| 00:00.21 | ChannelZ | You'd probably need to get the 'other' host to subscribe to the mailboxes. I think * will only send notifys directly to the peer in question |
| 00:00.57 | docelmo | So the phone would subscribe to this host and then this host would need to subscribe to the asterisk box |
| 00:03.41 | ChannelZ | Yes. What is the 'other' host running? |
| 00:04.15 | docelmo | Its probably a dirty word in here but Freeswitch because of Shared Line Appearence |
| 00:06.24 | ChannelZ | Well two things come to mind, 1. whether or not FS can be configured to do that, and/or 2. what sort of access restrictions there are on the * side for an outsider subscribing to a bunch of mailboxes |
| 00:08.27 | coreyf1513 | docelmo: asterisk 11 seams to support remote mwi subscriptions. not sure what version it was added. search your sample sip.conf for 'OUTBOUND MWI'. |
| 00:10.44 | ChannelZ | I think that's the other way around though. |
| 00:11.50 | ChannelZ | (IE asterisk subscribing to MWI messages from somewhere else) |
| 00:13.08 | coreyf1513 | oh.. right so only useful if he needs to get mwi from one asterisk to another (the asterisk with phones would subscribe to mwi on the asterisk with voicemail) |
| 00:20.12 | docelmo | ok thanks! |
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| 07:20.50 | schmidts | good morning |
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| 07:48.19 | madhatt3r | helloAll, fridayAtLast |
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| 08:04.13 | bulkorok | friday! |
| 08:08.58 | schmidts | TGIF!!!!! |
| 08:09.46 | bulkorok | hell yeah :-> |
| 08:10.23 | madhatt3r | fire up that asterisk, call your friends, let them know :D |
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| 08:14.38 | din3sh | gdmorning all |
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| 08:43.59 | Chainsaw | drmessano: Just curious, how is your machine doing? |
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| 08:51.31 | duffman | quick question about ring strategies: I have a ring group set to dial a total of 3 extensions, with memoryhunt. My issue is however is that while it's ringing the second extension on the list the first extension still rings. |
| 08:51.44 | duffman | Worst case scenario I've got three phones going off at once. |
| 08:53.46 | duffman | I could shorten the ring time from 20 to like 10 |
| 08:54.11 | phix | hai |
| 08:54.24 | duffman | hello phix :) |
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| 08:54.56 | duffman | damn it would be so much easier to specify amount of rings instead of ring time |
| 08:55.24 | duffman | the first extension on the list usually picks up quickly, second extension does not as they're not there all the time |
| 08:55.45 | duffman | regardless first and second extension usually ring together after 3-5 rings on the first extension |
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| 09:00.11 | duffman | I'm guessing I'm just debating my own questions at this point. |
| 09:06.48 | *** join/#asterisk jacekowski (jacekowski@jacekowski.org) |
| 09:06.51 | jacekowski | hi people |
| 09:07.01 | jacekowski | i've got a problem with FAR end echo |
| 09:07.02 | ChannelZ | * doesn't necessarily know how long a ring lasts |
| 09:07.33 | jacekowski | which echo canceller is the best to deal with that |
| 09:08.24 | ChannelZ | What is the interface? |
| 09:08.35 | jacekowski | ISDN BRI |
| 09:08.36 | jacekowski | on my side |
| 09:08.54 | jacekowski | on far side, probably analog line |
| 09:08.58 | jacekowski | judging by |
| 09:08.59 | jacekowski | PRI Span: 1 < Ext: 1 Progress Description: Called equipment is non-ISDN. (2) ] |
| 09:09.49 | ChannelZ | well I use MG2 on analog and it works just fine |
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| 09:09.58 | ChannelZ | dunno for you tho |
| 09:10.17 | jacekowski | how many taps? |
| 09:10.48 | ChannelZ | good question, default.. |
| 09:10.49 | ChannelZ | looks |
| 09:11.19 | ChannelZ | 128 |
| 09:11.54 | ChannelZ | but again my situation is not at all yours |
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| 09:16.12 | ectospasm | jacekowski: wait, the far end hears echo? that's the purview of the far end. |
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| 09:16.52 | ectospasm | ...nothing you can do in Asterisk will affect far end echo. they've probably got a hybrid near to that far end which is acting up. |
| 09:17.26 | jacekowski | no |
| 09:17.30 | jacekowski | i hear the echo from far end |
| 09:17.44 | jacekowski | as in, delayed by half a second or so |
| 09:17.48 | ectospasm | ah... which card is it? |
| 09:17.58 | jacekowski | 05:00.0 ISDN controller: Cologne Chip Designs GmbH ISDN network Controller [HFC-4S] (rev 01) |
| 09:18.11 | jacekowski | it's digital on my end |
| 09:18.47 | ectospasm | hmmm... don't know that card at all. |
| 09:19.06 | jacekowski | openvox clone |
| 09:19.10 | ectospasm | you can try one of the software echo cancelers: mg2,kb1,sec,sec2 |
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| 09:36.56 | duffman | the hell... international dialing just stopped working |
| 09:38.55 | ectospasm | pastebin the ACLI output |
| 09:38.56 | phix | yay |
| 09:39.03 | phix | duffman: That means your phone bill will be reduced |
| 09:39.09 | duffman | no |
| 09:39.11 | duffman | hold on |
| 09:39.12 | phix | thrusts in your direction |
| 09:40.33 | duffman | here is the outpu |
| 09:40.35 | duffman | *output |
| 09:40.37 | duffman | http://pastebin.com/uMKjG6ND |
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| 09:41.57 | ectospasm | you need to find the cause of this, duffman: Got SIP response 480 "Temporarily Unavailable" back from OURIP:5060 |
| 09:42.22 | kaldemar | cause 19 = no answer |
| 09:42.25 | duffman | undestood |
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| 09:42.35 | duffman | however, if I dial my cell or a number in norway it works |
| 09:42.42 | duffman | and international dialing worked fine earlier |
| 09:43.19 | kaldemar | what changed since earlier? |
| 09:43.37 | duffman | nothing as far as I know |
| 09:43.40 | duffman | I haven't touched the dialing patterns |
| 09:43.54 | duffman | I guess there could be some conflict there though |
