with #, call inmedialty? |
| 17:03.07 | Qwell | nuken: The phone has a dialplan that controls those timeouts. |
| 17:03.13 | nuken | Yes navaismo |
| 17:03.14 | Qwell | Read the manual for your phone. |
| 17:03.30 | nuken | when I terminate with # it's immediatky |
| 17:04.03 | navaismo | Yep as Qwell says its related to the dialplan of the phone |
| 17:06.24 | *** join/#asterisk kemix (~kemic@75-144-79-242-Michigan.hfc.comcastbusiness.net) |
| 17:06.49 | nuken | huum ok... |
| 17:06.59 | nuken | i will see on my IP Phone |
| 17:07.08 | nuken | thanks for help ! |
| 17:13.36 | *** join/#asterisk luckyaba (~Lucky@ip72-194-218-169.sb.sd.cox.net) |
| 17:14.49 | *** join/#asterisk hobodave (~hobodave@pdpc/supporter/professional/hobodave) |
| 17:19.54 | Katty | hi |
| 17:20.35 | navaismo | hi |
| 17:26.12 | cyril_sneer | hi |
| 17:26.51 | p3nguin | hi |
| 17:27.08 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 17:30.59 | *** join/#asterisk fenlander (~fenlander@82.152.81.57) |
| 17:31.15 | *** join/#asterisk abw1oim (~winkler@p5DE8CD8B.dip.t-dialin.net) |
| 17:34.07 | *** join/#asterisk fullstop (~fullstop@173.210.91.4) |
| 17:34.12 | fullstop | happy friday everyone |
| 17:36.54 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 17:42.43 | *** join/#asterisk treborsux (~treborsux@75-144-117-117-Jacksonville.hfc.comcastbusiness.net) |
| 17:43.05 | Qwell | what's a trebor? |
| 17:43.20 | treborsux | lol |
| 17:43.32 | treborsux | I used to work for Sit Tech |
| 17:43.44 | *** join/#asterisk kemix (~kemic-fn@2001:470:1f10:11ec::2) |
| 17:43.48 | p3nguin | I'd guess it's similar to a trevor, but in a different nationality. |
| 17:44.23 | treborsux | I did the beta testing for bane of the cosmic fordge and crusaders of the dark savant |
| 17:44.39 | Qwell | I see. |
| 17:44.41 | Qwell | so, what's a trebor? |
| 17:44.43 | treborsux | Wizardry it was first role playing graphiocal game for PC |
| 17:45.12 | treborsux | characters name in wizardry 1 was treborsux to make fun of the writer of the games partner robert |
| 17:45.24 | treborsux | robert backwards |
| 17:45.34 | treborsux | and sux is a bit self explanatory |
| 17:47.08 | Qwell | So basically, a 30-year old troll? |
| 17:47.32 | Qwell | That's quite a grudge indeed. |
| 17:47.42 | *** part/#asterisk wizbit (wiz@newelite1.bshellz.net) |
| 17:50.26 | coppice | trebor mints are a minty bit stronger |
| 17:50.39 | Qwell | Are they stronger than dailing minties? |
| 17:51.21 | coppice | dunno. I prefer sharp's extra strong mints |
| 17:51.32 | Qwell | I think you missed the joke. That's okay. |
| 17:52.40 | eduzimrs | anyone can help with overflow calls? |
| 17:54.59 | navaismo | can you explain us more? |
| 17:55.00 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 17:57.03 | *** join/#asterisk irroot (~irroot@197.169.21.98) |
| 17:57.19 | acidfoo | Qwell, Is your real name is North ? |
| 17:57.20 | *** join/#asterisk serafie (~erin@nat/digium/x-ukergvorvoikpnkb) |
| 17:57.28 | acidfoo | or its just an alias |
| 17:57.34 | acidfoo | I was wondering ;) |
| 17:58.12 | eduzimrs | yeap, a i have a GSM interface into my * server, it has 4 chips of the mobile operator. When the 4 lines are busy, the fifth one is auto dropped by *. Thats my call overflow |
| 18:01.37 | eduzimrs | navaismo: copy? |
| 18:02.00 | navaismo | copy |
| 18:02.05 | navaismo | and you try to do |
| 18:03.32 | Qwell | That's not Asterisk dropping it. I doubt Asterisk ever sees it. |
| 18:03.48 | Qwell | hell, the card never sees it |
| 18:04.15 | Qwell | It may not even get to your tower. |
| 18:04.19 | navaismo | how looks your dialplan |
| 18:04.23 | navaismo | for that calls |
| 18:04.42 | eduzimrs | ill paste |
| 18:04.53 | *** part/#asterisk StaRetji (~BigAll@80.93.240.171) |
| 18:05.10 | Qwell | Or just ignore me. Either way. |
| 18:06.21 | eduzimrs | http://pastebin.com/5pRu2P4J |
| 18:06.42 | eduzimrs | @Qwell sry, i didnt understand |
| 18:08.56 | navaismo | I dont understand too |
| 18:09.55 | atheos | eduzimrs , so you have 4 lines, when you use 4 lines you can't make calls. Am I getting this right? |
| 18:10.13 | eduzimrs | atheos YEAP, thats it |
| 18:10.17 | Qwell | atheos: He said "into my * server", so presumably he meant incoming |
| 18:10.51 | atheos | either way, the problem is the same. you can't make more calls than the hardware supports. |
| 18:10.51 | navaismo | @Qwell aaa ok know i got it |
| 18:11.03 | eduzimrs | @Qwell i mean the device is physicaly into the server that runs * |
| 18:11.26 | eduzimrs | atheos yeap, but i have a E1 trunk too |
| 18:11.42 | navaismo | is the second dial in your dialplan? |
| 18:11.53 | atheos | eduzimrs - then you need that included in your dial plan, after line 5 |
| 18:11.57 | eduzimrs | YES, but doesnt work |
| 18:12.05 | atheos | oh, nv |
| 18:12.36 | atheos | eduzimrs - so, your problem isn't with your GSM hardware, it's with your E1 trunk |
| 18:13.36 | atheos | eduzimrs if you omit line 4, can you make outgoing calls? |
| 18:13.59 | eduzimrs | yes |
| 18:14.00 | eduzimrs | sure |
| 18:14.36 | eduzimrs | but like i said before, its just when i have more than 4 calls in GSM trunk |
| 18:14.47 | atheos | eduzimrs what do your logs show on these dropped calls? |
| 18:15.07 | eduzimrs | "CHANNEL CONGESTION, BUSY HERE" |
| 18:15.07 | navaismo | when you have 4 calls and calls what is the return status for that trunk, i mean what see en the cli? |
| 18:15.15 | navaismo | ok got it |
| 18:15.36 | eduzimrs | func "ChanIsAvail" can sove? |
| 18:16.04 | navaismo | do you try with gotoif and dialstatus? |
| 18:16.14 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 18:16.15 | eduzimrs | no |
| 18:16.30 | atheos | eduzimrs - probably, give it a go. |
| 18:17.23 | eduzimrs | atheos: resuming im trying to create a redundance between the two trunks |
| 18:17.56 | eduzimrs | atheos: if the 4 GSM are busy jump to E1 |
| 18:18.21 | p3nguin | eduzimrs: ChanIsAvail() is an application, not a function. |
| 18:18.55 | atheos | eduzimrs - yea, I do the same if my PRI is busy (fall back on SIP). not sure why your call doesn't travel through to the next Dial though. |
| 18:20.21 | Katty | ahoy me hearties! |
| 18:20.29 | navaismo | yep ist weird |
| 18:20.36 | navaismo | its* |
| 18:21.02 | eduzimrs | too weird |
| 18:21.05 | *** join/#asterisk mtbf (~in9836@89-76-9-252.dynamic.chello.pl) |
| 18:21.20 | navaismo | mmm try adding the Noop(${DIALSTATUS}) an if congestion send to the other trunk |
| 18:21.59 | eduzimrs | ok |
| 18:24.13 | atheos | eduzimrs - you could also define those outbound calls to a group, example: Set(GROUP()=outbound), then before you dial the GSM, evaluate GROUP_COUNT(outbound). move any number over 4 to the E1 |
| 18:28.43 | *** join/#asterisk kaushal (~kaushal@14.99.147.119) |
| 18:28.50 | kaushal | p3nguin: hi |
| 18:29.09 | p3nguin | hello |
| 18:29.30 | eduzimrs | atheos ok i´ll try tks |
| 18:30.21 | *** join/#asterisk atheos_ (~atheos@208.119.68.16) |
