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| 02:30.31 | Sean-Der | I have been passing my func_odbc queries by just having readsql=${ARG1} and my full query in dialplan |
| 02:30.42 | Sean-Der | I can't seem to get the same to work with writesql any ideas? |
| 02:31.06 | *** join/#asterisk vlad_starkov (~vlad_star@77.41.88.210) |
| 02:34.03 | jpsharp | I believe it uses VAR1 instead of ARG1 when you writesql |
| 02:35.56 | Sean-Der | I am getting an issue with whitespace, I can't seem to pass VAR1/ARG1 Like I am with readsql |
| 02:36.17 | Sean-Der | The syntax for strings is just " " ? |
| 02:36.26 | Sean-Der | I think thats what I am messing up |
| 02:36.46 | Sean-Der | same => n,Set(ODBC_SQL()="Update blah") |
| 02:42.28 | jpsharp | Lemme paste what I use. |
| 02:43.23 | jpsharp | writesql=UPDATE `accounts` SET `account_balance` = `account_balance` - ${VAL1} where account_code = "${ARG1}" |
| 02:44.36 | jpsharp | exten => _1XXXXXXXXXX,n(success),Set(ODBC_FAXDEBIT(${CDR(accountcode)})=\(${FAXPAGES} * .25\)) |
| 02:44.46 | jpsharp | And that was what's in extensions.conf |
| 02:49.38 | Sean-Der | Thanks jpsharp! I see you escaped your set \(${FAXPAGES} * .25\) , also you have whitespace so it should work I just don't get why whitespace is killing my queries. Do you know if Asterisk performs the math or does your DB, maybe it just sends a float to your DB. |
| 02:52.44 | ChannelZ | it's not an expression |
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| 02:56.35 | jpsharp | In my case, Asterisk does the math. |
| 02:56.58 | jpsharp | And I just drop a floater into the db. |
| 02:57.30 | jpsharp | Er, that wasn't the best choice of words : |
| 02:57.32 | jpsharp | : |
| 02:57.33 | jpsharp | :) |
| 03:01.23 | Sean-Der | Hahah yes it was, well I am way closer now I am getting "to your MySQL server version for the right syntax to use near 'UPDATE voice_allowed SET Language = English' " so the full query is making it to MySQL |
| 03:04.23 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 03:04.29 | Sean-Der | Yea I don't get it, the query is correct :/ Its late and I have gotten to the 'make lazy guesses till it works point blaah' |
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| 06:12.08 | v0lZy | morning |
| 06:12.24 | v0lZy | hey guys, how do i terminate an active confbridge from cli? |
| 06:13.29 | jpsharp | I dont think you can. I think you just have to boot everyone out one at a time. |
| 06:13.48 | v0lZy | how can i do that? |
| 06:14.02 | jpsharp | confbridge kick <conference> <channel> |
| 06:14.32 | v0lZy | how do i see the channels that are connected to the conference? |
| 06:14.54 | v0lZy | ah |
| 06:14.55 | v0lZy | found it |
| 06:15.04 | v0lZy | confbridge list <conferance name> |
| 06:18.04 | v0lZy | hm |
| 06:18.06 | v0lZy | for some reason |
| 06:18.13 | v0lZy | my confbridge hangsup |
| 06:18.17 | v0lZy | ... |
| 06:21.02 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 06:21.05 | schmidts | good morning |
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| 06:46.04 | v0lZy | guys |
| 06:46.07 | v0lZy | can i paste something... |
| 06:47.10 | v0lZy | http://bpaste.net/show/hEWIi1D3AxVnT2plKN5U/ |
| 06:47.28 | v0lZy | this is my announcing system confbridge configuration .. |
| 06:48.34 | v0lZy | for some reason my application hangs... doesnt join me in... but other phones remain joined.. |
| 06:48.38 | v0lZy | waiting for my announcement |
| 06:49.43 | v0lZy | hm |
| 06:49.44 | v0lZy | funny |
| 06:49.46 | v0lZy | now it works again. |
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| 07:32.50 | aruntomar | my asterisk server is not sending the "link" event to adhearsion. how can i troubleshoot that. |
| 07:41.29 | *** join/#asterisk DND (~DND@94.200.7.26) |
| 07:43.07 | DND | hi guys i need help. we have 2 asterisk servers connected via iax. so that server 1 can call extensions on server 2 and vice versa. but here's the problem. server 1 can do that just fine. but when server 2 calls any extension on server 1, call just drops. here's a pastebin. http://pastebin.com/raw.php?i=rzty9udZ |
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| 07:55.24 | pbxMan | hello asterisk people |
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| 08:04.34 | jol | Using AMI Action: Originate, Async: true and ActionID: 4242, I receive a queued response but no OriginateResponse event when the call timed out. Could there be some configuration I have missed? Asterisk 1.8.4 in Debian |
| 08:16.06 | WIMPy | DND: That log says the call was answered. So nothing wrong at that end. |
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| 08:28.35 | DND | WIMPy, but the call was dropped |
| 08:28.50 | DND | i mean i just tried t ring it but it didnt dropped instantly |
| 08:29.17 | jol | (Solved my problem - manager access level was the problem) |
| 08:29.32 | DND | sorry there should be a dot after didnt |
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| 08:29.53 | emate | Hi there |
| 08:30.04 | mealstrom | hi , what for event NewAccountCode? |
| 08:31.02 | mealstrom | there is OldAccountCode and AccountCode fields. They are emty. |
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| 08:39.27 | oquidave | hello am trying to start asterisk at boot up by adding this line...as:2:respawn:/etc/init.d/asterisk restart ....to my /etc/inittab file...but it's so far failing to start |
| 08:51.42 | emate | is it possible to establish dial with external phone and one of agents in queue, so first available agent can pick up and talk? |
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| 09:17.33 | RZero | Hi all, has any one come across this warning before http://pastebin.com/92Yr2CSx ? |
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| 09:34.03 | nirWana | rules |
| 09:34.09 | nirWana | Hi |
| 09:35.05 | nirWana | I have a question about haw to get asterisk working with dahdi_dummy module |
| 09:36.16 | nirWana | I'm using asterisk 1.8.13 on a Linux Debian Squeeze with kernel 2.6.32-5 |
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| 09:38.10 | nirWana | I installed all packages about dahdi (dahdi-linux, dahdi-source, dahdi, asterisk dahdi). I'm trying to load dahdi_dummy modules to make running MeetMe(), but no modules found. |
| 09:38.30 | nirWana | Either dahdi, not dahdi_dummy |
| 09:38.43 | nirWana | Any idea? |
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| 09:48.21 | mcolombo | hi all |
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| 11:19.10 | giany | Anyone can explain me what this means : AST_STATE_OFFHOOK |
| 11:20.11 | leifmadsen | a devices state went off hook |
| 11:20.16 | leifmadsen | (receiver picked up) |
| 11:20.57 | leifmadsen | nirWana: if you install a recent version of dahdi, then you just need to load dahdi since dahdi_dummy as a module no longer exists (part of the internals now) |
| 11:21.17 | leifmadsen | if the dahdi module is loaded, then you're done |
| 11:23.26 | giany | thx leifmadsen |
| 11:31.13 | *** join/#asterisk [ProB]CrazyMan (~chatzilla@mx40.roterschnee.com) |
| 11:33.06 | [ProB]CrazyMan | hi, I have an problem with the dial command, when I call an number which does not exist normaly you can heare the anouncment from the provider "this numer does not exist...." but with asterisk i jus here this voice vor one second and then I get the hangup signal |