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| 09:49.20 | duffman | hmm |
| 09:49.30 | duffman | this applies to all international numbers it seems |
| 09:49.33 | duffman | it just times out |
| 09:50.10 | duffman | Does anyone have experience with norwegian dial patterns? |
| 09:50.26 | duffman | I really don't, and I'm having a hard time understanding it properly. |
| 09:55.36 | phix | duffman: thrust? |
| 09:55.40 | duffman | what? |
| 09:56.17 | phix | duffman thrusts in your direction |
| 09:56.44 | phix | You should get bac =k to your roots duffman |
| 09:56.51 | phix | s/ =// |
| 09:57.04 | phix | yay! regex bot! |
| 09:57.11 | duffman | :/ |
| 09:57.19 | phix | :) |
| 09:57.25 | phix | Keep the dream alive |
| 09:57.34 | duffman | you make no sense man |
| 09:57.44 | phix | Thrust like you have never thrusted before |
| 09:58.10 | phix | *shrugs* you chose the nick name |
| 09:58.20 | duffman | I get the refrence though :P |
| 09:58.25 | duffman | duff man can't breath, oh no! |
| 09:58.30 | phix | Prefer for duffman references |
| 09:58.32 | phix | :) |
| 09:58.38 | phix | Prefer = prepeare |
| 09:58.44 | duffman | *Prepare |
| 09:58.47 | duffman | lololl |
| 09:58.49 | phix | that oe |
| 09:58.50 | phix | one |
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| 09:58.56 | phix | fucsadibng keyboard |
| 09:59.39 | duffman | my dial pattern starts with 00. |
| 09:59.43 | duffman | 00 (to dial internationally) |
| 09:59.48 | duffman | . = wildcard |
| 09:59.52 | phix | ok |
| 09:59.58 | duffman | maybe the problem isn't with asterisk, but with my provider? |
| 10:00.00 | phix | 000 is australian emergency call |
| 10:00.05 | duffman | -_- |
| 10:00.10 | phix | 00 is US? |
| 10:00.15 | duffman | 001 is US |
| 10:00.15 | phix | 0011 is AU |
| 10:00.30 | duffman | wtfff |
| 10:00.33 | duffman | now international dialing works |
| 10:00.38 | phix | yup |
| 10:00.41 | duffman | wait |
| 10:00.51 | phix | maybe you need another beer duffman |
| 10:01.05 | duffman | so why does 001 (800) 444-4444 not work, but 001 (800) 555-1212 works |
| 10:01.06 | duffman | ? |
| 10:01.10 | duffman | maybe mci is down |
| 10:02.09 | phix | 001 is for calling international lines? |
| 10:02.14 | duffman | yes |
| 10:02.15 | duffman | to the uS |
| 10:02.31 | phix | so calling from the US to an international country? |
| 10:02.39 | duffman | no |
| 10:02.42 | duffman | calling from norway |
| 10:02.46 | phix | oh |
| 10:03.00 | phix | 001 to call international lines from norway? |
| 10:03.02 | duffman | I think the problem is with the ANAC number 800 444-4444 (MCI) |
| 10:03.04 | duffman | no |
| 10:03.05 | duffman | fuck |
| 10:03.11 | duffman | 001 = dialing to the US from norway |
| 10:03.14 | duffman | 00 is for international |
| 10:03.17 | duffman | 1 is the country code |
| 10:03.19 | phix | ok |
| 10:03.23 | duffman | if I were to dial to england it would be 0044 |
| 10:03.29 | duffman | etc |
| 10:03.40 | phix | ok call this number: 0061294811111 |
| 10:03.44 | phix | for testing purposes |
| 10:03.50 | duffman | ? |
| 10:04.11 | phix | Order me a large supreme with BBQ sauce |
| 10:04.12 | duffman | that's AU? |
| 10:04.15 | phix | yup |
| 10:04.21 | phix | Pizzahut |
| 10:04.34 | duffman | yup that works |
| 10:04.37 | phix | :) |
| 10:04.47 | duffman | basically I want my provide to trunkate numbers as much as possible |
| 10:04.59 | duffman | makes it easier |
| 10:05.00 | phix | is that word? |
| 10:05.04 | phix | trunkate? |
| 10:05.09 | phix | or truncate? |
| 10:05.15 | duffman | woops |
| 10:05.19 | phix | truncate means to reduce / |
| 10:05.21 | duffman | dunno |
| 10:05.30 | duffman | brb |
| 10:06.53 | phix | it does though |
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| 11:01.07 | puzzled | hi |
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| 11:01.28 | duffman | hello |
| 11:03.13 | phix | hi |
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| 11:40.01 | Rumbles | ahoi hoi |
| 11:43.54 | Ice_Strike | Hi |
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| 12:04.10 | StaRetji | Folks, is this okay to block anonymous calls? |
| 12:04.11 | StaRetji | exten => _./anonymous,1,Hangup() |
| 12:04.50 | StaRetji | also, does it matter if it is Anonymous and anonymous, will it block both? |
| 12:04.52 | StaRetji | thx |
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| 12:08.12 | leifmadsen | StaRetji: it is not case sensitive afaik |
| 12:08.31 | leifmadsen | regardless, I don't think I've ever used that format. I always just do a CALLERID() check |
| 12:10.04 | StaRetji | thx leifmadsen |
| 12:10.18 | StaRetji | how about empty? |
| 12:10.35 | StaRetji | exten => _./,1,Hangup() |
| 12:10.40 | leifmadsen | I don't use that format |
| 12:11.25 | leifmadsen | GotoIf($["${CALLERID(name)" = "" | "${CALLERID(num)" = ""]?HangupClause,start,1) |
| 12:13.41 | WIMPy | Only a / used to work in 1.4, it didn't work in 1.6 any more. Haven't tried it since then. |
| 12:16.57 | leifmadsen | it's kind of an old method to use |
| 12:17.09 | leifmadsen | use of the CALLERID() function should be considered the new standard |
| 12:17.18 | StaRetji | leifmadsen: GotoIf($["${CALLERID(name)" = "" | "${CALLERID(num)" = ""]?HangupClause,start,1) |
| 12:17.26 | leifmadsen | yes, that's what I typed |
| 12:17.28 | StaRetji | where is the list of calle ide? |
| 12:17.33 | StaRetji | yes, lol, I know |
| 12:17.35 | StaRetji | sorry |
| 12:17.41 | leifmadsen | you asked about empty strins |
| 12:17.44 | leifmadsen | strings |
| 12:17.46 | leifmadsen | I showed an example |
| 12:17.50 | StaRetji | oh, yes, thx |
| 12:18.14 | StaRetji | but I have also a list of callerid in block.conf |
| 12:18.21 | StaRetji | I made #include |
| 12:18.31 | leifmadsen | otherwise, if you need a list of items to parse on, create a subroutine |
| 12:18.50 | StaRetji | are you guys saying exten => _./anonymous,1,Hangup() doesn't work on 1.8? |
| 12:18.58 | leifmadsen | we're saying it might not |
| 12:19.03 | leifmadsen | because no one really uses that formatting anymore |
| 12:19.03 | StaRetji | got it |
| 12:19.22 | StaRetji | but in case it work, will exten => _./,1,Hangup() do the job? |
| 12:19.32 | leifmadsen | unknown |