| 18:30.44 | *** join/#asterisk michael-i (~anonymous@204.11.230.58.static.etheric.net) |
| 18:31.51 | *** join/#asterisk lcat (~lcat@187.45.254.117) |
| 18:32.30 | *** join/#asterisk niekie_ (~niek@CAcert/Assurer/niekie) |
| 18:33.22 | michael-i | Having trouble setting up a hint on a ConfBridge in 10 beta. Googling "confbridge hint" just spits back results that it may or may not be broken. Anyone have this working? |
| 18:34.17 | Qwell | michael-i: Do you know how to setup hints for apps? It should be the same for confbridge as it was for meetme. |
| 18:34.50 | Qwell | exten => 1234,hint,ConfBridge:1234 |
| 18:34.53 | Qwell | basically |
| 18:34.58 | michael-i | Qwell: in the same context as where the conf extensions are defined, I have "exten => 501,hint,ConfBridge:501" |
| 18:35.14 | Qwell | core show hints |
| 18:35.20 | michael-i | shows idle every time... |
| 18:36.09 | *** join/#asterisk Aut0ExeC (~Jack@24.244.156.75) |
| 18:36.19 | Aut0ExeC | hi gys.. what version has google voice? |
| 18:36.24 | Aut0ExeC | or google talk rather |
| 18:36.35 | Aut0ExeC | 1.8 right? |
| 18:36.37 | atheos_ | Aut0ExeC 1.8+ |
| 18:36.40 | Aut0ExeC | ok ok |
| 18:36.40 | Aut0ExeC | sux |
| 18:36.52 | Aut0ExeC | deb repos is only at 1.6 |
| 18:37.04 | Aut0ExeC | which means compile source |
| 18:37.05 | Qwell | michael-i: try making it lowercase. confbridge:501 |
| 18:37.07 | atheos_ | apt-get install build-essential is all you need |
| 18:37.08 | Aut0ExeC | booooo |
| 18:37.15 | Aut0ExeC | yeah i know |
| 18:37.19 | Aut0ExeC | i built before |
| 18:37.23 | Aut0ExeC | hated it |
| 18:37.37 | Aut0ExeC | too many dependencies that i'm not sure of |
| 18:37.46 | Aut0ExeC | end up over installing libs |
| 18:37.52 | Aut0ExeC | just to maek sure i cover it |
| 18:38.28 | atheos_ | Aut0ExeC - I just keep a handy list of apt-get dependencies on my wiki. makes installing asterisk on debian a breeze |
| 18:38.33 | eduzimrs | atheos can u paste to me how should be this? |
| 18:38.51 | Aut0ExeC | atheos_: where is ur wiki? |
| 18:38.53 | michael-i | Qwell: Worked! Thanks :) |
| 18:39.01 | Aut0ExeC | atheos: ur wiki is where? |
| 18:39.03 | Qwell | excellent. though I would consider that to be a bug. |
| 18:39.22 | atheos | Aut0ExeC - it's private, but I'll post the relevant portion, just a sec. |
| 18:39.28 | Aut0ExeC | ok |
| 18:39.33 | michael-i | if the old syntax followed case and this doesn'tâ¦yeah |
| 18:39.35 | Aut0ExeC | thanks brother |
| 18:39.38 | Qwell | michael-i: Would you mind opening an issue on issues.asterisk.org/jira/ ? It should be case insensitive. |
| 18:40.11 | michael-i | Qwell: I can't connect to it today for some reason. I had to browse through cached versions of it on google |
| 18:40.16 | Qwell | weird |
| 18:40.17 | michael-i | As soon as I can though, I will open one. |
| 18:40.22 | Qwell | thanks |
| 18:40.26 | atheos | Aut0ExeC http://pastebin.com/TXviKcry |
| 18:40.43 | atheos | eduzimrs just a sec |
| 18:41.05 | Aut0ExeC | atheos: omg dude awesomeness |
| 18:41.07 | Aut0ExeC | thanx |
| 18:41.14 | atheos | np |
| 18:41.18 | michael-i | I might open another issue on there about multiple Bridge and Unlink events firing upon DTMF presses. Can't find any reason for them to show up like that. |
| 18:41.40 | Aut0ExeC | can i bookmark pastbins? how long are they current? |
| 18:42.20 | Aut0ExeC | oh this one says expires never |
| 18:42.21 | *** join/#asterisk pzn (~pzn@pdpc/supporter/active/pzn) |
| 18:42.22 | Aut0ExeC | thanks |
| 18:42.33 | *** part/#asterisk abw1oim (~winkler@p5DE8CD8B.dip.t-dialin.net) |
| 18:42.47 | *** join/#asterisk weinerk (~user@unaffiliated/weinerk) |
| 18:43.05 | p3nguin | apt-get install bison ncurses-dev libssl-dev libnewt-dev zlib1g-dev inetutils-inetd cvs procps sox libsox-fmt-gsm ffmpeg libxml2 libxml2-dev openbsd-inetd tftpd |
| 18:43.41 | Aut0ExeC | p3nguin: this is for me? |
| 18:43.50 | navaismo | too late? |
| 18:43.53 | Aut0ExeC | yes |
| 18:44.10 | p3nguin | I'm just pointing out that you need only one single install command for that many packages. |
| 18:44.21 | Aut0ExeC | p3nguin: oh yes |
| 18:44.23 | Aut0ExeC | of course |
| 18:44.53 | Aut0ExeC | I think he did it that way to be neat |
| 18:44.58 | *** join/#asterisk jks (jks@193.189.93.254) |
| 18:45.01 | atheos | the reason I do multiple, is if for someone reason one of them is not in the apt repository I'm attached to, it doesn't fail all of them. |
| 18:45.15 | Aut0ExeC | or that |
| 18:45.17 | Aut0ExeC | :) |
| 18:45.49 | *** join/#asterisk Dibbler (~Dibbler@87-194-103-72.bethere.co.uk) |
| 18:48.04 | Aut0ExeC | thanks guys |
| 18:48.09 | atheos | eduzimrs try this http://pastebin.com/bRiT7SyH it's completely untested, and might not work. good luck. |
| 18:48.43 | *** part/#asterisk Aut0ExeC (~Jack@24.244.156.75) |
| 18:49.37 | eduzimrs | atheos: thanks i`ll put it on testing |
| 18:52.16 | *** join/#asterisk rdahlin_1 (~rdahlin_1@2001:16d8:cc97:1:c8d1:2ad8:666d:f55b) |
| 18:52.49 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 19:01.43 | *** join/#asterisk jdoe (~jdoe@falseprophet.ca) |
| 19:05.13 | Katty | hi |
| 19:05.27 | Katty | my asterisk does not make calls what is wrong with it??? |
| 19:06.02 | p3nguin | Is there a way with dahdi to use all configured groups rather than defining a single group in the Dial()? |
| 19:06.31 | Katty | mmmm |
| 19:06.37 | Katty | i seem to recall a way |
| 19:06.39 | Katty | uno moment |
| 19:07.09 | p3nguin | Dial(Dahdi/g0/${EXTEN}) would limit me to just group 0. I would need to allow all 7 groups there in case I have more calls than channels in a single group. |
| 19:07.31 | p3nguin | Maybe Dahdi/g0-7/${EXTEN} ? |
| 19:08.05 | p3nguin | err, all 8 groups I meant. 0-7 |
| 19:10.14 | *** join/#asterisk corretico (~luis@200.12.40.17) |
| 19:11.10 | Katty | mmm here we go |
| 19:11.19 | Katty | i was using a macro for failover |
| 19:13.54 | p3nguin | I guess it wouldn't be too terribly hard to write a counter and increase the group number accordingly. |
| 19:13.59 | Katty | however i believe dahdi chans can belong to multiple groups |
| 19:13.59 | kaushal | channel originate DAHDI/g0-7/xxxxxxxxxx Application MP3Player /home/kaushal/obd-demo.mp3 |
| 19:14.15 | kaushal | it just called only once :( |
| 19:14.30 | p3nguin | kaushal: What? |
| 19:14.33 | Katty | you could add a new group number, which contains all the groups |
| 19:14.44 | Katty | like..uhh.. 63 |
| 19:14.52 | Katty | <PROTECTED> |
| 19:15.13 | Katty | err stick all of your channels in a new group |
| 19:15.22 | Katty | well not new, just...another...one |
| 19:15.24 | p3nguin | Can I include groups into that group, or would I need to use individual channels in that new group? |
| 19:15.33 | kaushal | Katty: can you please pastebin chan_dahdi.conf ? |
| 19:15.46 | carrar | I head g63 takes you straight to a porn line |
| 19:15.57 | carrar | heard |
| 19:16.25 | Katty | mmm, no i don't think you can do groups within groups |