| 11:33.38 | [ProB]CrazyMan | is there an possibility to set an time how long I here the providers message? bevore I process the hangup signal? |
| 11:33.57 | WIMPy | That's a lot alread. What do you call out on? |
| 11:34.11 | WIMPy | +y |
| 11:34.37 | [ProB]CrazyMan | hi WIMPy, I use still LCR |
| 11:35.02 | WIMPy | Ah |
| 11:35.22 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 11:35.24 | [ProB]CrazyMan | but I didn't find an setting in LCR or in the Dial cmd to fix this |
| 11:35.45 | WIMPy | The bad thing is that my provider fails to provide announcements as well, do it's hard for me to say. |
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| 11:36.36 | [ProB]CrazyMan | I get " == Everyone is busy/congested at this time (1:0/0/1)" |
| 11:36.45 | [ProB]CrazyMan | and then its over |
| 11:37.30 | [ProB]CrazyMan | isnt there a way to have an fixed time to let the line open and here the providers announcments or his ringsignals ? |
| 11:37.35 | WIMPy | The rel trouble is that Asterisk doesn't support disconnected channels. |
| 11:38.28 | WIMPy | I usually get the announcement from LCR. |
| 11:38.56 | din3sh | hey ppl |
| 11:39.07 | WIMPy | (but that's only for phones connected to it, off course) |
| 11:39.12 | [ProB]CrazyMan | so LCR has to forward the signal, is there an possibility to delay the hangup |
| 11:39.33 | WIMPy | No |
| 11:40.07 | WIMPy | There is one for dahdi, but I couldn't make it work there anyway, either. |
| 11:40.12 | [ProB]CrazyMan | so no way to here that anouncements |
| 11:40.45 | WIMPy | I don't thnk so. |
| 11:42.04 | WIMPy | But it's a missing feature I've been talking about for a looong time. |
| 11:42.57 | [ProB]CrazyMan | but I do not understand is this behavior a problem from asterisk or from the providers? |
| 11:43.09 | WIMPy | Asterisk |
| 11:43.16 | [ProB]CrazyMan | ok |
| 11:43.40 | WIMPy | And it's a general issue with SIP as well, as far as I understand. |
| 11:43.44 | [ProB]CrazyMan | and it is not solved because nobody cares? |
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| 11:44.04 | leifmadsen | [ProB]CrazyMan: because nobody has provided a patch |
| 11:44.14 | WIMPy | That seems to be the way it is, yes. |
| 11:44.31 | leifmadsen | or it hasn't been reported and been made high enough priority for a Digium developer to resolve it |
| 11:44.34 | [ProB]CrazyMan | leifmadsen: thats the same, |
| 11:44.51 | leifmadsen | has someone reported the issue before? or just moaned about it in here? |
| 11:44.57 | WIMPy | Yes as I said a few days ago. I'd like to see two new channelstates. |
| 11:45.21 | WIMPy | Well, it's clearly a new feature and "not a bug". |
| 11:47.36 | [ProB]CrazyMan | so I'm not so good in all that dialing state, is there an standard? because asterisk tells me its busy/congested (and there is an voice message) so is there an different signaling type coming in and asterisk takes it for the same state as busy/congested? |
| 11:48.21 | WIMPy | That's jsut the standard message. For real information, you need to look at ${HANGUPCAUSE}. |
| 11:48.37 | *** join/#asterisk robl^ (~robl^@pdpc/supporter/active/robl) |
| 11:50.02 | WIMPy | And BTW: I think it would be good if announcements after hangup couldn;t only be received, but would be possible in the dialplan as well. |
| 11:50.09 | *** join/#asterisk jkroon (~jkroon@dsl-244-22-108.telkomadsl.co.za) |
| 11:50.19 | WIMPy | That's probably a bigger feature. |
| 11:50.54 | jkroon | is there any way to log which dahdi channel answered a call? Eg, when I send a call to DAHDI/r1/number, I'd like to know which BRI channel eventually answered the call ... ? |
| 11:52.09 | WIMPy | Are you asking about channels or do you have multiple interfaces? |
| 11:52.20 | din3sh | I keep getting this "chan_dahdi.c: PRI got event: HDLC Bad FCS (8) on D-channel of span 1" |
| 11:53.00 | din3sh | using a TE220 card |
| 11:53.16 | din3sh | I have checked for missed IRQs, but havent found any |
| 11:53.16 | WIMPy | Sounds like a bad line or interrupt issues. |
| 11:54.37 | din3sh | have run dahdi_test for over 5 hours with results "Best: 100.000% -- Worst: 99.862% -- Average: 99.995729%" |
| 11:54.49 | din3sh | dahdi_tool shows no IRQ missed |
| 11:55.00 | din3sh | might be the ISDN cable? |
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| 11:55.33 | WIMPy | It might. Doesn't hurt to try another cable. |
| 11:55.44 | din3sh | oki |
| 11:56.20 | [ProB]CrazyMan | WIMPy: so if I want to have better understanding for the calling party... I have to make my own voice messages for the 50 Hangup causes ? |
| 11:57.21 | WIMPy | You could just copy them from LCR. |
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| 11:58.36 | nirWana | Thanks for your answer. In fact my issue is that dahdi module seems not avilable as a debian package. I got source from digium, compiled all needed ans installed it and it worked! |
| 11:59.13 | *** join/#asterisk magwas (~mag@mag.webhome.hu) |
| 12:00.18 | magwas | is it possible to use asterisk for conferencing including chat (and preferably also video) on sip? |
| 12:01.26 | magwas | I see there is meetme module but does it support text messaging? |
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| 12:09.04 | leifmadsen | magwas: no |
| 12:09.37 | leifmadsen | you should use an XMPP server for that |
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| 12:25.36 | *** join/#asterisk afournier (~alex@ven69-2-82-228-116-153.fbx.proxad.net) |
| 12:25.45 | afournier | hi |
| 12:27.31 | afournier | does anyone know why old misdn v1 packages are not available anymore ? the last url i have looked like this http://www.misdn.org/downloads/releases/mISDN-1_1_9_2.tar.gz |
| 12:28.18 | WIMPy | Because we have had v2 for many years? |
| 12:28.41 | afournier | even if v1 is damn old, it's still used |
| 12:29.06 | WIMPy | heard about that. |
| 12:30.18 | WIMPy | There are (at least) two forks of the old version. |
| 12:31.15 | *** join/#asterisk rhce7320 (~rhce7320@59.167.200.141) |
| 12:31.18 | WIMPy | Hmm, yes. The original seems to have completely vanished. |
| 12:33.33 | WIMPy | Links updated |
| 12:34.58 | jkroon | hi guys, I've got an NT2 continuously giving me HDLC errors and dropping calls. Using digium B410P with dahdi-linux and dahdi-tools 2.6.1, ast 1.8.15.1 and libpri 1.4.12. |
| 12:35.46 | jkroon | http://pastebin.com/mFTw3L9A contains the span's config for the span that's causing this - any ideas on fixing this? |
| 12:35.57 | WIMPy | Oh, right, in addition to a bad line or IRQ issues it could be the timing configuration. |
| 12:38.48 | jkroon | WIMPy, aimed at me? |
| 12:39.13 | afournier | so mISDN 1.1.9.2 has gone forever ? |
| 12:39.14 | jkroon | according to the client it worked fine with the previous system (same Digium card but using mISDN) |
| 12:39.24 | jkroon | no idea what version of mISDN. |
| 12:39.45 | WIMPy | jkroon: Yes |
| 12:39.48 | afournier | jkroon: i was talking about the tar.gz on misdn.eu |
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| 12:40.45 | WIMPy | jkroon: Well, with misdn you don't have all the options dahdi offers to shoot yourself in the foot. |
| 12:40.50 | jkroon | WIMPy, other ports on that card is working find, so I think we can eliminate IRQ issues. Also, I'm taking timing *from* the PSTN (ie, i'm the slave), so as I understand it timing should be OK ... how can I confirm? |