| 12:19.33 | leifmadsen | try it |
| 12:19.38 | StaRetji | ok |
| 12:19.40 | StaRetji | thx |
| 12:19.43 | leifmadsen | that is a simple test that you could have already gotten the answer to |
| 12:19.46 | WIMPy | Try it an we will know if iot has been fixed. |
| 12:20.24 | leifmadsen | regardless to the answer, due diligence warrants a test to validate things happen as expected |
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| 12:23.54 | acidfoo | someone have a cisco call manager here and is using the Groupe LIsten Feature on an sccp 7906 or 7911, I have some questions, and would appreciate a network dump between the phone and the CCM ;) |
| 12:24.06 | acidfoo | http://www.cisco.com/en/US/docs/voice_ip_comm/cuipph/7906g_7911g/8_0/english/user/guide/06speak.html <- the CCM feature in question |
| 12:24.19 | StaRetji | will do, need to wait when it happens, a phone call |
| 12:24.31 | leifmadsen | you could always generate a phone call with no callerid.... |
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| 12:34.05 | StaRetji | hm, I don't know how to do it lol and don't want to bother you guys |
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| 12:49.57 | volga629 | Hello Everyone. How I can debug IAX2 call to check if encryption in place ? |
| 13:05.15 | jacekowski | i've enabled mg2 echo canceller |
| 13:05.22 | jacekowski | and echo is gone |
| 13:05.29 | jacekowski | but sound is bit strange |
| 13:05.36 | jacekowski | like sort of crackling |
| 13:07.12 | phix | GO TEAM! |
| 13:09.28 | *** part/#asterisk deo (~deo@203.177.214.75) |
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| 13:43.10 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 14:34.44 | jacekowski | is it possible to execute some cli commands automatically after restart |
| 14:34.53 | jacekowski | (i'm trying to make sure that pri debugging is always enabled) |
| 14:35.12 | *** join/#asterisk ruben231 (~OpenDial@112.198.90.176) |
| 14:35.26 | leifmadsen | jacekowski: see cli.conf |
| 14:37.19 | ruben231 | hi guys i ahve installed asterisk 1.8 but after install when i run asterisk -rvvvv or asterisk -vvvvvvvvvvvvvvvgc i got this error-------------------> Illegal instruction <-----------------any idea on this..? |
| 14:38.03 | file | what are you running it on? |
| 14:38.13 | *** part/#asterisk StaRetji (~LittleAll@80.93.240.171) |
| 14:38.34 | jacekowski | sounds like it's compiled for different processor |
| 14:38.42 | jacekowski | or newer procesor |
| 14:38.54 | jacekowski | ruben231: can you try running it on gdb |
| 14:38.55 | leifmadsen | package force installed? |
| 14:39.31 | ruben231 | jacekowski: how to run on gdb..? |
| 14:39.32 | *** join/#asterisk igcewieling (~igcewieli@ip98-183-25-31.pn.at.cox.net) |
| 14:39.34 | *** part/#asterisk igcewieling (~igcewieli@ip98-183-25-31.pn.at.cox.net) |
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| 14:39.57 | jacekowski | gdb asterisk |
| 14:40.10 | jacekowski | and then after gdb loads the thing just do |
| 14:40.17 | jacekowski | run -rvvvvvvvv |
| 14:40.23 | ruben231 | ok ill try now |
| 14:43.51 | ruben231 | root@vicidev:~# gdb asterisk ---> -bash: gdb: command not found <------------this is the message |
| 14:44.36 | ruben231 | jacekowski: thats the output |
| 14:46.34 | ruben231 | jacekowski: still there..? |
| 14:47.44 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 14:47.46 | *** join/#asterisk saint_ (~saint@c-68-38-56-184.hsd1.nj.comcast.net) |
| 14:47.49 | saint_ | hi all.. |
| 14:48.07 | [TK]D-Fender | ruben231, you don't appear to have it installed. have you considered trying to install it? |
| 14:48.34 | saint_ | MMhh.. I just discovered that google voice is not encrypted... does anyone know if it's possible to use SIPTLS with google voice ? |
| 14:49.45 | [TK]D-Fender | saint_, Google Voice is not SIP |
| 14:50.13 | file | the XMPP signaling is encrypted using TLS |
| 14:50.19 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 14:50.21 | file | the audio stream is not |
| 14:52.34 | jacekowski | ruben231: install gdb |
| 14:54.53 | saint_ | file: is there any way to do that ? (encrypt the audio stream) ? |
| 14:55.54 | file | if you are using it to access the PSTN for inbound or outbound, no |
| 14:56.20 | saint_ | ha. you just ruined my intentions of using google voice at home them ... |
| 14:57.31 | jmetro | how many people actually encrypt the audio stream |
| 14:58.22 | saint_ | jmetro: I would say when you are within the same company / building , not too many. but if you use SIP for outgoing/incoming like with ATT or VERIZON, then you want that. |
| 14:58.27 | igcewieling | jmetro: virtually none as far as I can tell. |
| 14:58.32 | saint_ | who's going to discuss confidential information over the phone if it's unencrypted ? |
| 14:58.55 | igcewieling | saint_: you know regular phone lines are not encrtyped, right? They are also far easier to intercept |
| 14:59.00 | saint_ | saint_ |
| 14:59.01 | jmetro | ^ |
| 14:59.18 | saint_ | igcewieling: nothing to do with that. in order to listen to a T1, you need to physically tape into it |
| 14:59.24 | igcewieling | grabs a phone and a janitor's uniform and reports to work at saint's employer |
| 14:59.29 | WIMPy | If you start to encrypt audio, you will have to pay more taxes. |
| 15:00.03 | leifmadsen | also sacrificial kittens are required |
| 15:00.19 | jmetro | i believe encrypted audio requires at least two or three firstborn kittens |
| 15:00.22 | leifmadsen | and the amount of encryption on the audio stream in many places is pretty much non-existant |
| 15:00.22 | saint_ | igcewieling: with sip, you don t need to physically tape in the network you are targetting |
| 15:00.26 | leifmadsen | jmetro: +1 |
| 15:00.38 | leifmadsen | just man in the middle it |
| 15:01.03 | igcewieling | saint_: since audio generally goes direct there is often not a single point to tap. My point is that VoIP is not any LESS secure than PSTN |
| 15:02.05 | saint_ | igcewieling: i do not agree. You agree that let's say you could either tap PSTN or SIP , right ? |
| 15:02.26 | saint_ | It is easier to encrypt SIP, and therefore make it harder to listen to. |
| 15:02.37 | saint_ | So I think it makes it a little bit more secure |