| 19:16.31 | p3nguin | Freud strikes again. |
| 19:16.35 | Katty | BUT |
| 19:16.39 | Katty | you could do group = 63 |
| 19:16.43 | Katty | channels => 1-23 |
| 19:16.52 | carrar | this is hot |
| 19:18.01 | Katty | this is all theoretical of course |
| 19:18.17 | p3nguin | group = 63 and channels => 1-15,17-31,32-46,48-62 etc. |
| 19:18.43 | Katty | there ya go |
| 19:19.03 | Katty | but i wouldn't do 17-31,32-46 |
| 19:19.07 | Katty | that's far too much typing |
| 19:19.21 | Katty | unless you typoed and meant 42 |
| 19:19.48 | p3nguin | Some channels are being skipped... I was just copying the channels from the other groups as an example. |
| 19:20.08 | Katty | there's no point in doing 31,32 tho |
| 19:20.12 | Katty | you'd just do 17-46 |
| 19:20.27 | Katty | not saying it won't work, of course |
| 19:20.31 | p3nguin | Like I said, I was just copying from existing groups. Current group1 is 1-15,17-31. Current group2 is 32-46,48-62. |
| 19:20.35 | carrar | BUT you could if you wanted too!! |
| 19:20.49 | Katty | YES! |
| 19:20.58 | p3nguin | Had I been paying attention, I could have used 17-46. |
| 19:21.06 | Katty | attention is not required on fridays |
| 19:21.16 | p3nguin | But the current groups are split, and what's where I copied the channels from. |
| 19:22.39 | Katty | carrar: are you the very model of a modern major general?! |
| 19:22.44 | p3nguin | kaushal: I guess you can try it this way. |
| 19:22.52 | carrar | Yes I am |
| 19:23.03 | Katty | carrar: do you have information animal vegetable and mineral?! |
| 19:23.05 | carrar | I'm rode my Harley today, instead of my "Horse" |
| 19:23.15 | Katty | like a boss? |
| 19:23.23 | carrar | I owned the road |
| 19:24.05 | carrar | Parking is much cheaper in Seattle on a motorcycle |
| 19:24.51 | carrar | I ate a animal for lunch today |
| 19:25.06 | carrar | was yummie |
| 19:25.22 | Katty | what sort of animal |
| 19:25.25 | p3nguin | group = 63 and channels => 1-15,17-46,48-77,79-108,110-139,141-170,172-201,203-232,234-248 |
| 19:25.26 | Katty | and did you eat the whole animal? |
| 19:25.32 | carrar | BBQ PIG |
| 19:25.38 | Katty | oink. |
| 19:25.41 | carrar | no, just a small portion of it |
| 19:26.41 | carrar | (Peco's BBQ) one of the best BBQ places in seattle |
| 19:26.58 | Katty | om nom nom? |
| 19:27.08 | Katty | i'm not good at bbqing things. |
| 19:27.12 | Katty | perhaps i should fix that. |
| 19:27.21 | carrar | across the street from the HQ of Starbucks |
| 19:28.08 | p3nguin | I'm good at BBQing, but I don't have a pit or a smoker right now. I'm thinking of making a smoker soon. |
| 19:28.27 | carrar | I grill at home |
| 19:28.35 | p3nguin | me too! |
| 19:29.01 | carrar | BBQ takes too long and my propain grill doesn't go that low on heat settings |
| 19:30.36 | kaushal | Katty: hi |
| 19:30.57 | Katty | ohai |
| 19:31.06 | kaushal | http://pastebin.ubuntu.com/670303/ |
| 19:31.14 | kaushal | group = 63 and channels => 1-15,17-46,48-77,79-108,110-139,141-170,172-201,203-232,234-248 |
| 19:31.19 | kaushal | since it has 8 PRI ports |
| 19:31.22 | kaushal | how would it fit there ? |
| 19:31.30 | Katty | that's a lot of channels. |
| 19:32.17 | carrar | Should use NFAS |
| 19:32.20 | Katty | err, i would put it at the bottom |
| 19:32.42 | kaushal | carrar: NFAS ? |
| 19:33.22 | carrar | Non-Facilities Associated Signaling |
| 19:33.25 | p3nguin | katty: The problem is that the channels are configured into groups and defined per port. |
| 19:36.39 | Katty | that's ok |
| 19:36.42 | Katty | just make a new one |
| 19:37.21 | Katty | you shouldn't need to specify ports |
| 19:37.28 | p3nguin | So the ports don't need any specific configuration? |
| 19:37.34 | p3nguin | Just base it on the channels? |
| 19:37.48 | Katty | yep |
| 19:37.51 | Katty | just ignore the port bits |
| 19:37.53 | p3nguin | oki |
| 19:38.46 | *** join/#asterisk rdahlin_1 (~rdahlin_1@78-73-19-238-no168.tbcn.telia.com) |
| 19:38.53 | p3nguin | http://pastebin.com/MHrusnJ2 |
| 19:39.00 | Katty | you need all the other bits tho. signaling, context and stuffs |
| 19:39.23 | carrar | And with that excellent piece of advice ignored I will go for a ride. |
| 19:40.08 | p3nguin | You're suggesting that I ignored the advice? |
| 19:40.23 | carrar | unless you are in china |
| 19:40.25 | kaushal | Katty: can you please comment signaling, context and stuffs ? |
| 19:40.40 | carrar | 8 PRI's |
| 19:40.47 | p3nguin | kaushal: I already did it. Look at the pastebin. |
| 19:40.52 | kaushal | yes |
| 19:40.58 | carrar | gain back some B channels from all those un needed D channels |
| 19:40.59 | kaushal | i did looked at it |
| 19:41.05 | _Corey_ | p3nguin: You're trying to do one big group and sub-groups? |
| 19:41.09 | kaushal | please comment signaling, context and stuffs ? |
| 19:41.20 | Katty | err..what? |
| 19:41.23 | _Corey_ | or am I reading wrong? |
| 19:41.45 | kaushal | < Katty> you need all the other bits tho. signaling, context and stuffs |
| 19:41.45 | p3nguin | _corey_: kaushal has a ton of channels, which are broken into 8 groups. He wants to dial out from ALL channels. |
| 19:42.03 | p3nguin | kaushal: Stop bugging her about it and look at what I pasted. |
| 19:42.08 | _Corey_ | ah, I skimmed back but apparently not far enough... |
| 19:42.09 | Katty | could someone please repeat the question |
| 19:42.11 | Katty | i are confuzzled. |
| 19:42.22 | kaushal | p3nguin: sorry |
| 19:42.25 | Katty | i believe NFAS dumps you into the t1 entirely |
| 19:42.39 | Katty | but i don't really know how to do it |
| 19:42.59 | _Corey_ | I was going to suggest using the group=x,y option, as it looks like it hadn't been mentioned. I may have missed it though. |
| 19:43.09 | Katty | eh it's a lot of work |
| 19:43.12 | p3nguin | carrar: Just so you know, it's not me who needs to be taking the advice. kaushal isn't paying me to read, understand, and implement this for him. |
| 19:43.16 | Katty | easier just to make a new group, and put the specific channels into it |
| 19:43.27 | carrar | He isn't!!!?? |
| 19:43.30 | carrar | What a looser |
| 19:43.41 | p3nguin | I was just trying to be helpful and get him some assistance. |
| 19:43.44 | Katty | i accept payment in form of cookies. |
| 19:43.52 | Katty | oh and cupcakes. |
| 19:43.54 | Katty | also rollerblades. |
| 19:43.56 | carrar | wine? |
| 19:43.59 | Katty | eww, no |
| 19:44.02 | Katty | wait |
| 19:44.04 | Katty | boones farm? |
| 19:44.04 | p3nguin | cupcak |
| 19:44.09 | carrar | heh |
| 19:44.23 | p3nguin | Mmmmm... Wild Island |
| 19:44.25 | Katty | what's wrong with cupcakes |
| 19:44.32 | Katty | ARE YOU QUESTIONING THE AWESOMENESS OF CUPCAKES |
| 19:44.59 | chuckf | why are cupcakes suddenly being sold everywhere? |
| 19:45.04 | _Corey_ | Cupcakes are currency in many places |
| 19:45.06 | p3nguin | kaushal: carrar is suggesting that you use another method. I guess you need to go learn about NFAS. |