| 12:40.52 | WIMPy | Like timing configuration. |
| 12:41.21 | WIMPy | That's your dahdi/system.conf |
| 12:41.33 | WIMPy | Are the two lines in sync? |
| 12:41.56 | jkroon | they're connected to the same provider, different NTUs though. |
| 12:42.01 | jkroon | http://pastebin.com/yf8L9Bue |
| 12:42.15 | jkroon | span 1 + 3 is connected to two NT1 units, span 2 is connected to an NT2. |
| 12:42.28 | WIMPy | AFAIK the HFC-[48]S should be able to hande unsyncronised lines, but I have NFI if the driver needs to enable that. |
| 12:42.59 | WIMPy | Looks ok to me |
| 12:43.12 | jkroon | wcb4xxp 0000:04:01.0: Did not do the highestorder stuff <-- this is the ONLY thing from dmesg that raises an eyebrow. |
| 12:43.31 | jkroon | WIMPy, what else can I check - i've checked everything else that makes sense to me. |
| 12:43.38 | WIMPy | What ever that means. |
| 12:43.50 | jkroon | exactly why it raises an eyebrow. |
| 12:43.57 | WIMPy | The hardware is all the same? |
| 12:44.14 | WIMPy | All lines terminated corectly? |
| 12:45.45 | WIMPy | afournier: You could try to ask on the isdn4linux mailing list if you want the official old version outside of an kernel that had it included. |
| 12:46.27 | *** part/#asterisk giany (~giany@shifu.x83.org) |
| 12:46.39 | afournier | WIMPy: only the v2 is included no ? |
| 12:46.59 | WIMPy | But if you want to use it with the old chan_misdn, you might want to try the digium branch. |
| 12:47.23 | WIMPy | afournier: Depends on the version. |
| 12:47.23 | afournier | the digium branch... never heard of it |
| 12:47.57 | WIMPy | http://voice.yeti.dk/Asterisk_vs_ISDN/6 |
| 12:48.28 | afournier | ok thanks |
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| 12:50.25 | jkroon | WIMPy, B410P Digium card, 4 ports, the NTs are 2 x NT1 on port 1+3, NT2 on port 2. |
| 12:51.34 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 12:51.37 | jkroon | all spans comes up OK according to dahdi_scan (/proc/dahdi/*) and the spans are "Up, Active" on asterisk's side, so to the best of my knowledge it's terminated correctly. |
| 12:54.42 | jkroon | WIMPy, can the "Remote type: Unknown node type" have anything to do with the problem? |
| 12:56.18 | WIMPy | Where do you get that? |
| 12:56.25 | jkroon | pri show span 2 |
| 12:57.14 | jkroon | WIMPy, http://pastebin.com/mFTw3L9A |
| 12:58.06 | WIMPy | On the only box running dahdi I don't have that information. |
| 12:58.19 | WIMPy | What do the other spans show there? |
| 12:58.40 | WIMPy | Oh, no got it. |
| 12:58.58 | jkroon | also unknown |
| 12:59.28 | WIMPy | Looks like that should be network. But as the Type looks correct, I don;t see what else there might be to configure. |
| 13:01.02 | WIMPy | What about the termination thing? |
| 13:01.38 | jkroon | signalling=bri_cpe_ptmp |
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| 13:01.59 | WIMPy | The physical line termination. |
| 13:02.11 | WIMPy | Termination resistors |
| 13:02.13 | jkroon | if the line was dodgy then surely mISDN would have given issues too? |
| 13:02.28 | WIMPy | Or jumpers on the card / the NT. |
| 13:02.45 | WIMPy | So it was only a software change? |
| 13:02.50 | jkroon | hmm, as in someone flipped the resistor switches ... no, because when swapping the cables on ports 1 + 2 the problem moves with the NTu |
| 13:02.52 | jkroon | correct. |
| 13:03.25 | jkroon | but upgraded the entire OS everything to newer stuff, so very hard to do a flip one thing at a time back ... or I would have been doing that already. |
| 13:04.21 | WIMPy | Well, FCS is at maximum at driver level. |
| 13:08.44 | *** join/#asterisk rhce7320 (~rhce7320@59.167.200.141) |
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| 13:17.01 | jkroon | WIMPy, FCS? (sorry for delay in responses, phones are going crazy this side) |
| 13:18.09 | WIMPy | Oh, sorry, that was din3sh with the FCS issue. |
| 13:18.14 | jkroon | ok, but I'm not getting "Bad FCS", I'm getting HDLC Abort ... ah ok. |
| 13:18.26 | WIMPy | But HDCL would be at the same level. |
| 13:18.46 | jkroon | ok, so how/where do I flip what to get that to a "lower" level to see if I can get away with that? |
| 13:18.59 | WIMPy | Or it should be below, actually, if you have hardhdlc configured. |
| 13:19.36 | jkroon | ok, what alternative do I have to hardhdlc ? |
| 13:20.07 | jkroon | hardhdlc=3,6,9,12 for the four spans in dahdi/system.conf |
| 13:20.52 | *** join/#asterisk bchia (~Adium@nat/digium/x-gprphlehgsijnnpg) |
| 13:20.57 | WIMPy | "dchan" would make it in the driver instead of on the chip. |
| 13:21.26 | WIMPy | So hardhdlc should be the better choice. |
| 13:21.34 | jkroon | http://docs.tzafrir.org.il/dahdi-tools/README.html <-- according to tzafrir (which I consider to be rather knowledgeable) seems to indicate that you HAVE to use hardhdlc for b410p |
| 13:21.46 | jkroon | "BRI spans serviced by the wcb4xxp driver must use hardhdlc channels for the signalling channels. " |
| 13:22.28 | WIMPy | I'm not sure you must, but I'm pretty sure you want to. |
| 13:22.37 | jkroon | makes sense |
| 13:23.40 | jkroon | hmm, that README.html is pretty insightful. |
| 13:23.51 | jkroon | going to give dspan a shot, just for testing... |
| 13:24.14 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 13:24.18 | WIMPy | dchan |
| 13:24.36 | jkroon | that one :) |
| 13:25.02 | WIMPy | But I'm pretty sure misdn versions will use the framer on the chip. |
| 13:26.09 | *** join/#asterisk rcaskey (~Rob@dumbledore.athenshousing.org) |
| 13:26.12 | jkroon | nethdlc looks handy too if you want to use your dahdi card as a cheap WAN router. |
| 13:26.14 | jkroon | very interesting. |
| 13:26.42 | WIMPy | Yes, that's the advantage of dahdi. |
| 13:26.51 | rcaskey | hey all, i'm trying to figure out what kind of phones to buy and I keep hearing glowing reviews of Polycom stuff. Are there in fact any down-sides or better alternatives to the IP 550? |
| 13:29.00 | [TK]D-Fender | rcaskey, IP 550 isn't a 650 (no expansion), and it's only plus is the large screen. If you need a regular phone I'd pick a lower model (321/335/450), or aim higher |
| 13:29.13 | [TK]D-Fender | rcaskey, 550 specifically is an odd-duck |
| 13:29.33 | jkroon | WIMPy, with dchan= the channel won't even come up. |
| 13:29.51 | Katty | [TK]D-Fender: query. |
| 13:29.57 | *** join/#asterisk serafie (~erin@nat/digium/x-ywrrsqncggegwwfo) |
| 13:30.04 | [TK]D-Fender | Katty, arg[-1] |
| 13:30.13 | Katty | [TK]D-Fender: absolute zero....theoretically means atoms aren't moving. |
| 13:30.24 | Katty | [TK]D-Fender: but what about the electrons? |
| 13:30.26 | *** part/#asterisk kontinuity (~Adium@122.178.203.210) |
| 13:30.29 | Katty | [TK]D-Fender: and if they also stop moving.... |
| 13:30.32 | [TK]D-Fender | Katty, You flunked the theory ... see if you can at least nail the practical ;) |
| 13:30.44 | rcaskey | [TK]D-Fender, what do you mean by expansion? |
| 13:30.51 | rcaskey | you can't plug in sidecars? |
| 13:30.51 | Katty | [TK]D-Fender: what happens to that 90% of volume it creates. |
| 13:30.58 | [TK]D-Fender | rcaskey, Attendant consoles |
| 13:31.02 | [TK]D-Fender | rcaskey, USB, etc |
| 13:31.17 | rcaskey | [TK]D-Fender, ok my mind is blown...what do you use uSB for? |
| 13:31.24 | jkroon | [TK]D-Fender, any experience with BRIs ? |
| 13:31.39 | [TK]D-Fender | Katty, Lack of movement means that atoms can sit CLOSER and takes up LESS volume. Except for ice..... |