| 15:02.52 | saint_ | I'm not talking about taping in right now, but if you got a piece of recording |
| 15:02.58 | leifmadsen | well, encrypting SIP itself won't encrypt any audio |
| 15:02.59 | saint_ | granted you taped in |
| 15:03.06 | saint_ | leifmadsen: ok, i m talking about SRTP |
| 15:03.11 | jmetro | both endpoints must support the encryption on the entire route from what i remember |
| 15:03.22 | saint_ | jmetro: yes |
| 15:03.43 | saint_ | jmetro: but nowadays it is easier to encrypt IP than it is to encrypt analog |
| 15:03.50 | leifmadsen | if we're being pedantic about level of security around PSTN vs internet media streams, lets be pedantic in our use of terms |
| 15:04.17 | igcewieling | saint_: if it is so important then set up a VPN |
| 15:04.19 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 15:04.27 | WIMPy | SRTP is no good for that. You'd need ZRTP. |
| 15:04.31 | WIMPy | And what is analog? |
| 15:04.38 | igcewieling | also, what do you expect for a free service? |
| 15:05.04 | saint_ | igcewieling: you can't run on VPNs, that opens more can of worms |
| 15:05.24 | *** join/#asterisk navaismo (~navaismo@189.241.118.172) |
| 15:05.26 | volga629 | Is any body know iax2 client with support encryption ? |
| 15:05.27 | saint_ | igcewieling: i just expected that google, as it is on of the top worldwide companies, would provide this service.. |
| 15:05.43 | igcewieling | saint_: yes, you can and many many people do. In fact, Verizon wholesale SIP service REQUIRES you have VPN between you and VZ. |
| 15:05.51 | jmetro | seeing as they are the top of the world in services and they dont support encryption, that might tell you something. |
| 15:06.24 | *** join/#asterisk nricciar (~nricciar@64.207.62.66) |
| 15:06.54 | igcewieling | you need a VPN which doesn't so something stupid like tunnel over TCP or even HTTPS, of course, but I don't consider those real "VPNs" anyway |
| 15:09.21 | saint_ | Anyway... I'll go with VOIP.MS that supports siptls and srtp. |
| 15:10.15 | [TK]D-Fender | saint_, And not T.38 |
| 15:10.37 | saint_ | [TK]D-Fender: I use fax.com |
| 15:11.23 | [TK]D-Fender | saint_, For now maybe... but just before you make another potential assumption as to how things work.... |
| 15:11.48 | ruben231 | [TK]D-Fender: i tried re compiling but still----> Illegal instruction |
| 15:12.39 | saint_ | [TK]D-Fender: seriously ? I don t even know who's using fax anymore. I barely use my account. Everything is done through email nowadays. |
| 15:13.15 | jmetro | or should be... fax is very much alive unfortunately |
| 15:13.24 | mjordan | ruben231: are you building on a VM? |
| 15:14.08 | mjordan | ruben231: what is the CPU architecture on your VM? |
| 15:14.33 | ruben231 | mjordan: yes building on a VM |
| 15:14.44 | mjordan | mutters about having to write a wiki article |
| 15:14.52 | [TK]D-Fender | ruben231, You aren'te showing anything.... and I wasn't the one who was going to follow-up on the results of GDB |
| 15:15.00 | mjordan | ruben231: run 'make menuselect'. Go to the build options, disable BUILD_NATIVE |
| 15:15.01 | mjordan | re-build |
| 15:15.10 | saint_ | oh man |
| 15:15.14 | mjordan | that *may* fix your problem. If not, you need to specify your CPU architecture to the build system |
| 15:15.22 | saint_ | voip.ms does NOT support encryption neither. |
| 15:15.23 | mjordan | most likely, you've got some funky virtual CPU architecture |
| 15:15.37 | [TK]D-Fender | saint_, And there goes that.... |
| 15:15.48 | saint_ | [TK]D-Fender: yeah.. |
| 15:16.18 | saint_ | I like their "Passwords are MD5 hashed. Basically the communication is secured but not encrypted" . |
| 15:16.32 | jmetro | unless youre talking about government secrets all the time [who as far as i know, the DoD is the only place that encrypts..] you probably dont need srtp |
| 15:17.09 | saint_ | jmetro: not really the point.. would you give your credit card number, social security number, or other personal information if you knew that anyone could listen to it ? |
| 15:17.28 | jmetro | people do that over cell in busy places constantly. |
| 15:17.59 | saint_ | jmetro: i iknow, that is how i got my database of 5000 cc numbers |
| 15:18.08 | saint_ | but more seriously .. :D |
| 15:19.33 | *** join/#asterisk miesekatze (d4ca6c82@gateway/web/freenode/ip.212.202.108.130) |
| 15:19.37 | miesekatze | hello |
| 15:20.17 | miesekatze | I have a problem with CID strings |
| 15:22.43 | *** join/#asterisk Leddy (leddy@krypton.evosurge.com) |
| 15:24.10 | ruben231 | mjordan:i already make menuselect--what part i should select BUILD_NATIVE..? |
| 15:24.36 | miesekatze | CALLERID(all) appers to return ""+XXYYZZZ" <VYYZZZ>" |
| 15:24.39 | miesekatze | WARNING[7016]: ast_expr2.fl:445 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected '<token>', expecting $end; Input: ""+XXYYZZZ" <VYYZZZ>"=" |
| 15:24.39 | *** join/#asterisk gerhard7 (~gerhard7@82-169-24-72.ip.telfort.nl) |
| 15:24.56 | igcewieling | miesekatze: show us the actual dialplan line |
| 15:25.14 | ruben231 | <PROTECTED> |
| 15:25.16 | *** join/#asterisk evilman_home (kvirc@95-29-217-118.broadband.corbina.ru) |
| 15:25.27 | [TK]D-Fender | miesekatze, that format is perfectly normal... |
| 15:26.01 | navaismo | ruben231, try in compiler flags |
| 15:26.15 | miesekatze | exten = s,1,ExecIf($[ "${CALLERID(all)}"="" ]?Set(CALLERPRES()=unavailable)) |
| 15:26.51 | igcewieling | miesekatze: what version of Asterisk? |
| 15:26.56 | miesekatze | 1.6 |
| 15:26.58 | saint_ | [TK]D-Fender: so do you know any sip provider that supports tls/srtp ? |
| 15:27.13 | [TK]D-Fender | saint_, Not off-hand. Get hunting.... |
| 15:27.22 | miesekatze | [TK]D-Fender: the inner " are not a problem? |
| 15:27.33 | igcewieling | miesekatze: at first glance, it looks to me like it should work. did you remember to reload after making changes to extensions.conf? |
| 15:27.44 | miesekatze | yes |
| 15:28.20 | miesekatze | its just a warning so, but if this is normal, it's ok with me :) |
| 15:28.23 | ruben231 | navaismo:---->http://pastebin.com/BMabdmne<---------------this is the output |
| 15:28.53 | igcewieling | miesekatze: quotes in Asterisk are.....odd. try running Set(CALLERID(all)=Testing <1234>) before that execif and see if you still have the error. (notice the lack of quotes) |