| 19:45.22 | carrar | NFAS will gain you more B channels |
| 19:45.32 | Katty | 1 more, per pr1 |
| 19:45.37 | Katty | but i don't think he wants to use the entire pri |
| 19:45.40 | Katty | just specific channely bits |
| 19:45.43 | *** join/#asterisk kerx (~kerx@76-240-161-60.lightspeed.irvnca.sbcglobal.net) |
| 19:45.44 | carrar | why |
| 19:45.47 | _Corey_ | <opinion>NFAS is not worth the trouble</opinion> |
| 19:45.47 | Katty | idk |
| 19:45.51 | carrar | heh |
| 19:45.58 | Katty | i do not know all. |
| 19:46.00 | Katty | just...eat...all |
| 19:46.02 | p3nguin | As long as he can dial out one single large group of channels, I doubt he cares how it is accomplished. |
| 19:46.03 | Katty | kind of. |
| 19:46.35 | carrar | Dial(dahdi/g1/8675309) |
| 19:46.43 | Katty | now i want to eat cupcakes. |
| 19:46.45 | p3nguin | My other idea was to create a counter and roll over to the next group when the channels are reached on the current one. |
| 19:46.57 | kemix | I have every extension setup to record phone calls, the problem is (i've narrowed it down a bit since i first stated the problem in here) recorded incoming caller audio has a 5 second delay so when the out and in files are mixed together, the in (caller audio) is 5 seconds out of sync with the out, this makes recorded incoming phone calls difficult to understand. any ideas? (Asterisk 1.8.5.0 |
| 19:46.57 | kemix | PBXIAF Purple) |
| 19:47.08 | carrar | let groups take care of rolling over |
| 19:47.20 | p3nguin | That happens all by itself? |
| 19:47.27 | Katty | yes'r |
| 19:47.39 | carrar | What she said |
| 19:47.51 | p3nguin | If, for example, g0 is 30 channels, and you dial out g0, aren't you limited to 30 calls? |
| 19:47.51 | Katty | kemix: are you manually mixing? |
| 19:48.09 | *** join/#asterisk xpot-mobile (~james@166-70-100-198.ip.xmission.com) |
| 19:48.19 | kemix | Katty: i dont think so :o |
| 19:48.34 | Qwell | Answers like that are why we don't support FreePBX users here. |
| 19:48.36 | Katty | post your mixey recordey line |
| 19:48.48 | p3nguin | and PiaF sux |
| 19:48.53 | Katty | yes that IS the technical term for it too, thank you |
| 19:48.57 | Katty | Qwell: CUPCAKES |
| 19:49.04 | _Corey_ | p3nguin: Yeah, definitely on the 30 channel thing unless other channels are also in g0. |
| 19:49.09 | Qwell | Katty: You're sending me CUPCAKES?! |
| 19:49.11 | carrar | if for example g1 is 8 PRI's NFS, you can hang ~183 concurrent calls to g1 |
| 19:49.18 | carrar | NFAS |
| 19:49.21 | p3nguin | _corey_: Channels are in groups 0 through 7. |
| 19:49.23 | carrar | have |
| 19:49.26 | carrar | oh my my spelling is horried |
| 19:49.32 | Katty | you mean |
| 19:49.35 | Katty | 192 |
| 19:49.44 | carrar | phear my math |
| 19:49.49 | Katty | err, can you use the dchan on nfas |
| 19:49.51 | carrar | 191 |
| 19:49.53 | Katty | you can yes? |
| 19:49.58 | Katty | cept for1 |
| 19:49.58 | carrar | 1 D channel for all 8 |
| 19:50.01 | Katty | so 191 |
| 19:50.06 | Qwell | Katty: that's what NFAS does. It uses 1 D for several sets of Bs |
| 19:50.24 | kemix | ha sorry Qwell hand a great deal of help in the freepbx channel thought i would try my luck hear. good to know that you dont support the non asterisk geniuses :| |
| 19:50.37 | kemix | here* |
| 19:50.40 | Katty | kemix: still waiting for your recordy mixy line |
| 19:50.43 | Qwell | so instead of 23 * N, it's like 23 * N + (N / 4) |
| 19:50.45 | _Corey_ | p3nguin: I'm obviously missing a PB somewhere, I thought you were asking in general. |
| 19:50.51 | Katty | yeah, like 8x24-1 |
| 19:50.55 | p3nguin | _corey_: http://pastebin.ubuntu.com/670303/ |
| 19:50.58 | Katty | do math easier. |
| 19:51.05 | p3nguin | _corey_: Group 0 uses channels 1-15 and 17-31, and so on. |
| 19:51.05 | Qwell | I don't know if you'd want 8 on a D |
| 19:51.10 | Qwell | maybe that's normal though |
| 19:51.18 | Katty | anywho |
| 19:51.30 | Qwell | basically, if that 1 T1 goes down, you'd lose 8 instead of 1. |
| 19:51.36 | carrar | backup D |
| 19:51.43 | Qwell | hence 4 :) |
| 19:51.45 | _Corey_ | p3nguin: I fail to understand why they're not in one group if that's the goal and context is the same everywhere... |
| 19:52.01 | carrar | Go with POTS lines!! |
| 19:52.07 | Katty | kicks carrar out |
| 19:52.08 | p3nguin | _corey_: But the goal is to have a single Dial() command that will accommodate 200+ calls. I thoguht Dial(Dahdi/g0/${EXTEN}) would limit it to ~30 calls. |
| 19:52.23 | p3nguin | _corey_: 8 ports on the card |
| 19:52.31 | _Corey_ | p3nguin: It will, so you could simply go group=0 on everything |
| 19:52.45 | Katty | or add more than 30 channels onto your group |
| 19:52.51 | Katty | orrrr |
| 19:52.53 | Katty | use centrex lines |
| 19:52.57 | Katty | or don't |
| 19:53.00 | Katty | cause it's a pain, really |
| 19:53.01 | p3nguin | Next, kaushal has to understand and do this stuff. |
| 19:53.07 | Katty | YES |
| 19:53.12 | Katty | good point. |
| 19:53.13 | p3nguin | Talking to him is similar to talking to a bread box. |
| 19:53.15 | carrar | Get a DS3 of voice |
| 19:53.18 | Katty | let's stick with everything in a group |
| 19:53.23 | carrar | or go SIPness |
| 19:53.32 | carrar | SIP TRUNK!! |
| 19:53.33 | Katty | nomnomds3 |
| 19:53.34 | Katty | i want one |
| 19:53.36 | carrar | heh |
| 19:53.48 | Katty | let's stick with cupcakes |
| 19:53.51 | Katty | much easier to understand. |
| 19:53.54 | Katty | 1. unwrap |
| 19:53.56 | Katty | 2. stuff face |
| 19:54.00 | Katty | 3. repeat |
| 19:54.04 | _Corey_ | p3nguin: Yeah, I thought you were the one asking earlier or I would probably have ignored |
| 19:54.25 | p3nguin | strawberry cupcak with cream cheese frosting |
| 19:55.10 | p3nguin | _corey_: I initially wanted to help him out, but when we have to say the same thing multiple times and it still doesn't get any changes made, I lose interest. |
| 19:55.23 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 19:56.22 | p3nguin | I do want to understand it, though. With that existing configuration in the pastebin, if I tried to shove 100 calls out g0, will they all go, or will calls over ~30 be congested and lost? |
| 19:56.25 | _Corey_ | p3nguin: My philosophy is that I need to be getting paid if I'm going to repeat myself. |
| 19:56.38 | _Corey_ | congestion |
| 19:56.58 | Katty | p3nguin: i are confuzzed. |
| 19:57.03 | p3nguin | So will a single group containing all the channels be the way to do it? |
| 19:57.15 | Katty | p3nguin: you mean to tell me he wants to place 100 simultanious calls over a total of 30 channels? |
| 19:57.16 | carrar | yes |
| 19:57.16 | _Corey_ | sure |
| 19:57.17 | p3nguin | like this http://pastebin.com/MHrusnJ2 |
| 19:57.29 | mtbf | How can i get the destination extension from AGI script called from a macro? |
| 19:57.49 | mtbf | I get 's' and it's not what i need, i need the exact DID number the call comes from and i have many of them. |
| 19:57.50 | p3nguin | katty: He wants to send 200+ calls out all his 240 channels. |