| 13:31.44 | [TK]D-Fender | water = wierd |
| 13:32.00 | [TK]D-Fender | jkroon, Nope, I'm on the other side of the water... |
| 13:32.31 | jkroon | [TK]D-Fender, two words: lucky you. |
| 13:32.42 | [TK]D-Fender | rcaskey, Go read up on the IP 650+ for the goodies that come at that price-point. |
| 13:33.10 | jkroon | WIMPy, any other ideas? |
| 13:33.46 | rcaskey | [TK]D-Fender, it's comfortably within my price range so i'm not concerned about that |
| 13:33.57 | [TK]D-Fender | rcaskey, What do you actually need in a phone? |
| 13:34.05 | rcaskey | I need to impress my bosses boss |
| 13:34.13 | rcaskey | but it can't be too showy either |
| 13:34.21 | WIMPy | jkroon: Even if it worked before, I'd check for correct termination. If it's wrong, jst changing a cable might make the difference beween wors and doesn't work. |
| 13:34.26 | rcaskey | if it kinda looked like it was a Thinkpad that would be a plus |
| 13:34.28 | rcaskey | :P |
| 13:35.00 | rcaskey | but seriously we have very basic on-the-desk phone needs and a decent budget |
| 13:35.02 | WIMPy | jkroon: Do you have any real issues, BTW? Maybe dahdi is just more verbose? |
| 13:35.15 | jkroon | yes, calls are dropping. |
| 13:35.33 | [TK]D-Fender | rcaskey, Going to have dedicated wiring? What is your lighting like? Any headset usage? What kind of call-colume? |
| 13:35.37 | jkroon | and a small percentage of incoming calls are failing (not even seeing them in asterisk) |
| 13:35.38 | WIMPy | Ok. So there is a real difference. |
| 13:35.50 | jkroon | afraid so ... |
| 13:36.21 | WIMPy | Definitely not just the occasional glitch. |
| 13:36.59 | rcaskey | [TK]D-Fender, dedicated CAT 5, lighting is about average, could be a bit better but its not bad in most places but a lot of phones sit in a shadow of a credenza. Headset usage only on the front desk. Call volume ... depends how loud they talk...no...most phones do maybe 30 min a day? |
| 13:38.28 | [TK]D-Fender | rcaskey, IP 335. |
| 13:38.57 | rcaskey | polycoms website is slooooowww :( |
| 13:39.04 | rcaskey | good old amazon to the rescue |
| 13:39.43 | rcaskey | also were doing POE, not that pretty much every phone doesn't come in either option |
| 13:40.22 | *** join/#asterisk serafie (~erin@nat/digium/x-hpcgkvxayhouaoyh) |
| 13:40.34 | rcaskey | [TK]D-Fender, screen is a bit smaller than what my users already have os i'm afraid they will see it as a down-grade |
| 13:40.53 | rcaskey | I know it's dumb but i think it's probably accurate |
| 13:41.20 | [TK]D-Fender | rcaskey, I had an IP 600 at my desk and an Aastra 57i CT with a giant-ass screen. I traded FOR an IP 335 |
| 13:41.49 | [TK]D-Fender | rcaskey, At least on this I can read the time without budging and the form factor does waste desk space. As you said, 30min/day |
| 13:43.11 | AkkerKid | What's the easiest way to provision deskphones over the internet and have some level of safety? SFTP? FTPS? surely,not TFTP... |
| 13:44.03 | rcaskey | [TK]D-Fender, i know, but form is sometimes more important than reality to end-users |
| 13:44.06 | [TK]D-Fender | AkkerKid, You seem to already know the answer..... |
| 13:44.07 | mcolombo | exit |
| 13:44.11 | *** part/#asterisk mcolombo (~mcolombo@212.29.130.12) |
| 13:44.32 | rcaskey | [TK]D-Fender, I will say the Axxess phones look & feel great considering their age |
| 13:44.32 | *** join/#asterisk mcolombo (~mcolombo@212.29.130.12) |
| 13:44.40 | mcolombo | hi all |
| 13:45.08 | AkkerKid | Is there a utility that'll create the required files that I can stick on? |
| 13:45.11 | rcaskey | [TK]D-Fender, i think we definately need some dedicated line busy button/LED pairs |
| 13:45.28 | WIMPy | AkkerKid: Where's the difference? If you know (or guess) the url, you get the data. What does it matter if someone elce can see the traffic? |
| 13:45.47 | [TK]D-Fender | rcaskey, How many is your idea range? |
| 13:46.43 | AkkerKid | WIMPy: I'm not worried about the voice being tapped. I just want the phone to be set up with minimal configuration on the remote end and not hand the extension passwords to the world. |
| 13:46.52 | mcolombo | i have a problem with asterisk 10.9.0-rc1 and Call Hold |
| 13:46.58 | mcolombo | My problem is that Asterisk don't send re-invite when i pick up call from hold. |
| 13:47.05 | mcolombo | I already insert canreinvite=no in all my sip channels, set dtmfmode=info in sip.conf and my Dial() command dont insert option like t, T", "h", "H", "w", "W" or "L" (with multiple arguments). |
| 13:47.06 | WIMPy | AkkerKid: That's what I was talking about. |
| 13:47.16 | mcolombo | I run debug with asterisk, and i see that the re-invite are made by asterisk, but in the TO is present the IP of the asterisk server. |
| 13:47.18 | mcolombo | This is the log : http://pastebin.com/ARUC0j4t |
| 13:47.23 | mcolombo | It a bug? i'm already search on google, but i dont find anything. |
| 13:47.25 | mcolombo | Thanks |
| 13:47.56 | mcolombo | about a bug |
| 13:48.53 | *** join/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 13:49.01 | AkkerKid | WIMPy: there's an endpoint configurator within Elastix but it's taylored to TFTP and doesn't allow for phones that are outside of the LAN subnet. It used NMAP. |
| 13:49.16 | WIMPy | mcolombo: Can you please describe what you're doing, what happens and what you would like to happen (and why)? |
| 13:51.45 | [TK]D-Fender | ideal* |
| 13:52.25 | [TK]D-Fender | AkkerKid, You just went from asking about specific protocol to some other builder app for the actual configs... |
| 13:52.58 | AkkerKid | WIMPy: I guess what I need is a simple interface to create config files for phones where I type in the MAC, the interface on the server it'll be connecting from and it's extension and it should make those config files available to the phone via FTP |
| 13:53.47 | *** join/#asterisk qbx (0c19f064@gateway/web/freenode/ip.12.25.240.100) |
| 13:56.21 | WIMPy | AkkerKid: Write one. |
| 13:56.30 | AkkerKid | EEK! |
| 14:01.25 | RZero | Hi all, has any one come across this warning before http://pastebin.com/92Yr2CSx ? |
| 14:03.02 | WIMPy | RZero: Yes, you should be able to find something on http://issues.asterisk.org/jira/ |
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| 14:05.53 | RZero | thanks WIMPy I will take a look |
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| 15:04.10 | TimeRider | After a bunch of cheap priced reliable sip office phones, (preferably branded) that will work with Asterisk, in the UK.... Anyone any ideas? |
| 15:04.58 | Qwell | SIP phones don't really care about what country they're in. |
| 15:05.56 | Qwell | TimeRider: You should definitely buy Digium phones. I'm not biased at all. http://www1.digium.com/en/products/phones |
| 15:06.17 | mcolombo | WIMPy: call -> set hold -> cancel hold -> bye |
| 15:06.35 | mcolombo | when i cancel hold, the re-invite |
| 15:07.33 | *** join/#asterisk jeffspeff (~jeffspeff@12.49.160.131) |
| 15:07.46 | mcolombo | made from asterisk, in the filed TO there is my IP and NOT the next hot IP |
| 15:07.57 | mcolombo | hop* |
| 15:08.56 | mcolombo | so, the re-invite is not sent from asterisk |
| 15:09.01 | mcolombo | to the next hop |
| 15:09.58 | WIMPy | I'm not so sure about the country thing. |
| 15:12.11 | mcolombo | What do you mean with "country thing." |
| 15:12.13 | mcolombo | thanks |