| 15:29.19 | miesekatze | ok, will try |
| 15:29.45 | igcewieling | also try a "core show function CALLERPRES" to verify it still exists in 1.6 |
| 15:30.15 | [TK]D-Fender | igcewieling, Not the issue yet |
| 15:30.46 | navaismo | ruben231, you cant scroll down |
| 15:30.49 | *** join/#asterisk slav3_kitten (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 15:30.50 | navaismo | ? |
| 15:31.26 | *** part/#asterisk mihamina (~mihamina@ip-41-190-237-66.orange.mg) |
| 15:31.40 | navaismo | ruben231, and which version are you trying to compile? |
| 15:34.38 | miesekatze | igcewieling: the reult is [Feb 22 17:36:55] WARNING[7138]: ast_expr2.fl:445 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected '<token>', expecting $end; Input: ""Testing" <1234>"="" |
| 15:35.59 | igcewieling | miesekatze: you didn't put quotes in your Set? |
| 15:36.32 | miesekatze | igcewieling: exten = s,1,Exec(Set(CALLERID(all)=Testing <1234>)) |
| 15:36.48 | miesekatze | that's what in extensions.conf |
| 15:37.13 | igcewieling | miesekatze: looks to me like it should work, I don't know why it isn't. |
| 15:37.33 | [TK]D-Fender | miesekatze, pastebin the complete call attempt and dialplan, not just tiny fragments |
| 15:37.34 | jmetro | did you reload afterwards to commit the change? |
| 15:37.57 | jmetro | exit |
| 15:38.00 | jmetro | woops |
| 15:39.30 | miesekatze | [TK]D-Fender: ok, cannot do this atm, have to check the full config again firts. thnak you for your help |
| 15:39.45 | miesekatze | igcewieling: thank you for your help! |
| 15:40.12 | navaismo | yup failing here too ^ |
| 15:41.24 | miesekatze | [TK]D-Fender: btw, the dialplan contains much stuff from asterisk gui and i have to check wirst if it influences with this part |
| 15:42.35 | [TK]D-Fender | miesekatze, saying you can't show us the call is a very bad sign |
| 15:43.47 | saint_ | anyone ever tried GSM gateways with Asterisk ? |
| 15:45.34 | navaismo | [TK]D-Fender, igcewieling miesekatze ---> http://pastebin.com/e0aij48X |
| 15:45.48 | navaismo | removing quotes the error gone |
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| 15:46.07 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 15:46.27 | igcewieling | miesekatze: try putting a space around the = and see if that makes any difference. It shouldn't, but worth trying |
| 15:46.37 | navaismo | saint_, I have used GSM gateways from teltonika,2n and lyric/YXwireless |
| 15:46.50 | saint_ | navaismo: which one did you prefere ? |
| 15:48.10 | navaismo | lyric/YXwireless |
| 15:49.31 | saint_ | navaismo: nice design ... |
| 15:51.24 | navaismo | yep |
| 15:51.47 | saint_ | navaismo: which one did you use ? X2 , X4, ot X6 ? |
| 15:52.23 | navaismo | igcewieling, the quotes are breaking the string cid, the result take ""<--empty string then CDINAME<--floating value and finally "<CIDNUM>"<--another string |
| 15:52.45 | navaismo | saint_, I have tested the X2 and X6 |
| 15:52.57 | igcewieling | navaismo: and yet I've never had that problem |
| 15:52.59 | saint_ | navaismo: do you know the price of the X2 by any chance ? |
| 15:53.33 | navaismo | did you see my pb, i have the same error |
| 15:53.35 | navaismo | :( |
| 15:53.39 | navaismo | saint_, nope |
| 15:53.54 | igcewieling | miesekatze: you could try using CALLERID(num) and CALLERID(name) instead. |
| 15:54.34 | *** join/#asterisk lorsungcu (~anonymous@75-146-164-138-Minnesota.hfc.comcastbusiness.net) |
| 15:54.46 | igcewieling | navaismo: I don't have a non-production server I can test on easily so I'll take your word for it. |
| 15:54.47 | navaismo | saint_, in their page have the starting prices |
| 15:54.57 | saint_ | navaismo: where at ? |
| 15:55.11 | navaismo | igcewieling, better see this http://pastebin.com/e0aij48X |
| 15:55.30 | *** join/#asterisk lorsungcu (~anonymous@75-146-164-138-Minnesota.hfc.comcastbusiness.net) |
| 15:55.37 | navaismo | igcewieling, if I remove the quotes from CALLERID(all) no more error |
| 15:55.48 | navaismo | saint_, http://www.yx.cl/lyric_voip.html |
| 15:55.54 | igcewieling | navaismo: I believe you. |
| 15:56.13 | saint_ | navaismo: holy crap ! |
| 15:56.43 | igcewieling | an alternative would be ExecIf($[ '${CALLERID(all)}'='' ]?Set(CALLERPRES()=unavailable)) or ExecIf($[ X${CALLERID(all)}=X ]?Set(CALLERPRES()=unavailable)) |
| 15:56.57 | leifmadsen | always remember that quotes in the dialplan are literal characters |
| 15:57.05 | leifmadsen | and are not indicators that something is a string |
| 15:57.17 | leifmadsen | a " is no different than an x |
| 15:57.20 | igcewieling | leifmadsen: except in this case they appear to be. |
| 15:57.22 | Qwell | leifmadsen: Your face is a string. |
| 15:57.27 | igcewieling | which is why I'm confused |
| 15:57.39 | leifmadsen | Qwell: an infinite string? |
| 15:58.27 | navaismo | igcewieling, same issue using ' |
| 15:58.45 | igcewieling | navaismo: how about using X? |
| 16:00.11 | igcewieling | you could try using shrinkcallerid=yes in sip.conf |
| 16:02.27 | navaismo | igcewieling, if the CID isnt empty & using the X same result |
| 16:04.36 | igcewieling | navaismo: I'd ask that guy who wrote the Asterisk Book 8-) |
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| 16:39.19 | drmessano | Chainsaw: You around? |
| 16:39.56 | Chainsaw | drmessano: Yes./ |
| 16:41.13 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 16:42.24 | drmessano | Those patches didn't help the crashing. I moved up to 11.3 rc1 with fingers crossed. |
| 16:43.04 | drmessano | I did apply those patches though, since I am sure those unresolved issues will pop up too.. if I can get past 12 hours uptime lol |
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| 16:54.20 | shinao1 | hi please im using asterisk-based call center application with a sip fxo gateway (grandstream gxw4018). when calls come in via the fxo gateway and end, asterisk doesnt close the channel and new calls have a busy message. Please how can i fix this? |
| 16:55.29 | igcewieling | shinao1: the sip/fxo gateway is the device which should signal Asterisk that the call has ended. |
| 16:55.51 | shinao1 | i see |