| 19:57.58 | _Corey_ | p3nguin: Yeah, that should work fine |
| 19:58.01 | ChannelZ | Anyone using jabber/gtalk? |
| 19:58.15 | Katty | oh |
| 19:58.17 | ChannelZ | Been getting "res_jabber.c:2282 aji_handle_presence: Got presence packet from xxx@gmail.com, someone not in our roster!!!!" a whole bunch in the last day or two |
| 19:58.18 | p3nguin | mtbf: Store the extension into a variable before it goes to the macro. |
| 19:58.21 | Katty | then g 63 1-240 |
| 19:58.21 | WIMPy | p3nguin: Are you doing the relay now? |
| 19:58.30 | mtbf | p3nguin: thnx, i'll try that. |
| 19:58.43 | ChannelZ | A whole bunch will spit out in a row. I wonder if Gooooooogle has changed something or what is happening |
| 19:58.51 | Katty | hmm |
| 19:59.10 | carrar | ChannelZ, maybe you are no longer in the "circle" |
| 19:59.19 | ChannelZ | heh |
| 19:59.25 | *** join/#asterisk pgrace (~pgrace@hermes.vsix.me) |
| 19:59.25 | _Corey_ | p3nguin: The only reason not to do that would be if there was a need to bring some channels into a different context or something and then you still have the option of doing group=0,1 or whatever |
| 19:59.30 | ChannelZ | My own club that doesn't even want me as a member |
| 19:59.34 | Katty | aww |
| 19:59.37 | Katty | poor ChannelZ |
| 19:59.37 | p3nguin | wimpy: I'm still trying to help him out while I can try to understand it for myself. |
| 19:59.54 | Katty | group = 63 |
| 19:59.59 | Katty | channels => 1-240 |
| 20:00.05 | Katty | sumthin like that? |
| 20:00.07 | WIMPy | Didn't he get coding, yet? |
| 20:00.14 | pgrace | Hi! I'm trying to get Music On Hold working when a sip call gets the sdp packet notifying of inactive status. I've set CHANNEL(music) to default, which is the only musiconhold.conf stanza, but, no hold music when a=inactive is received. |
| 20:00.15 | p3nguin | Doubt it. |
| 20:00.19 | _Corey_ | Katty: You'll run into some D channels that way :) |
| 20:00.27 | pgrace | Is there some "Master On" switch for moh via chan_sip? |
| 20:00.28 | p3nguin | I even wrote it for him http://pastebin.com/MHrusnJ2 |
| 20:00.48 | p3nguin | I don't know if he ever understood. |
| 20:00.53 | Katty | pgrace: you define moh's per context |
| 20:01.09 | p3nguin | hmm |
| 20:01.09 | Katty | and i'll complete that thought in another 10 seconds |
| 20:01.12 | p3nguin | what? |
| 20:01.19 | WIMPy | With the trouble he has, I wouldn't put all channels in to one group. |
| 20:01.48 | p3nguin | I was hoping it was as easy as doing Dial(Dahdi/g0-7/${EXTEN}). |
| 20:01.53 | Katty | sooo |
| 20:02.00 | Katty | for an incoming SIP call it goes to a context |
| 20:02.05 | Katty | you can specify what moh context to use |
| 20:02.15 | _Corey_ | p3nguin: The '- |
| 20:02.17 | _Corey_ | er |
| 20:02.24 | pgrace | Katty: in this case I'm originating the call from asterisk |
| 20:02.26 | _Corey_ | the -7 being to grab channel 7? |
| 20:02.26 | p3nguin | g0:7? |
| 20:02.31 | WIMPy | p3nguin: I didn't pay attention here, but you'd have to use only one group. |
| 20:02.34 | pgrace | Katty: but the far end is sending the hold event. |
| 20:02.44 | Katty | pastebin your dial plan bits for that segment |
| 20:02.50 | _Corey_ | groups won't work that way |
| 20:02.52 | pgrace | sure, one sec. |
| 20:03.23 | jaytee | http://tinyurl.com/3jhk7de |
| 20:03.24 | WIMPy | p3nguin: But as he is too fast, I suggested, he sould use the interfaces sequentially. |
| 20:03.27 | p3nguin | moh can be suggested per PEER, or set per EXTENSION. |
| 20:03.29 | Katty | hai jaytee |
| 20:03.45 | jaytee | hai Katty |
| 20:03.54 | Katty | that photo is WIN |
| 20:03.58 | pgrace | katty: http://pastebin.com/g18KvHU9 |
| 20:04.03 | *** join/#asterisk nny1 (~SM@cpe-174-107-223-014.sc.res.rr.com) |
| 20:04.26 | pgrace | katty: I hacked that channel(musicclass) change into my setcid gosub just to have a quick-and-dirty way to supply it. |
| 20:04.33 | Katty | eww |
| 20:04.36 | Katty | yeah let's take tha tout |
| 20:04.51 | p3nguin | Did it even work? |
| 20:05.01 | Katty | i would set the musiconhold context right before your dial command |
| 20:05.10 | Katty | oh wait |
| 20:05.11 | p3nguin | class, not context. |
| 20:05.20 | p3nguin | Why wouldn't you just set the class in the extension? |
| 20:05.28 | Katty | he's not calling a sip phone |
| 20:05.36 | Katty | it's not in sip.conf |
| 20:05.42 | p3nguin | It says he is. |
| 20:05.45 | Katty | oh? |
| 20:05.49 | Katty | are we looking at the same line? |
| 20:05.51 | p3nguin | Dial(SIP/...) |
| 20:05.59 | Katty | no mean like |
| 20:06.02 | Katty | [100] |
| 20:06.05 | Katty | context=foo |
| 20:06.22 | _Corey_ | p3nguin: I could be wrong, but I'm thinking this may be what you're trying to understand: http://pastebin.com/d2GQfMLb |
| 20:06.26 | Katty | tho i guess it s asip registration to a website |
| 20:06.27 | Katty | that would work |
| 20:06.31 | pgrace | ok, so, I can set music on hold class in the context |
| 20:06.38 | p3nguin | no |
| 20:06.40 | Katty | no |
| 20:06.45 | p3nguin | But you can set it per PEER, or you can set it per EXTENSION. |
| 20:06.58 | pgrace | right, sorry. |
| 20:07.04 | pgrace | the peer definition |
| 20:07.07 | _Corey_ | p3nguin: In that case you can Dial(DAHDI/g0/xxx) to grab any of them, or Dial(DAHDI/g10/xxx) to dial only on the first span |
| 20:07.09 | pgrace | mohinterpret= and mohsuggest=? |
| 20:07.31 | p3nguin | Those are overridden by setting the class in the extension. |
| 20:07.38 | Katty | there is SetMusicOnHold() |
| 20:07.41 | p3nguin | Those often help in sip.conf, though. |
| 20:07.48 | pgrace | SetMusicOnHold is depreciated as of 1.6 |
| 20:07.55 | Katty | sumofacupcake |
| 20:07.57 | Katty | WHO DID THAT |
| 20:07.59 | p3nguin | CHANNEL(musiconhold) |
| 20:08.01 | pgrace | 1.6.2.7 here |
| 20:08.09 | pgrace | musiconhold? |
| 20:08.13 | pgrace | the doc on voip-info said musicclass |
| 20:08.20 | Katty | oh yeah |
| 20:08.44 | Katty | nm, it's depereciated |
| 20:08.54 | Katty | but you could still do that |
| 20:08.57 | Katty | prior to the GoSubIf |
| 20:09.03 | pgrace | trying it with the added bits in the peer def |
| 20:09.04 | Katty | or dial |
| 20:09.07 | Katty | i'd prefer it before the dial |
| 20:09.16 | p3nguin | depereciated? has no more monetary value? |
| 20:09.23 | Katty | p3nguin: exactly |
| 20:09.42 | pgrace | hmmm, nothing |
| 20:09.56 | p3nguin | pgrace: whatever it is... I didn't go look it up for you. |
| 20:10.05 | pgrace | no, that's fine |
| 20:10.08 | pgrace | I looked it up in sip.conf |
| 20:10.10 | *** join/#asterisk [sr] (~kvirc@pal-213-228-140-150.netvisao.pt) |
| 20:10.15 | [sr] | hi my voice friends! |
| 20:10.27 | Katty | SetMusicOnHold(client) <- so that doesn't work anymore? |
| 20:10.48 | Katty | it's just channel musicclass foo |
| 20:10.48 | p3nguin | It might not, since it is deprecated. |
| 20:10.50 | [sr] | one question, is it possible to ignore an MSN? i mean, never respond to request on that MSN? |