| 15:15.55 | mcolombo | cancel hold = come back to communication |
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| 15:36.11 | *** join/#asterisk sss_chris (~chris_not@chris-notebook.ghb.hs-furtwangen.de) |
| 15:36.44 | sss_chris | hi |
| 15:36.54 | sss_chris | someone of you already worked with vtestcall? |
| 15:37.10 | sss_chris | or better say the iaxclient api |
| 15:37.27 | ska | Can I have simultaneous SIP and IAX2 registrations? |
| 15:37.48 | sss_chris | yes i think so |
| 15:37.57 | Qwell | ska: yes.. |
| 15:38.11 | sss_chris | i just worked with asterisk 3 weeks now but i think i already got that |
| 15:38.43 | ska | ok.. If you have a sample config file, please share. |
| 15:38.55 | sss_chris | i can give you one tomorrow at work |
| 15:39.07 | ska | cool.. ty. |
| 15:39.09 | sss_chris | np |
| 15:39.42 | ska | I realize now that writing a good call plan and conf is an art. |
| 15:40.10 | sss_chris | hm you just have to use some features to make it a little bit more compact |
| 15:40.44 | sss_chris | I read about that some people dont like macros but I think they reduce the content of the call plan a little bit |
| 15:40.54 | sss_chris | or using patterns is nice too=) |
| 15:41.38 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.111) |
| 15:42.49 | sss_chris | but like i said, i just started working with voip for 3 weeks |
| 15:44.20 | Qwell | sss_chris: Please do not message people without their permission. If you'd like private help, my fee is $299/hour, 4 hour minimum. |
| 15:44.30 | sss_chris | *g |
| 15:44.32 | sss_chris | ok |
| 15:44.49 | ska | Ouch... Still $299/hour is a bargain :) |
| 15:45.01 | sss_chris | ska: when are you online tomorrow? |
| 15:45.21 | ska | sss_chris: usually all day, but sometimes ... I'll PM you :) |
| 15:45.26 | sss_chris | nice |
| 15:45.34 | [TK]D-Fender | <ska> Can I have simultaneous SIP and IAX2 registrations? <--- these are completely separate. Yes, I can have a car and also haev dishes |
| 15:45.37 | [TK]D-Fender | have* |
| 15:46.36 | *** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu) |
| 15:46.48 | [TK]D-Fender | sss_chris, Who says you can't use patters AND macros? |
| 15:47.00 | sss_chris | i read about that some people dont like macros |
| 15:47.10 | sss_chris | thats what i meant |
| 15:47.26 | [TK]D-Fender | Some people are also twits. I'm suspecting a certain venn overlap..... |
| 15:49.20 | keith4 | [TK]D-Fender: thanks, I almost choked to death on my coffee |
| 15:49.34 | sss_chris | i read about that at a german site for asterisk |
| 15:51.49 | sss_chris | so im right that macros are really nice? :) |
| 15:52.15 | [TK]D-Fender | keith4, You don't choke on coffee.. you drown ;) unless it's iced coffee and you somhow keep the tmperature long enough to maintain a solid blockage ;) |
| 15:52.41 | [TK]D-Fender | sss_chris, I think you should just learn the dialplan and learn where macro's work best for you |
| 15:53.21 | sss_chris | i just use it for things i need very often:) |
| 15:53.33 | [TK]D-Fender | sss_chris, Isn't that the point? |
| 15:53.40 | sss_chris | yeah=) |
| 15:54.05 | *** join/#asterisk Galen (~Galen@rrcs-24-43-20-118.west.biz.rr.com) |
| 15:56.53 | sss_chris | someone could give me tips about why i have problems with video codec negotiation?:) |
| 15:57.00 | sss_chris | plz^^ |
| 15:57.32 | sss_chris | could someone* |
| 15:57.36 | WIMPy | Well, macros are officially deprecated and should be replaced by gosubs. |
| 15:57.46 | WIMPy | But I'm still using macros as well. |
| 15:57.55 | sss_chris | ok |
| 15:58.16 | Qwell | WIMPy: false |
| 15:58.20 | [TK]D-Fender | sss_chris, You should probably be showing use your actual calls that are having video codec negotiation issues.... |
| 15:58.35 | sss_chris | but anyway its not important for me to get a really nice dial plan, i need to create an iax voip client |
| 15:59.00 | sss_chris | what you meant? |
| 15:59.01 | sss_chris | :X |
| 15:59.05 | WIMPy | Qwell: Any pointers? I'd like to be proven wrong there, but that's what I learned so far. |
| 15:59.41 | Qwell | TimeRider: The msg thing goes for you as well. |
| 15:59.52 | Qwell | WIMPy: app_macro is still core supported. |
| 16:00.24 | *** join/#asterisk navaismo (~navaismo@189.144.58.34) |
| 16:00.41 | sss_chris | my task atm is to get the vtestcall project of the iax api for c running on 2 pc's with a working video connection |
| 16:01.03 | sss_chris | and at this point i get problems with video codec negotiation |
| 16:01.21 | Qwell | What is vtestcall? |
| 16:01.39 | Qwell | oh, IAXclient. Yeah. Good luck with that. |
| 16:01.56 | sss_chris | :) |
| 16:02.00 | Qwell | You should note that iaxclient doesn't actually support IAX2... |
| 16:02.02 | sss_chris | yes thats the point |
| 16:02.13 | sss_chris | really? |
| 16:02.29 | sss_chris | i mean, voice communication is running |
| 16:02.31 | Qwell | I don't believe so, no |
| 16:02.33 | sss_chris | only problem is video |
| 16:03.12 | sss_chris | i have to say im just a student atm and we need to work for 6 months |
| 16:03.21 | sss_chris | i never worked with voip before |
| 16:04.23 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 16:05.05 | navaismo | https://github.com/versatica/JsSIP |
| 16:05.20 | sss_chris | hm i read that at voip-info: "IAXClient: IAX2 softphone library" |
| 16:05.36 | ska | Does anyone know where I can find Paul Mahler's example files from his book? |
| 16:06.05 | Qwell | ska: who? |
| 16:06.47 | ska | Paul Mahler wrote a book a few years back on *. |
| 16:06.54 | Qwell | Throw it away. |
| 16:06.56 | Qwell | ~book |
| 16:06.56 | 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 |
| 16:07.17 | ska | Qwell: that bad eh.. |
| 16:07.24 | Qwell | It was published in 2004... |
| 16:07.40 | [TK]D-Fender | ska, And why is his book in particular of any elevated importance? |
| 16:08.37 | ska | Just want to use some of the examples.. |
| 16:08.48 | Qwell | His examples are wrong. |
| 16:08.54 | Qwell | Any examples written in 2004 are wrong. |
| 16:09.14 | ska | My * Oreilly 2nd edition is probabably ancient too. |
| 16:09.19 | [TK]D-Fender | 2004 ... is like ... Pre WHOLE NUMBERED ASTERISK |
| 16:09.34 | Qwell | ska: See the link above. |
| 16:09.41 | *** join/#asterisk qbx (0c19f064@gateway/web/freenode/ip.12.25.240.100) |
| 16:11.22 | ska | Qwell: thanks.. [TK]D-Fender thanks too. |
| 16:12.21 | sss_chris | Qwell: do you have an idea why i get video codec negotiations and what i could look for? |
| 16:12.48 | Qwell | sss_chris: No. You'll need to modify iaxclient to do what you need. |
| 16:13.06 | sss_chris | i only heard about that you shouldnt use more than one video codec on asterisk |
| 16:13.13 | sss_chris | to solve that, but i just use h264 |
| 16:13.31 | *** join/#asterisk beefcafe (~beefcafe@unaffiliated/beefcafe) |
| 16:15.07 | [TK]D-Fender | sss_chris, I'd lay bets you get video negotiations... because you ALLOW the codecs..... |
| 16:15.37 | sss_chris | what you mean? i set disallow=all before allowing h264 |
| 16:16.02 | [TK]D-Fender | facepalms |
| 16:16.45 | sss_chris | i mean |
| 16:16.59 | sss_chris | i get information that he fails at video codec negotiation |
| 16:17.00 | sss_chris | :) |
| 16:17.05 | sss_chris | sry |
| 16:17.48 | [TK]D-Fender | [11:58] <[TK]D-Fender> sss_chris, You should probably be showing use your actual calls that are having video codec negotiation issues.... |