| 16:56.20 | shinao1 | please does anyone have experience with grandstream gateways to see how this can be fixed? |
| 16:57.26 | navaismo | check the manual |
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| 17:26.22 | Ice_Strike | <PROTECTED> |
| 17:26.30 | Ice_Strike | I see this in CLI, what does this mean? |
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| 17:36.35 | navaismo | using amd or in the voicemail?? |
| 17:39.36 | *** join/#asterisk sipman (~slane@71-14-128-129.dhcp.ftwo.tx.charter.com) |
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| 17:59.09 | ChannelZ | Ice_Strike: You using WaitForSilence probably |
| 18:00.07 | *** part/#asterisk ashleyd (~ashleyd@188-221-41-129.zone12.bethere.co.uk) |
| 18:00.10 | ChannelZ | or maybe AMD, not sure if it says things like that too |
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| 18:12.24 | tompaw | Which package is needed on debian to build rec_config_pgsql? |
| 18:12.40 | tompaw | I installed libpq5-dev and still nothing. |
| 18:14.16 | shinao1 | thanks a lot everyone |
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| 18:15.48 | *** part/#asterisk ipiera (~Paul@ipiera.plus.com) |
| 18:17.46 | igcewieling | tompaw: do a "make distclean" re-run ./configure and run menuselect and make sure the module is enabled. |
| 18:21.50 | tompaw | igcewieling: amke distclean did the trick, thanks. |
| 18:22.22 | igcewieling | tompaw: in Asterisk never trust ./configure to pick up newly installed libraries. |
| 18:25.25 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 18:30.06 | *** join/#asterisk vlad_starkov (~vlad_star@81.22.194.213) |
| 18:31.19 | *** join/#asterisk stev3n (Stiv@94.181.214.28) |
| 18:31.32 | stev3n | hi, how to make a call from a SIP-phone, and it has initiate a call from dahdi-phone to this extension? |
| 18:31.41 | stev3n | for example I call 911 on sip-phone, and dahdi-phone calling to 911 |
| 18:33.09 | *** join/#asterisk Alagar (~helpdesk@vsusm15.vernalissystems.com) |
| 18:33.49 | *** join/#asterisk cmendes0101 (~cmendes01@wtnl.corp.tierra.net) |
| 18:38.04 | ChannelZ | Dial(DAHDI/1/911) |
| 18:38.19 | ChannelZ | where DAHDI/1 is the channel to use, or something like g1 if you have channel group 1 defined, etc. |
| 18:40.21 | stev3n | hmm, i.e. 911 now be calling to dahdi? |
| 18:40.44 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 18:41.11 | stev3n | there are two phones, sip and phones |
| 18:41.18 | stev3n | i call on sip 911 |
| 18:41.45 | stev3n | and dahdi phone starts ringing? |
| 18:42.43 | stev3n | on my side? |
| 18:43.05 | ChannelZ | I guess I don't understand what you want |
| 18:43.35 | navaismo | stev3n, can you be more clear, as far i understand you have a FXS phone and a SIP phone, your fxs phone its the peer 911 and you want to dial from the sip phone? |
| 18:43.45 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 18:43.45 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 18:44.11 | stev3n | i want initiate call on dahdi-phone to extension by sip-phone |
| 18:44.55 | navaismo | exten => 911,1,Dial(DAHDI/1) assuming the port 1 have a fxs phone connected |
| 18:45.32 | navaismo | still unclear to me |
| 18:45.48 | stev3n | example: sip soft phone ring to extension |
| 18:45.54 | ChannelZ | ~book |
| 18:45.54 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 18:46.08 | *** join/#asterisk Nobody08 (~chatzilla@d216-232-17-171.bchsia.telus.net) |
| 18:47.04 | stev3n | * save this extension and initiate call on dahdi to this extension |
| 18:47.54 | navaismo | your terminology(or my translation) confuse more and more |
| 18:48.08 | stev3n | asterisk save this extension and initiate call on dahdi to this extension |
| 18:48.24 | stev3n | sip soft phone ring to extension, asterisk save this extension and initiate call on dahdi to this extension |
| 18:48.33 | stev3n | still confuse? :( |
| 18:48.38 | jmetro | yes |
| 18:48.46 | navaismo | stev3n, asterisk do not save extensions... |
| 18:48.49 | ChannelZ | well all we can say is that if you want to dial 1234 and have it call DAHDI/1 you'd do exten => 1234,1,Dial(DAHDI/1) |
| 18:49.07 | stev3n | save value of extem |
| 18:49.12 | stev3n | save value of extension |
| 18:49.32 | ChannelZ | ${EXTEN} is the extension being dialed. |
| 18:49.48 | ChannelZ | exten => 1234,1,Dial(DAHDI/1/${EXTEN}) |
| 18:50.03 | ChannelZ | in this case ${EXTEN} would evaluate out to 1234 |
| 18:50.15 | stev3n | I know the basics |
| 18:50.30 | ChannelZ | I guess it's a language barrier issue then because we don't understand what you're asking |
| 18:50.56 | navaismo | stev3n, maybe a diagram would help |
| 18:51.34 | *** join/#asterisk bmg505 (~leon@196-209-120-97.dynamic.isadsl.co.za) |
| 18:51.42 | stev3n | ok, wait a minute plz |
| 18:51.43 | ChannelZ | Maybe he's talking about callbacks? |
| 18:52.00 | stev3n | I draw diagram |
| 18:52.59 | navaismo | maybe |
| 18:54.21 | ChannelZ | I do love the doodles |
| 18:54.35 | *** join/#asterisk vlad_starkov (~vlad_star@81.22.194.213) |
| 18:57.03 | drmessano | OMG |
| 18:57.16 | drmessano | Are we waiting for the glue to dry on the macaroni? |
| 18:57.36 | ChannelZ | ran out of glitter |
| 18:57.44 | drmessano | HA |
| 18:57.48 | *** join/#asterisk tekzilla (~jon@hmbg-4d069eee.pool.mediaWays.net) |
| 18:58.24 | drmessano | All out of Bejazzles.. so can't illustrate endpoints? |
| 18:58.57 | drmessano | "Macaroni for proxy too wet. Next time, do not boil" |
| 19:00.09 | ChannelZ | LOL |
| 19:00.37 | stev3n | http://dl.dropbox.com/u/16445500/my_diagramm.png |
| 19:00.50 | stev3n | :) |
| 19:01.25 | drmessano | Uh |
| 19:02.09 | drmessano | So you want the SIP call to initiate a call between a DAHDI device and an outside line? |
| 19:02.24 | stev3n | yes |
| 19:02.31 | drmessano | Originate() |
| 19:02.47 | stev3n | exten => s,1,Originate(SIP/myphone,exten,default,s,1) |
| 19:02.51 | stev3n | like this? |
| 19:03.08 | stev3n | exten => s,1,Originate(DAHDI/myphone,exten,default,s,1) |
| 19:03.30 | pabelanger | did you try it? |
| 19:03.35 | stev3n | no |
| 19:03.42 | pabelanger | so, what makes you think we will? |
| 19:03.58 | pabelanger | CLI*> core show application originate |
| 19:04.32 | drmessano | http://www.voip-info.org/wiki/view/Asterisk+cmd+Originate <-- That too |