| 20:10.52 | p3nguin | Assume it won't. |
| 20:10.55 | *** join/#asterisk bensons (~thai_chi@push.vonmossner.de) |
| 20:11.26 | jaytee | DEPRECATED. USe Set(CHANNEL(musicclass)=...) |
| 20:11.43 | Katty | soooo |
| 20:11.52 | [sr] | ideias? |
| 20:11.53 | bensons | hi there. i installed asterisk now 1.7 with asterisk version 1.6. After logging to the webinterface i wanted to create a SIP trunk but i am not able to save. the button is simply a noop... any advice? |
| 20:11.57 | Katty | pgrace: yes, guess mohinterpret or mohsuggest |
| 20:11.57 | WIMPy | [sr]: I don't think you can silence Asterisk. |
| 20:12.12 | jaytee | still works in 1.6.2.14 but it breaks in 1.8.x |
| 20:12.14 | WIMPy | [sr]: Why do you want to? |
| 20:12.18 | *** part/#asterisk otwieracz (~gonet9@v6.gen2.org) |
| 20:12.55 | [sr] | WIMPy: idea is when i have 10 MSN's, but i want asterisk to ONLY respond on one of them, all the other 9, just ignore them, so that call's don't get answered |
| 20:13.02 | pgrace | Katty: neither work :( |
| 20:13.17 | Katty | well let's start the old fashioned way |
| 20:13.20 | pgrace | wish there was some deeper debug |
| 20:13.27 | Katty | make a new, simple extension, set the moh class, and stick yourself on hold |
| 20:13.33 | Katty | does that work? |
| 20:13.36 | WIMPy | [sr]: If you don;t configure extensions for them, the calls won't get answered. |
| 20:13.44 | pgrace | katty: sure, I'll try that. |
| 20:13.57 | [sr] | WIMPy: it does, i get a "number not in service... bla bla" |
| 20:14.33 | Katty | the number not in service... is that a standard telco audio message? |
| 20:14.43 | WIMPy | [sr]: Hmm, Ok, so then configure extensions for them and simply Hangup(16) them. |
| 20:14.52 | [sr] | Katty: no, asterisk |
| 20:15.13 | WIMPy | That must be the telco. |
| 20:15.25 | [sr] | WIMPy: ah, but the thing is that i'll have an ISDN phone on the same NTBA that want the call to be answered there, got the point? |
| 20:15.27 | Katty | i don't recall an asterisk audio file that sounds like that |
| 20:15.28 | WIMPy | Unless you have an I extension or some pattern doing it. |
| 20:15.30 | [sr] | remember my backup scenario |
| 20:15.40 | Katty | BUT it might be there. |
| 20:15.47 | [sr] | Katty: maybe it's freepbx audio.. not sure |
| 20:15.55 | Katty | possibly |
| 20:15.59 | Katty | you have how many lines? |
| 20:16.03 | WIMPy | It doesn;t matter if Asterisk hangs up the call. The phone will still have it. |
| 20:16.04 | Katty | pots lines? channels? |
| 20:16.07 | Katty | what do you gots |
| 20:16.51 | [sr] | Katty: isdn |
| 20:17.04 | WIMPy | That's the whole point of ptmp lines. |
| 20:17.07 | Katty | so you have 2 channels, and 1 d channel,r ight? |
| 20:17.09 | [sr] | WIMPy: i already tested it, asterisk picks up the call 1st |
| 20:17.43 | Katty | or do you have an isdn pri? |
| 20:17.53 | [sr] | Katty: 2x BRI |
| 20:18.02 | Katty | so 4 channels, and 2 d channels, yes? |
| 20:18.14 | WIMPy | [sr]: Then make sure you don;t have a matching extension, not the i pr e extension, either. |
| 20:18.38 | WIMPy | Today is D-channel confusion day :-) |
| 20:18.53 | Katty | infobot: d channel |
| 20:18.58 | Katty | infobot: d |
| 20:18.58 | infobot | d is Soraya_'s unique version of the n disease. There is no cure. |
| 20:19.05 | Katty | infobot: channel |
| 20:19.05 | infobot | i heard channel is This refers to the group of resellers that supply most companies with software, hardware, and support. The channel is a force to be reckoned with, and it competes directly against companies like Gateway 2000 and Dell. A channel can also be a content container, like a television channel. You may be watching TV channels on the Internet someday. |
| 20:19.09 | [sr] | lol |
| 20:19.13 | Katty | infobot: :< |
| 20:19.13 | infobot | somebody said < was redirection of stdin to a program |
| 20:19.14 | nny1 | anyone have a func_odbc entry for Executing a stored procedure I could stare at for a sec? Mine isn't working as expected |
| 20:19.22 | pgrace | katty: <whistles innocently> |
| 20:19.30 | Katty | pgrace: what did you do |
| 20:19.31 | pgrace | katty: format_mp3 should be compiled first..... |
| 20:19.33 | pgrace | :P |
| 20:19.37 | Katty | nice. |
| 20:19.39 | pgrace | ls |
| 20:19.40 | [sr] | WIMPy: get the picture on the scenario? asterisk always answers non-configured MSN's :S |
| 20:20.01 | WIMPy | [sr]: It shouldn't |
| 20:20.15 | WIMPy | Well, I'd even say it must not. |
| 20:20.35 | Katty | i don't think i understand what MSN is |
| 20:20.36 | [sr] | WIMPy: http://pastebin.com/TSiFDC6G |
| 20:20.51 | WIMPy | Katty: Just a phone number. |
| 20:20.57 | [sr] | Katty: that's the number's |
| 20:21.22 | [sr] | ok there's freepbx stuff in the middle, but it always answers it |
| 20:21.34 | Katty | so...incoming calls? |
| 20:21.39 | WIMPy | That must be some FreePBX "feature" then. |
| 20:21.39 | [sr] | it must be an options to ignore msn's, on asterisk or dahdi |
| 20:21.41 | Katty | just regular, incoming calls? |
| 20:21.53 | pgrace | the MSN is one of the numbers on a pair of ISDN lines I believe, right? |
| 20:21.57 | pgrace | so it's sort of like the DNIS |
| 20:22.02 | Katty | oh |
| 20:22.19 | Katty | well it doesn't base the routing off of the DNIS unless it checks the callerid off something |
| 20:22.23 | Katty | like the blacklist database, or manually |
| 20:22.38 | Katty | it routes it based on the channel it came in out, and dumps it into the context |
| 20:22.52 | pgrace | you can do normal extension logic by ani and dnis but I dont know if the MSN is different |
| 20:22.54 | Katty | but as long as a channel is in a context... |
| 20:23.21 | WIMPy | Channels have no meaning. |
| 20:23.24 | [sr] | pgrace: DNIS? MSN's in which country? |
| 20:23.29 | Katty | the channels inside the BRI |
| 20:23.32 | Katty | the physical channels |
| 20:23.36 | Katty | with the bit frames |
| 20:23.41 | Katty | in the THINGY |
| 20:23.44 | pgrace | hahaha |
| 20:23.51 | WIMPy | Yes, you never care about them. |
| 20:23.59 | pgrace | don't listen to me, I cant even tell if I have format_mp3 compiled before asking #asterisk to fix my stuff |
| 20:24.01 | Katty | but they go into contexts |
| 20:24.17 | Katty | and then asterisk does stuff based on those contexts |
| 20:24.31 | Katty | i must not understand this MSN stuff tho. |
| 20:24.33 | WIMPy | Katty: Yes. But the extension is the important bit. |
| 20:24.50 | Katty | and by extension you mean the callerid? |
| 20:25.00 | WIMPy | Think of it as DIDs, even if they aren't technically. |
| 20:25.02 | Katty | routing based on the phone number the other person called |
| 20:25.04 | *** join/#asterisk navaismo (~navaismo@fixed-203-96-202.iusacell.net) |
| 20:25.24 | Katty | well the did extensions still have to be in the same [context] |
| 20:25.38 | WIMPy | yes |
| 20:25.42 | Katty | if your 4 channels on the bri are routed into [foo] but the dids are in [foo2] it's just gonna spaz |