| 16:18.11 | sss_chris | what should i show you exactly? |
| 16:18.24 | [TK]D-Fender | "your actual calls " |
| 16:18.31 | sss_chris | you mean the logs? |
| 16:18.42 | [TK]D-Fender | o.o |
| 16:19.10 | sss_chris | sry |
| 16:21.33 | *** join/#asterisk troyt (~troyt@2001:1938:240:3000::3) |
| 16:22.41 | beefcafe | call test anyone? |
| 16:23.57 | *** join/#asterisk cmendes0101 (~cmendes01@wtnl.corp.tierra.net) |
| 16:26.05 | WIMPy | doesn't have an extension named test. |
| 16:26.26 | navaismo | is Asterisk plannig to use the new OPUS codec? |
| 16:26.50 | Qwell | navaismo: There has been plenty of discussion of that on the various mailing lists. |
| 16:26.52 | beefcafe | test: sip:000777@iptel.org |
| 16:28.01 | navaismo | hmmm hate mailing list, maybe its time to subscribe |
| 16:28.07 | navaismo | thanks Qwell |
| 16:28.24 | navaismo | going to wash dishes brb |
| 16:32.27 | *** join/#asterisk aross42 (~aross@CPE009400809a9c-CMb89bc9d2e1a5.cpe.net.cable.rogers.com) |
| 16:39.02 | *** join/#asterisk onixx (1000@bas1-stetherese38-2925308144.dsl.bell.ca) |
| 16:40.11 | onixx | hello, I recently upgraded my computer running asterisk. Since this upgrade, I have a weir sound on my TDM400p randomly occuring. It lasts about a second and I can't determine what could be causing this |
| 16:44.42 | sruffell | onixx: does it sound like screeching? |
| 16:45.45 | onixx | sruffell: sort of, very short like swiffff... about a seocnd |
| 16:45.47 | sruffell | one hypothesis is that fax detection is now enabled. |
| 16:45.50 | sruffell | oh... |
| 16:45.54 | sruffell | ...probably not then. |
| 16:46.04 | sruffell | if "swiffffff" is the sound |
| 16:47.12 | onixx | sruffell: the conversation can carry on no problem... but the sound is annoying. I use the same asterisk version, same kernel |
| 16:47.20 | *** join/#asterisk YoMomma (~YoMomma@71-95-164-90.static.mtpk.ca.charter.com) |
| 16:48.03 | onixx | sruffell: I have connected an analog phone direct to line and it does not occur... it definately comes from the tdm400p |
| 16:49.02 | onixx | sruffell: the new machine is an HP Core2 Duo |
| 16:49.04 | sruffell | how do you know it's from the card and not Asterisk? |
| 16:49.25 | sruffell | and is it only on audio received by the card..or audio sent from the card to the PSTN? |
| 16:50.06 | onixx | sruffell: well, basically, thet software on the new pc is an image of the old... other than a kernel recompile to adjust cpu type and multi processing |
| 16:50.36 | *** join/#asterisk gg608f (~Adium@c-67-180-129-182.hsd1.ca.comcast.net) |
| 16:51.11 | sruffell | hmm....but same version of Asterisk and DAHDI? If so...can you try moving the card into another slot? |
| 16:51.41 | onixx | sruffell: I have not had remote notice this (however, I am not 100% on this yet) |
| 16:52.39 | sruffell | ok....could also be IRQ problems (but those normally sound like clicking). TDM400 doesn't have the same resiliency that the TDM410 does. |
| 16:53.09 | onixx | sruffell: sure ! I will try moving the card |
| 16:53.29 | onixx | sruffell: any advice on IRQ problems ? |
| 16:53.40 | sruffell | If it's an SMP system and interrupts might be a problem, you could try isolating the interrupts from the card on it's own CPU and not share with interrupts from other devices. |
| 16:54.05 | sruffell | and ideally one of the higher number cores in case the BIOS is running system management interrupts on the first core. |
| 16:55.24 | sruffell | but all this "interrupts are a problem" are only on the old cards....none of the new cards have these types of problems except in extreme cases (disks in PIO mode, bad framebuffer drivers scrolling in interrupt context, etc... ) |
| 16:57.30 | onixx | sruffell: my card is pretty old, had it for many years... |
| 16:58.06 | sruffell | nods |
| 16:59.50 | *** join/#asterisk Alex_Bkash (3a6198f7@gateway/web/freenode/ip.58.97.152.247) |
| 17:00.19 | onixx | sruffell: I have an output of /proc/interrupt at http://pastebin.com/GbZnwHra |
| 17:00.19 | onixx | sruffell: I am not sure what to do for this point |
| 17:02.57 | Alex_Bkash | i an using iax2 between two endpoints. but bandwidth is not minimized as it shows in asteriskgure bw calculator |
| 17:04.03 | WIMPy | Did you set trunk=yes? |
| 17:05.21 | onixx | sruffell: i guess I will move around the card and see if it helps. I suspect my bttv card might be interfering |
| 17:06.01 | sruffell | onixx: You might need to Google around since I won't be able to completely walk you through the setup...but basically you want to use /proc/irq/<irq num>/smp_affinity to limit all IRQ except the one for the card. |
| 17:06.21 | sruffell | oh..yeah..try just moving the card first in case it's electrical noise. |
| 17:08.05 | onixx | sruffell: I have just stopped my capture card (bttv) and I don't have the noise anymore ! |
| 17:08.15 | sruffell | h |
| 17:08.17 | sruffell | hmm |
| 17:08.24 | Alex_Bkash | http://pastebin.com/4LbAp9TQ |
| 17:08.25 | sruffell | did you have the card in your old system? |
| 17:08.41 | Alex_Bkash | plz check my server IAX2 config http://pastebin.com/4LbAp9TQ |
| 17:09.09 | onixx | sruffell: nope ! this was part of the "merge"... I had no clue IRQ could do this sort of thing !! |
| 17:09.20 | sruffell | it might not be IRQ...could be electrical noise. |
| 17:09.50 | Alex_Bkash | client side iax2 http://pastebin.com/Rzv9d1qh |
| 17:09.58 | onixx | sruffell: ok, I will set them apart, I think they are side by side now |
| 17:10.05 | sruffell | try this...run dahdi_test and see what sort of numbers you get. Then start the capture card along with the wctxxp driver and start dahdi_test again and see if it looks much worse. If so..it's IRQ...if not..it's probably noise. |
| 17:10.47 | sruffell | (and run dahdi_test like "chrt -f 99 dahdi_test" to make sure it gets first crack at the CPU) |
| 17:11.28 | onixx | just dahdi_test without the capture card "capturing", I get Best: 99.998% -- Worst: 99.993% -- Average: 99.996837% |
| 17:12.03 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) |
| 17:12.56 | [TK]D-Fender | Alex_Bkash, careinvite=no<--- this is spelled incorrectly... for a parameter name that does not even EXIST in IAX |
| 17:13.18 | [TK]D-Fender | Alex_Bkash, trnasfer=no <-- you have a SERIOUS problem with spelling... |
| 17:13.29 | [TK]D-Fender | Alex_Bkash, iaxcompate=yes <- MORE |
| 17:13.50 | *** join/#asterisk libryder (~libryder@libryder.com) |
| 17:13.56 | onixx | sruffell: with capture card: Best: 99.997% -- Worst: 99.994% -- Average: 99.996043% |
| 17:14.20 | libryder | i am on 1.8 and getting a segfault when playing GSMs through app_swift, core dump here: https://gist.github.com/4fbb6c2de5f20a177aa7 |
| 17:14.26 | [TK]D-Fender | Alex_Bkash, Do you have DAHDI instaled no both sides? |
| 17:14.43 | Alex_Bkash | no |
| 17:14.44 | onixx | i can see wdtdm shares something with bttv 21: 1877820 1877222 IO-APIC-fasteoi uhci_hcd:usb2, wctdm, uhci_hcd:usb5, bttv1, bttv5 |
| 17:14.50 | [TK]D-Fender | Alex_Bkash, it is REQURIED |
| 17:14.52 | onixx | but results is about the same |
| 17:14.58 | [TK]D-Fender | REQUIRED even. |
| 17:15.20 | sruffell | onixx: so it's probably electrical noise. |
| 17:15.24 | [TK]D-Fender | worries that Alex_Bkash's dyslexia my be communicable over IRC |
| 17:15.26 | Alex_Bkash | i am not sure |
| 17:15.42 | [TK]D-Fender | Alex_Bkash, what is there to not be sure of? |