| 19:04.42 | leifmadsen | pukes a bit |
| 19:04.43 | ChannelZ | That would work assuming DAHDI/myphone was valid, and the 's' extension was what was being dialed from your SIP phone... |
| 19:04.47 | leifmadsen | don't link to voip-info please |
| 19:04.55 | leifmadsen | especially for dialplan applications and functions |
| 19:05.16 | leifmadsen | wiki.asterisk.org has the info that has been imported from the source directly |
| 19:05.42 | drmessano | Someone needs to work on search engine optimization for the wiki then |
| 19:05.44 | ChannelZ | not quite as friendly but at least it's usually right :) |
| 19:05.53 | drmessano | Because it never shows up in results |
| 19:06.03 | ChannelZ | yah |
| 19:07.00 | drmessano | I'm not going to spend 10 mins trying to find a proper link when the intention is starting point |
| 19:07.05 | drmessano | is a* |
| 19:12.04 | stev3n | thanks for the help |
| 19:14.39 | *** join/#asterisk ashleyd (~ashleyd@188-221-41-129.zone12.bethere.co.uk) |
| 19:14.45 | *** join/#asterisk miltux (~miltux@62.1.139.246.dsl.dyn.forthnet.gr) |
| 19:14.53 | *** join/#asterisk cmendes0101 (~cmendes01@wtnl.corp.tierra.net) |
| 19:14.56 | ChannelZ | Hmm, I guess you can't cheat and use Dial as an app with Originate? |
| 19:15.01 | *** part/#asterisk ashleyd (~ashleyd@188-221-41-129.zone12.bethere.co.uk) |
| 19:15.58 | ChannelZ | oh nevermind, that does work. |
| 19:17.40 | ChannelZ | stev3n: I think you want exten => 1234,1,Originate(DAHDI/1,app,Dial,DAHDI/2/${EXTEN}) assuming DAHDI/1 was your 'dahdi phone' and DAHDI/2 was your ISDN, and 1234 was the number you were to dial to make all this happen and it would call 1234 as well |
| 19:19.37 | stev3n | yes, right |
| 19:19.48 | ChannelZ | adapt as needed |
| 19:27.53 | *** join/#asterisk boch (~boch@186.182.116.157) |
| 19:28.00 | boch | hi all |
| 19:29.07 | boch | quick question, is it possible to use options 'o' and 'B' togheter when calling ChanSPY app? I need to only listen both channels |
| 19:29.35 | Qwell | What happened when you tried? |
| 19:30.46 | *** join/#asterisk areski (~areski@80.174.255.57.dyn.user.ono.com) |
| 19:31.00 | boch | if both present, i hear single channel, if only B present, i hear both channels, but they listen to me |
| 19:33.24 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 19:33.24 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 19:33.43 | boch | its like option 'o' break option 'B' |
| 19:34.38 | drmessano | Don't use either |
| 19:35.31 | drmessano | You're forcing one direction, or barging in on the call. Leave the default |
| 19:35.57 | drmessano | In other words, you're trying too hard |
| 19:36.18 | boch | so, its not possible to only hear both channels ? |
| 19:36.35 | drmessano | YES |
| 19:36.39 | drmessano | BY DEFAULT |
| 19:36.52 | drmessano | NO OPTIONS |
| 19:39.39 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 19:39.59 | boch | thats weird because when using no options i only hear specified channel |
| 19:42.33 | *** join/#asterisk ashleyd (~ashleyd@188-221-41-129.zone12.bethere.co.uk) |
| 19:42.47 | *** part/#asterisk ashleyd (~ashleyd@188-221-41-129.zone12.bethere.co.uk) |
| 19:43.12 | igcewieling | boch: the "specified channel" will have two way audio all on the same single channel |
| 19:44.40 | boch | i understand, the fact is i hear a person talking with silence, ill dig deeper.. |
| 19:45.17 | boch | thanks for the answers |
| 19:50.38 | *** join/#asterisk BrokenArrow (~BrokenArr@unaffiliated/brokenarrow) |
| 19:50.39 | ChannelZ | yeah with just Chanspy(SIP,q) or whatnot I get both legs |
| 19:52.15 | ChannelZ | perhaps your other party is using sign language |
| 19:56.54 | *** mode/#asterisk [-b *!*chatzilla@*unaffiliated/t-dot-zilla/x-2830497] by Qwell |
| 19:57.56 | *** join/#asterisk t_dot_zilla (~chatzilla@unaffiliated/t-dot-zilla/x-2830497) |
| 19:57.58 | *** mode/#asterisk [+b *!*chatzilla@*unaffiliated/t-dot-zilla/x-2830497] by Qwell |
| 19:57.58 | *** kick/#asterisk [t_dot_zilla!~north@pdpc/sponsor/digium/Qwell] by Qwell (Nevermind.) |
| 19:58.22 | leifmadsen | nice |
| 19:58.29 | putnopvut | bitch-slapped |
| 20:01.04 | *** join/#asterisk Lars_G (~Lars@unaffiliated/lars-g/x-000001) |
| 20:01.06 | Lars_G | Hi all. |
| 20:03.46 | ChannelZ | ahoyhoy |
| 20:04.37 | Lars_G | Question, I'm planning on moving from PBX to asterisk, but the perfect environment to put a digium is a server we'll be getting for virtualizing a few services (XCP or Xen). My worry is, in your experience can I share the Digium pci-e cards with a DomU in a Xen environment? |
| 20:06.43 | leifmadsen | the hardware will not work inside the virtual environment |
| 20:06.57 | leifmadsen | the way to do it is to have hardware that is not virtualized, and share that access via SIP into the VMs |
| 20:07.07 | igcewieling | Lars_G: most of us do not recommend running a production Asterisk server inside a VM |
| 20:07.14 | leifmadsen | igcewieling: it works fine |
| 20:07.19 | leifmadsen | we have 60+ of them |
| 20:07.24 | igcewieling | leifmadsen may be an exception. |
| 20:07.32 | leifmadsen | you just need to know how to use virtual machines correctly |
| 20:07.49 | eirirs | hmm do I see a Norwegian name here? |
| 20:07.54 | eirirs | at least Scandinavian |
| 20:08.09 | igcewieling | Canadian, but the confusion is understandable 8-) |
| 20:08.11 | leifmadsen | most problems is with the way people setup the virtual machines, not asterisk inside a virtual environment |
| 20:08.19 | igcewieling | (assuming you mean leifmadsen) |
| 20:09.05 | eirirs | hehehe |
| 20:09.10 | eirirs | you hit it |
| 20:10.03 | Lars_G | igcewieling: A full vm... that is :) |
| 20:10.36 | Lars_G | igcewieling: I know a person here an asterisk guru, who runs 10 asterisks inside OpenVZ interconnected by the asterisk trunk protocol, to overcome the extension number limitation |
| 20:11.01 | Lars_G | I think I'll go with pci cards then, and shove them in an older server being decomissioned |
| 20:11.08 | file | ...what extension number problem? |
| 20:11.24 | jmetro | ^ |
| 20:11.27 | jmetro | i was just looking forthat |
| 20:11.33 | igcewieling | Lars_G: for every success story there are plenty of failure stories |