| 20:26.22 | Katty | unless it's just routing them based on the number of digits |
| 20:26.31 | Katty | it would be an all inclusive routing thingy if they did that |
| 20:26.32 | [sr] | i don't see any optioon :( |
| 20:26.46 | Katty | [sr]: tell me your question again |
| 20:26.47 | WIMPy | o.O number of digits? |
| 20:26.52 | Katty | XXX => do this |
| 20:26.55 | Katty | XXXX => do this isntead |
| 20:27.19 | Katty | i'm substituting msn for DID for this logic in my head |
| 20:27.28 | Katty | if all the 'DID's are being answered |
| 20:27.35 | Katty | there may be a XXXXXXXX => wah! |
| 20:27.37 | WIMPy | Katty: That only works with DDI, but I don;t see what sense that could make. |
| 20:28.03 | WIMPy | Yes, looks like FPBX does something like that, But who knows. |
| 20:28.03 | Katty | [sr]: repeat your question |
| 20:28.17 | [sr] | Katty: I want to ignore a MSN |
| 20:28.27 | [sr] | prevent asterisk from answer it! |
| 20:28.34 | Katty | ok |
| 20:28.42 | Katty | so...RIGHT NOW, what happens if you call that MSN |
| 20:28.47 | eduzimrs | atheos: just to let u know, that works fine. tks |
| 20:28.54 | Katty | does it get lumped into all the other calls? |
| 20:29.10 | Katty | [sr]: pastebin would be helpful to see exactly how it's being routed |
| 20:29.13 | [sr] | asterisk answeres it!!!!! |
| 20:29.18 | Katty | facepalms |
| 20:29.22 | [sr] | Katty: already did.... |
| 20:29.26 | Katty | link? |
| 20:29.28 | [sr] | http://pastebin.com/TSiFDC6G |
| 20:29.32 | [sr] | ai ai ai |
| 20:29.38 | [sr] | sent it a few lines before |
| 20:29.38 | [sr] | :p |
| 20:30.02 | WIMPy | [sr]: I fear, you have to take that to #freepbx |
| 20:30.05 | Katty | that link doesn't work for me |
| 20:30.14 | Katty | but if it's freepbx, i'm sure it's macro hell in there |
| 20:30.23 | [sr] | WIMPy: i'm trying... everyone's on coca-cola hour there.. :p |
| 20:31.11 | Katty | basically, what it comes down to, is how it's routed. |
| 20:31.19 | mtbf | Ok, i'm trying to set the variable in the context, before passing it to macro, like this: exten => s,n,Set(GLOBAL(DID_CALLER_ID)=${EXTEN}) |
| 20:31.35 | *** join/#asterisk qakhan (~qakhan@180.178.136.194) |
| 20:31.36 | [sr] | Katty: i don't want to route him, i want to ignore it |
| 20:31.38 | Katty | k... |
| 20:31.45 | qakhan | hi all |
| 20:31.50 | Katty | well to ingore him you're gonna have to send it to blacklist |
| 20:31.52 | pgrace | is there a way to load a module in 1.6 without a restart |
| 20:31.56 | Katty | or to another extension which does a hangup |
| 20:31.59 | Qwell | pgrace: module load foo.so |
| 20:31.59 | mtbf | and try to catch it in the macro body like this: exten => s,n,NoOp(Macro running and DID_CALLER_ID equals to ${DID_CALLER_ID}) |
| 20:32.00 | Katty | so to ignore, you have to route |
| 20:32.07 | qakhan | i want to leanr Asterisk AGI with php |
| 20:32.10 | mtbf | and i see in the rasterisk console it's empty. |
| 20:32.35 | qakhan | can any one help me or give me any reference |
| 20:32.51 | mtbf | The whole concept is wrong or do i just reffer to the global variable wrong way? |
| 20:33.18 | Qwell | mtbf: You shouldn't be using globals for that. |
| 20:33.19 | navaismo | @qakhan PHPAGI http://phpagi.sourceforge.net/ |
| 20:33.43 | Qwell | Why not just pass ${EXTEN} into the macro? |
| 20:34.06 | mtbf | As a first param, yup? How do i read it then? |
| 20:34.14 | Qwell | ${ARG1} |
| 20:34.17 | mtbf | Thnx. |
| 20:34.18 | qakhan | i dont have any exper on agi |
| 20:34.27 | qakhan | is this for beginner |
| 20:34.49 | Katty | cupcakes for beginners! |
| 20:34.54 | nny1 | what's the proper way to execute this function in dialplan from odbc_func: readsql=Exec [dbo].SP_AgentLogin ${ARG1} |
| 20:35.03 | navaismo | Yes im not expert too, but phpagi its easy |
| 20:35.04 | qakhan | yes :) |
| 20:35.12 | navaismo | and the examples help a lot |
| 20:35.23 | Qwell | nny1: You're missing a line or two. |
| 20:35.27 | qakhan | any PDF file? |
| 20:35.28 | Katty | infobot: cupcakes? |
| 20:35.40 | Qwell | ie; the section that is in |
| 20:35.40 | nny1 | Qwell I have the function name and dsn above |
| 20:35.51 | nny1 | Qwell: i.e. [name] dsn=dsn |
| 20:36.11 | *** part/#asterisk Bob_Pierce (~WESTMAN\p@216.36.132.162) |
| 20:36.14 | Qwell | Set(foo=${FUNC_name(myarg)}) |
| 20:36.23 | nny1 | 55510,n,Set(ODBC_AGENTLOGIN=${dagentid}) |
| 20:36.25 | nny1 | yeah |
| 20:36.26 | nny1 | odd |
| 20:36.32 | Qwell | err, not FUNC |
| 20:36.34 | Qwell | ODBC |
| 20:36.51 | Qwell | You can't write to that function. |
| 20:36.57 | Qwell | You've only specified a read |
| 20:36.57 | nny1 | QWELL if I run it from cli should it work or just output the results (i.e. odbc read ODBC_AGENTLOGIN 1111) ? |
| 20:36.58 | qakhan | any PDF of AGI PHP for beginners? |
| 20:37.06 | nny1 | Qwell: oh |
| 20:37.15 | nny1 | Qwell: so I need writesql? |
| 20:37.20 | Qwell | yes |
| 20:37.23 | nny1 | Qwell thanks |
| 20:37.27 | Qwell | and it wouldn't be ARG1 at that point |
| 20:37.35 | nny1 | oh? |
| 20:37.43 | Qwell | VALn |
| 20:38.26 | nny1 | ok |
| 20:38.33 | *** join/#asterisk ketas-av (~ketas@kvlt.eu) |
| 20:38.33 | Qwell | really, you shouldn't be doing that particular query as a write though. Kinda silly. |
| 20:38.37 | nny1 | so Exec [dbo].SP_AgentLogin ${VAL1} |
| 20:38.53 | Qwell | just use the set line I gave you |
| 20:38.53 | nny1 | Qwell: what's the best way to send info to a stored procedure ? |
| 20:39.38 | nny1 | Qwell: what if i need to pass multiple arguments where the result would be AutoDialCallResultInsert ${VAL1},${VAL2},${VAL3},${VAL4} |
| 20:40.31 | nny1 | basically I am sending arguments to a database as a call progresses, no local variable setting happens |
| 20:41.46 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 20:42.51 | navaismo | @qakhan I dont have it |
| 20:42.58 | [sr] | WIMPy: Katty, think i have a solution |
| 20:43.09 | navaismo | you should read php tutorials and then merge with phpagi |
| 20:44.17 | hardwire | meeeeerge |
| 20:44.25 | *** join/#asterisk Aut0ExeC (~Jack@24.244.156.75) |
| 20:44.29 | Aut0ExeC | w/c |
| 20:44.30 | *** part/#asterisk Aut0ExeC (~Jack@24.244.156.75) |
| 20:45.52 | hardwire | heh |
| 20:47.22 | navaismo | My tip: make simples examples with google, php manual and phpagi examples to understand |
| 20:47.26 | nny1 | Qwell think I understand, testing now |
| 20:47.50 | mtbf | Ok, now i'm trying to pass additional variable to the AGI perl script, but it doesn't work neither after space (no such file or dir), after comma and after | as according to http://www.voip-info.org/wiki/view/Asterisk+AGI |
| 20:48.12 | mtbf | changing some of agi variables before calling the script would also do the trick for me. |
| 20:48.39 | [sr] | WIMPy: katty, create a virtual extension, create the inbound route and redirect the route to the virtual extension, and define a "pause before answer" |
| 20:48.47 | [sr] | think it's solved! |