| 17:15.46 | [TK]D-Fender | These are YOUR systems |
| 17:16.02 | Alex_Bkash | im new |
| 17:16.06 | onixx | sruffell: on, I will move things around and see !!! thanks for all your help, much appreciated |
| 17:16.12 | Alex_Bkash | how can i chk? |
| 17:16.20 | [TK]D-Fender | Alex_Bkash, "new" should not mean you don't even know what you have installed. |
| 17:17.13 | Alex_Bkash | can u plz chk my server? |
| 17:17.40 | [TK]D-Fender | Alex_Bkash, No. You go check if you have it installed. |
| 17:20.29 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 17:20.48 | WIMPy | [TK]D-Fender: Is dahdi still required for IAX trunks? |
| 17:21.22 | Alex_Bkash | im running asterisk 1.8 |
| 17:21.43 | Alex_Bkash | Asterisk 1.8.15.1 |
| 17:24.09 | [TK]D-Fender | I see a ton of SIP SETTINGS in an IAX2 config. Only ONE side of the picture. No idea what's on the other at all. No debug to look at. And a guy who has no clue what he installed on his own server. No, I'm not going to waste my time when even the basics are screwed this badly. |
| 17:24.36 | *** part/#asterisk libryder (~libryder@libryder.com) |
| 17:27.36 | Alex_Bkash | no dhadi is installed |
| 17:27.49 | Alex_Bkash | i just did apt-get install asterisk |
| 17:28.53 | WIMPy | At least in the past you needed dahdi timing for IAX trunks. Not sure if that's still the case. |
| 17:29.50 | file | in recent versions of Asterisk it is not (1.8+, maybe lower - don't have the lower branches checked out) |
| 17:32.52 | *** join/#asterisk oneTurk (~oneTurk@78.171.51.140) |
| 17:33.27 | WIMPy | I would not expect that change before 1.8. |
| 17:35.02 | *** join/#asterisk bipul (~vipul@unaffiliated/bipul/x-4918593) |
| 17:41.01 | *** join/#asterisk lvlolvlo (~lvlolvlo@unaffiliated/lvlolvlo) |
| 17:41.26 | *** join/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 17:42.35 | acidfoo | we have 1800 phones (sip) that re-register every 20sec, and we feel that it too much because sometime a lot of them get disconnect. We feel that the kernel UDP buffer is not being read fast enough by chan_sip and then the kernel has to drop packets... |
| 17:42.50 | acidfoo | it is* |
| 17:43.09 | acidfoo | anyone has experienced something similar ? |
| 17:43.45 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 17:46.19 | *** join/#asterisk AkkerKid (~AkkerKid@50-200-18-202-static.hfc.comcastbusiness.net) |
| 17:47.03 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 17:47.37 | acidfoo | leifmadsen, do you have asterisk installation with > 1800 phones ? and has experienced disconnection ? thank you |
| 17:48.09 | leifmadsen | no |
| 17:48.25 | leifmadsen | with that many phones, I'd have OpenSIPS or something in front |
| 17:48.41 | leifmadsen | 1800 phones at 20s re-reg is crazy |
| 17:49.49 | [TK]D-Fender | 20s re-reg itself = crazy |
| 17:50.35 | *** join/#asterisk vader-- (~me@pool-72-92-38-165.phlapa.fios.verizon.net) |
| 17:50.42 | vader-- | does anyone know off the top of their heads how to make internal calls on polycom 335 and 670 phones with Freepbx 2.10+? |
| 17:50.48 | acidfoo | ok, thanks for the hint, i'm pushing up the information towards the people that are in charge of the installation ;) |
| 17:52.06 | leifmadsen | ya I'd never have allowed 1800 phones to register against a single asterisk box |
| 17:52.15 | leifmadsen | I'd have put a proxy in front long ago :) |
| 17:56.19 | *** join/#asterisk sezuan (bouncer@irc.scheff32.de) |
| 17:57.39 | *** join/#asterisk aross42 (~aross@CPE009400809a9c-CMb89bc9d2e1a5.cpe.net.cable.rogers.com) |
| 17:58.19 | acidfoo | I'm just a developer 5000km away from the people that made that installation :P |
| 17:58.31 | leifmadsen | I'm just telling you what I'd have done :) |
| 17:58.34 | leifmadsen | right or wrong |
| 17:59.10 | acidfoo | yeah :) |
| 17:59.53 | acidfoo | well, thank for you inputs ;) |
| 18:13.57 | *** join/#asterisk sustav (~vpp@nat/digium/x-zipbjdqsyfsjpxre) |
| 18:14.51 | newtonr | vader--: that question is too vague to be answered. however it would be better to pursue that in #freepbx |
| 18:16.04 | vader-- | Do you guys happen to know the default alterinfo values for the polycom phones? |
| 18:16.37 | newtonr | vader--: http://www.voip-info.org/wiki/view/Polycom+auto-answer+config |
| 18:16.59 | newtonr | Googled "polycom alert-info" |
| 18:19.11 | navaismo | LOl next time share the link http://lmgtfy.com/?q=polycom+alert-info |
| 18:19.56 | newtonr | I try not to be too condescending, lol |
| 18:20.12 | vader-- | i did google it and got a bunch of useless info |
| 18:20.20 | vader-- | thats why i was poking around for hands on experience doing it |
| 18:21.21 | *** part/#asterisk ska (~ska@unaffiliated/ska) |
| 18:22.02 | newtonr | vader--: on that note, i wouldn't trust everything you read on voip-info either. In this case, the alert-info stuff should be correct IIRC |
| 18:23.21 | *** join/#asterisk Nemus (~Nemus@12.195.147.119) |
| 18:23.41 | newtonr | a google search like "alert-info site:polycom.com" shows a bit more reliable sources for that question |
| 18:24.05 | vader-- | oh i know |
| 18:24.10 | vader-- | but poorly written faqs |
| 18:24.12 | vader-- | i found this one |
| 18:24.14 | vader-- | http://community.polycom.com/t5/VoIP/FAQ-How-can-I-change-my-Ringtone-or-Ring-in-a-special-manner-for/td-p/5167 |
| 18:24.21 | vader-- | it describes 14 different ring tones |
| 18:24.26 | vader-- | that don't exsist anywhere |
| 18:24.29 | *** join/#asterisk jkroon (~jkroon@dsl-244-22-108.telkomadsl.co.za) |
| 18:24.50 | Qwell | You mean the attachment on the FAQ that you linked? |
| 18:24.53 | Qwell | That doesn't exist? |
| 18:24.57 | vader-- | download it :-) |
| 18:25.03 | vader-- | it has two wav files in it |
| 18:25.07 | *** join/#asterisk aross42 (~aross@CPE009400809a9c-CMb89bc9d2e1a5.cpe.net.cable.rogers.com) |
| 18:25.17 | vader-- | loudring and warble |
| 18:26.10 | navaismo | vader--, did you tried with the standard alert-info that i gave you in fpbx? |
| 18:27.19 | vader-- | they don't work for polycom |
| 18:27.39 | vader-- | i think only older cisco and aastra phones |
| 18:27.41 | navaismo | :( |
| 18:31.32 | newtonr | the "Ring Answer" always worked in the past for me with Polycom |
| 18:32.17 | newtonr | however,last time I messed with that was probably two years back |
| 18:40.15 | jkroon | doees anybody know whether it's possible to get hold of mISDNuser tar archive downloads for v2 somewhere? I'd really prefer to not have to do the git checkout thing. makes it really hard to package without those downloads. |
| 18:41.01 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.111) |
| 18:41.05 | WIMPy | All tars seem to have gone. Both the old and the new :-( |
| 18:42.19 | jkroon | ok, so it's not just me. |
| 18:42.40 | jkroon | ok, jkroon proceeds to screw his install with a make && make install of git variant ... |
| 18:43.06 | *** join/#asterisk autofsckk (~que@unaffiliated/autofsckk) |
| 18:43.07 | WIMPy | I presonally don't care, but I think it's bad not to have release versions. |
| 18:43.29 | WIMPy | Are you trying the third way now? |
| 18:43.51 | autofsckk | hi, how can i transfer a call but i want to talk with the extension owner before he receives the call just to tell who is calling? is it possible? |
| 18:43.51 | jkroon | if the third way is to ditch dahdi and use misdn, then yes. |