| 20:12.03 | Lars_G | file: As far as I heard from him, at least it used to be so 3 or 4 years ago, there was a limitation to around 256 extensions per server due to an interrupt problem/limitation, he didn't delve into it. |
| 20:12.46 | igcewieling | Lars_G: DAHDI CHANNELS, not EXTENSIONS |
| 20:12.51 | Lars_G | igcewieling: Yes, I think I'll go with digium pci, and the older machine. A core 2 duo with 4Gb should be much MORE than able to handle 90 extensions, 16 external COs and ivr and voice mail. |
| 20:12.58 | Lars_G | igcewieling: Sorry, my bad. |
| 20:15.49 | Lars_G | shouldn't it? |
| 20:17.32 | *** join/#asterisk vlad_starkov (~vlad_star@178.176.152.133) |
| 20:21.00 | Lars_G | Ok, a different question, do you guys know of any existing easy calculator for calculating bandwidth? in case I go with a gateway |
| 20:24.25 | jmetro | http://www.asteriskguru.com/tools/bandwidth_calculator.php |
| 20:25.38 | Lars_G | Thank you |
| 20:25.50 | *** part/#asterisk volga629 (~volga629@host7.pythian.com) |
| 20:26.04 | Lars_G | How I wish I could get a grandstream gw here, but it's been a bitch to find |
| 20:27.09 | jmetro | i like polycoms |
| 20:27.19 | Lars_G | I have an aversion to polycom, IDK why |
| 20:27.32 | jmetro | aastra? |
| 20:28.17 | Lars_G | IDK them, but I'll keep looking for a gateway |
| 20:28.29 | Lars_G | Specially since right now the CO FXB is on the hall. |
| 20:28.37 | Lars_G | And out own old and useless FXB is on floor 5. |
| 20:28.41 | Lars_G | And they want to also move us. |
| 20:29.22 | Lars_G | So I preffer to put two or three 8xFXO gateways in the hall in the FXB box, ventilate it some, cascade them together, and then run asterisk on my server whenever they move us to. |
| 20:34.18 | igcewieling | Adtran TA 90x boxes make great PSTN/SIP gateways |
| 20:34.30 | Lars_G | igcewieling: I need it to support fax as well btw. |
| 20:34.55 | Lars_G | So I need T38 |
| 20:35.00 | igcewieling | Lars_G: sucks to be you. The Adtrans support T.38. |
| 20:35.06 | Lars_G | and I HAVE to buy locally, so if there's no distributor, i'm effed |
| 20:35.23 | igcewieling | Lars_G: looks to me like you are setting yourself up to fail. |
| 20:35.34 | Lars_G | igcewieling: It's law |
| 20:35.47 | Lars_G | I work for govt, I can't buy anything in foreign coin |
| 20:36.11 | jmetro | so locally as in within the country |
| 20:37.10 | *** join/#asterisk melter (~Melter@2001:4930:116:0:1826:b35a:4576:31a5) |
| 20:38.10 | Lars_G | jmetro: Correct |
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| 21:09.20 | *** part/#asterisk tech_travis (~Travis@174.46.237.60) |
| 21:09.23 | igcewieling | Lars_G: Adtran is a *huge* telecom equipment manufacturer, they likely have a local presense. |
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| 21:48.43 | jmetro | Dialplan wise, i can smush two variables together like ${PATH_TO_FILE}${EXTEN}-AutoAttendant and have it evaluate to /path/to/file/100-AutoAttendant right? |
| 21:51.18 | WIMPy | sure |
| 22:08.43 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
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| 22:54.12 | phix | jmetro: correct |
| 22:55.20 | Ice_Strike | Where do you guys save background process on linux? |
| 22:55.32 | Ice_Strike | <PROTECTED> |
| 22:55.49 | phix | save a background process? |
| 22:55.58 | Ice_Strike | a script file i meant |
| 22:56.01 | phix | ah |
| 22:56.06 | phix | where ever you like |
| 22:56.22 | phix | I stick everything under /srv/asterisk |
| 22:56.22 | Ice_Strike | I might do in /usr/local/bin |
| 22:56.56 | phix | but others may use /var/lib/asterisk or /usr/local/[s]?bin |
| 22:58.05 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:58.46 | Ice_Strike | Ah |
| 23:00.06 | phix | G'day [TK]D-Fender! |
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| 23:09.35 | *** join/#asterisk resist0r (~resist0r@c-174-57-0-74.hsd1.nj.comcast.net) |
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| 23:42.15 | ChannelZ | :copy |
| 23:42.20 | ChannelZ | oops |
| 23:44.08 | *** join/#asterisk polysics (~Adium@host243-3-dynamic.180-80-r.retail.telecomitalia.it) |
| 23:44.12 | polysics | hello |
| 23:44.20 | polysics | trying to run a SIPp with auth test |
| 23:44.38 | polysics | why could Asterisk not even be answering my SUBSCRIBES with 401? |
| 23:44.56 | polysics | I just see the SUBSCRIBE go in and Asterisk does nothing, which is odd |
| 23:45.40 | [TK]D-Fender | Show us |
| 23:45.47 | [TK]D-Fender | with FULL SiP debug enabled |
| 23:48.04 | polysics | how do I enable the ful ldebug, please? I only know of "sip set debug on" |
| 23:48.59 | ChannelZ | that would be the one |
| 23:49.53 | polysics | https://gist.github.com/polysics/a7d89c9c48a4f83266c8 |
| 23:50.14 | polysics | and that is all. The SUBSCRIBE looks identical to the one Blink sends that works |
| 23:52.54 | ChannelZ | Do you need an Allow: line? |
| 23:53.40 | polysics | Blink has a Supported line |
| 23:53.44 | polysics | but no Allow |
| 23:54.04 | [TK]D-Fender | Those are REGISTERS |
| 23:54.16 | [TK]D-Fender | and I find the lack of any response highly suspect |
| 23:54.24 | polysics | https://gist.github.com/polysics/0bfcc324ab5522816542 works |
| 23:54.41 | polysics | me too, not for any other reason if not "why no response"? :-) |
| 23:54.50 | polysics | I mean, Asterisk should say SOMETHING |
| 23:55.11 | polysics | not mad at Ast btw, just puzzled |
| 23:55.27 | [TK]D-Fender | We don't see what version you are working with and I'm wondering what your configs, FW, etc all look like... |
| 23:56.40 | polysics | firewall is iptables -F |
| 23:56.43 | ChannelZ | This is some custom client or something you're using/writing? |
| 23:56.51 | polysics | this is SIPp |
| 23:56.51 | *** join/#asterisk ruben231 (~OpenDial@112.198.90.86) |
| 23:57.00 | polysics | SVN, running against, one sec |
| 23:57.01 | ChannelZ | oh |
| 23:57.11 | polysics | Asterisk 1.8.10.1~dfsg-1ubuntu1 |
| 23:57.32 | polysics | need sip.conf? I had excluded that since Blink works |
| 23:57.44 | ChannelZ | well I don't know the SIP spec enough to know what is required and what isn't, I do see some differences between your working/unworking registers tho |
| 23:58.18 | polysics | waht do you see, please? anything helps :-) |