| 20:48.48 | nny1 | Well pbx.c:3494 ast_func_read: Function ODBC_AGENTLOGIN cannot be read |
| 20:48.53 | nny1 | er Qwell: pbx.c:3494 ast_func_read: Function ODBC_AGENTLOGIN cannot be read |
| 20:49.03 | [sr] | just have to test near the isdn fone to confirm it's ringing |
| 20:49.36 | WIMPy | Sounds urgs. |
| 20:49.56 | mtbf | exten => s,1,Set(agi_extension=${ARG1}) is fine? |
| 20:50.11 | [sr] | WIMPy: urgs = ? |
| 20:50.20 | tonsofpcs | <PROTECTED> |
| 20:50.45 | WIMPy | Try to pronounce it :-) |
| 20:51.06 | [sr] | WIMPy: no idea what it can be.. hehe say it |
| 20:52.11 | *** part/#asterisk bensons (~thai_chi@push.vonmossner.de) |
| 20:52.50 | WIMPy | A sound like when you puke. |
| 20:54.03 | [sr] | oh i see |
| 20:54.27 | [sr] | but there's an extra problem, grr, every two call's it doesn't start ringing and hangs up with cause 81 |
| 20:54.52 | [sr] | ah wait forget, already read the logs |
| 20:55.02 | [sr] | i tryed blacklist before, my fault, going to remove |
| 20:57.37 | [sr] | now happens with cause 102, grr |
| 20:57.48 | [sr] | where can i see the cause reasons? |
| 20:59.13 | [sr] | i saw it, recovery on timer expire, no idea what is.. |
| 20:59.17 | Katty | ohai |
| 20:59.22 | Katty | i was doing some reading up on astrophotography |
| 20:59.48 | jaytee | back in the day we had to pack film in dry ice to make it hyper sensitive |
| 21:00.00 | jaytee | back when they used film |
| 21:00.23 | Katty | why did the cold make the film extra sensitive? |
| 21:00.42 | Katty | i presume you mean more affected by light |
| 21:00.42 | *** part/#asterisk qakhan (~qakhan@180.178.136.194) |
| 21:01.05 | jaytee | more light sensitive and less prone to fogging from infra-red |
| 21:01.24 | Katty | i wonder why tho |
| 21:01.38 | *** join/#asterisk fenlander (~fenlander@82.152.81.57) |
| 21:02.16 | [sr] | seams to be an NT problem, WIMPy? |
| 21:02.47 | WIMPy | NTs don;t do anything. |
| 21:03.54 | Katty | i'm hungry. :< |
| 21:03.56 | Qwell | jaytee: weird, I'd think cold would make something more prone to fogging |
| 21:04.11 | Qwell | Katty: I'm making pizza when I get home. Be there! |
| 21:04.16 | Katty | :> |
| 21:04.17 | Katty | k |
| 21:04.17 | [sr] | now hungup with 101 |
| 21:04.36 | Qwell | Katty: lies :p |
| 21:04.49 | thehar | :D |
| 21:05.15 | Katty | :< |
| 21:05.20 | Katty | tis true |
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| 21:15.46 | Katty | Qwell: you're so clever. |
| 21:17.09 | *** part/#asterisk nny1 (~SM@cpe-174-107-223-014.sc.res.rr.com) |
| 21:19.23 | Qwell | Katty: just pedantic |
| 21:20.25 | [sr] | WIMPy: & Katty seems to work OK :) tomorow i'll confirm! |
| 21:20.55 | Katty | woot |
| 21:22.13 | Katty | so i txt the boy to see when he's taking breaks at work tnight |
| 21:22.19 | Katty | cause he's been doing a lot of overtime and such |
| 21:22.28 | Katty | and the reply i get is 'dinnerish time likely or movie time or boht' |
| 21:22.44 | Katty | whatever happened to using TIMES >.< |
| 21:24.20 | Katty | dinnerish time and movie time mean nothing to me!! alsdfjlakjsdflfoan |
| 21:24.35 | Katty | Qwell: have a word with him. kthx. |
| 21:25.15 | [sr] | WIMPy: there here some sort of "blackhole destination" |
| 21:25.36 | Katty | [blackhole] |
| 21:25.43 | Katty | exten => s,1,dostuff |
| 21:25.56 | Katty | maybe.. Hangup()! |
| 21:26.05 | [sr] | that |
| 21:26.05 | Katty | maybe Background(rickroll) |
| 21:26.16 | Katty | maybe...forward them to pizza hut! |
| 21:26.29 | [sr] | pizzahut it's good |
| 21:26.31 | [sr] | yammi |
| 21:29.06 | Katty | goto(blackhole,s,1) |
| 21:29.30 | hardwire | hehe |
| 21:33.15 | [sr] | problem solved! |
| 21:33.26 | [sr] | it's not exactly an ignore but solved my issue |
| 21:48.09 | Katty | i've never done anything like this, so not sure if it's possible. |
| 21:48.16 | Katty | i have a remote office, with an old digital phone |
| 21:48.31 | Katty | is there /something/ i can plug that old digital phone into and make it function, and register, as a sip phone? |
| 21:49.19 | navaismo | An ATA |
| 21:49.25 | [sr] | Katty: for analog/isdn i think i saw devices for that |
| 21:49.25 | navaismo | like PAP2T |
| 21:49.41 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-146.cablep.bezeqint.net) |
| 21:50.17 | paulc | Katty: What kind of phone is it, exactly? |
| 21:50.30 | Katty | i don't know. i've not actually seen it yet |
| 21:50.42 | Katty | i just had a whole bunch of crap dumped onto my lap by the boss |
| 21:50.52 | Katty | and i'm not too happy about it |
| 21:50.53 | Qwell | Katty: if it's digital, no, nothing to speak of. there was some Nortel digital ATA, but it was *expensive* |
| 21:51.05 | Qwell | expensive enough to buy a new high-end SIP phone |
| 21:51.14 | Katty | ok |
| 21:51.29 | Katty | let's pretend it's a craptacular analog phone from walmart |
| 21:51.57 | Katty | the linksys pap2t would work? |
| 21:52.24 | Katty | and allows you to register as a sip phone would |
| 21:54.23 | _Corey_ | Katty: What you're looking for is from Citel |
| 21:54.29 | *** join/#asterisk Pimmetje (~Pimmetje@s5146b5f9.adsl.wanadoo.nl) |
| 21:54.57 | _Corey_ | I believe they cost more than PAP2s or other ATAs per-port though |
| 21:55.11 | Katty | k |
| 21:55.15 | _Corey_ | but if you really want to sip-enable your digital phone inventory, that's what they do |
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| 21:56.04 | *** part/#asterisk passthru (~ramon@201009179146.user.veloxzone.com.br) |
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| 21:56.35 | Katty | i really hate that my boss wants to do everything cheap. |
| 21:56.52 | Katty | irritates the crap out of me. |
| 21:56.56 | p3nguin | katty: ss-noservice |
| 21:57.20 | _Corey_ | yeah, every time someone has asked me to salvage digital stuff I've made an ROI case for new phones |
| 21:58.27 | Katty | yeah |
| 21:58.28 | _Corey_ | Citel isn't bad though, they pretty much make all the digital stuff work |
| 21:58.32 | Katty | meh |
| 21:58.40 | Katty | if it's even over half the cost of a polycom, they should just go with a polycom |
| 21:58.46 | Katty | even a refurb |
| 21:59.43 | p3nguin | For each of my contexts for calls from an ITSP, I define every extension for active phone numbers, then I have a catch-all extension for any others which plays that ss-noservice file. |
| 22:03.46 | *** join/#asterisk cerienjean (~iper@95.138.77.91) |
| 22:07.08 | [sr] | bye ppl |
| 22:33.31 | carrar | werd |
| 22:33.33 | carrar | y0 |
| 22:33.57 | carrar | Smacks up Katty, I HEAR YA |
| 22:34.14 | carrar | Love Smack, that is |
| 22:34.31 | carrar | not those Evil bad haterid smacks |
| 22:34.56 | carrar | continues to Justify! |
| 22:36.43 | *** join/#asterisk cerienjean (~iper@95.138.77.91) |
| 22:55.56 | drmessano | I would rather "represent" |
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| 23:26.48 | carrar | he |
| 23:26.50 | carrar | h |
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