| 18:43.54 | *** join/#asterisk af_ (~getsmart@88-149-246-253.v4.ngi.it) |
| 18:44.03 | jkroon | frankly, i lost track of how many different options I tried. |
| 18:44.17 | WIMPy | autofsckk: Using what kind of phone? |
| 18:44.33 | jkroon | autofsckk, it's called an attended transfer and yes, it's possible, and reasonably phone dependent. |
| 18:44.38 | WIMPy | jkroon: But you came from the old misdn, didn't you? |
| 18:45.04 | autofsckk | WIMPy: yealink |
| 18:45.34 | jkroon | yea, but i didn't install that machine ... |
| 18:45.38 | WIMPy | autofsckk: Just place another call and after consulting, press transfer. |
| 18:46.20 | jkroon | so this will be the second time in almost five years that I install misdn, and it seems the process only got worse. |
| 18:46.24 | vader-- | does the alertinfo have to be in " " or <>? |
| 18:46.28 | autofsckk | WIMPy: like putting on hold the call and then place that call and after that transfer? |
| 18:46.50 | WIMPy | autofsckk: yes |
| 18:46.57 | *** join/#asterisk mmcji (~mmcji@65.172.54.10) |
| 18:47.26 | autofsckk | WIMPy: so for that kind of transfers and that stuff i have to see the yealink manual right? is that possible to do with phones connected to pap2 too? |
| 18:47.27 | WIMPy | jkroon: If you (can) use the kernel version it's not that bad. |
| 18:47.43 | jkroon | WIMPy, you still need mISDNuser |
| 18:47.53 | jkroon | already have the stuff in the kernel compiled in and the kernel modules loaded. |
| 18:48.12 | WIMPy | autofsckk: Yes, yes, and see the manual again. Or use feature transfers. |
| 18:48.31 | WIMPy | jkroon: Yes |
| 18:48.46 | mmcji | I am using a sangoma wanpipe card with asterisk. Will I see RX/TX activity through the T1 connected to the card when the hdlc frame is sent to the remote switch? also is this frame sent by asterisk? |
| 18:48.57 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 18:48.57 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 18:49.09 | autofsckk | cool, thanks a lot WIMPy |
| 18:59.29 | *** join/#asterisk aross42 (~aross@CPE009400809a9c-CMb89bc9d2e1a5.cpe.net.cable.rogers.com) |
| 19:09.27 | *** join/#asterisk rhce7320 (~rhce7320@59.167.200.141) |
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| 19:11.03 | *** join/#asterisk anonymouz666 (~anonymouz@189-25-32-59.user.veloxzone.com.br) |
| 19:13.02 | ageis | where are voicemail wavs stored? |
| 19:14.35 | anonymouz666 | I really do not understand why using tshark in the outgoing interface shows packet loss. What's the meaning of that? If I want to measure for sure the quality of point A (src) I need to go to point B (dst). From A (src) point of view, should be no packet loss on its interface. |
| 19:15.24 | WIMPy | Unless it can't cope. |
| 19:16.39 | *** join/#asterisk Luke-Jr (~luke-jr@2001:470:5:265:222:4dff:fe50:4c49) |
| 19:18.07 | jkroon | WIMPy, any ideas on getting support for fixing that BRI issue of mine? chan_misdn is dead, mISDNuser v2 is still OK to install, but chan_misdn can't use it, so now you need LCR + chan_lcr ... which is just more crap that I really don't want |
| 19:18.34 | anonymouz666 | 192.168.253.20 25832 192.168.253.176 7000 0x0F8924F7 ITU-T G.711 PCMA 9752 51 (0.5%) 60.00 1.38 0.41 X |
| 19:19.16 | anonymouz666 | 0.5 of packet loss from 192.168.253.20 to 192.168.253.176 but this dump was collected on eth0 that is 192.168.253.20! |
| 19:19.24 | WIMPy | jkroon: I know that feeling. But after I finally did it anyway it worked very well for me. |
| 19:20.00 | WIMPy | Unfortunatly I have no idea, where the issue you see comes from. |
| 19:22.02 | WIMPy | Oh, and for some reason, chan_misdn does not to seem dead at all. |
| 19:22.24 | jkroon | yea, bunch of people still using v1.1 ... even without the download archives. |
| 19:22.42 | jkroon | ok, so you suggest just pushing through and going for it via chan_lcr? |
| 19:23.09 | jkroon | is feeling very badly dropped with dahdi |
| 19:23.12 | WIMPy | You can use irroots or Digiums version instead. |
| 19:23.49 | jkroon | ... willing to give that a shot too |
| 19:23.55 | jkroon | where can I find those? |
| 19:24.00 | WIMPy | And for the dahdi part you could off course contact digium support and see if they've got any idea. |
| 19:24.12 | WIMPy | http://voice.yeti.dk/Asterisk_vs_ISDN/6 |
| 19:27.12 | jkroon | *sigh* |
| 19:27.29 | jkroon | will contact digium, they need to support their hardware. |
| 19:27.49 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 19:28.11 | Qwell | What hardware do you have? If it's Digium hardware, why are you not using DAHDI? |
| 19:29.17 | jkroon | Qwell, because dahdi is dropping calls. |
| 19:29.25 | sruffell | jkroon: yeah....why not just use DAHDI and a newer version of Asterisk? The only people who haven't been able to update from mISDN I know of are using older versions of Asterisk. |
| 19:29.32 | jkroon | It's a B410P card, unfortunately I don't have the serial number. |
| 19:30.14 | Qwell | Switching to an unsupported software stack to fix dropping calls, rather than try to debug the problem seems a bit much. |
| 19:30.15 | jkroon | sruffell, Qwell - well, now you know of someone who just lost a day trying to figure out why the same card worked with with misdn but is dropping calls with dahdi 2.6.1 (bri_cpe_ptmp mode). |
| 19:30.31 | sruffell | it's probably not DAHDI..... |
| 19:30.37 | jkroon | Qwell, agreed, would love to fix this correctly. |
| 19:30.38 | WIMPy | For older Asterisk versions tehre's even an chn_misdn2. Unfortunately unreleased. |
| 19:30.42 | sruffell | did you also update libpri, and Asterisk? |
| 19:31.09 | jkroon | sruffell, then please explain why misdn worked and libpri 1.4.12, ast 1.8.15.1 + dahdi-{linux,tools}-2.6.1 doesn't? |
| 19:31.23 | jkroon | and yes, the old install was a 1.4 + some misdn version install. |
| 19:31.28 | sruffell | because of ptmp support |
| 19:31.29 | jkroon | brand new install. |
| 19:31.44 | sruffell | this is when I wish rmudgett was in this room.... |
| 19:31.55 | jkroon | ok, so can ptmp support be sorted out? works fine against an NT1, but fails against NT2. |
| 19:32.17 | jkroon | client is refusing to have the telco replace the NT2 with an NT1 ... even though I know from experience that fixes the issue. |
| 19:32.23 | sruffell | refreshes his knowledge of history. |
| 19:32.24 | WIMPy | That doesn't make HDLC errors. |
| 19:32.28 | sruffell | history of libpri/chan_dahdi |
| 19:32.35 | sruffell | oh...you're getting HDLC errors? |
| 19:32.38 | jkroon | yes. |
| 19:32.42 | jkroon | HDLC Abort |
| 19:32.48 | sruffell | hmm...telco power savings? |
| 19:33.00 | jkroon | possibly, but it happens during calls. |
| 19:33.09 | sruffell | i.e...is layer one being ...during calls.../me thinks |
| 19:33.12 | jkroon | telco power saving wouldn't do that during a call. |
| 19:33.14 | WIMPy | That shouldn't give HDLC aborts, either. |
| 19:33.40 | jkroon | sruffell, i'll email you a beer if you can get this sorted for me using dahdi ... |
| 19:33.43 | WIMPy | Actually I have seen that, but that's a different story. |
| 19:33.59 | sruffell | thinks |
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| 19:34.09 | jkroon | is not particularly keen on trying to make misdn work ... again |
| 19:34.13 | mjordan | sruffell: rmudgett says HDLC errors are coming from DAHDI |
| 19:34.36 | sruffell | mjordan: yeah, thanks...I didn't catch that part. |
| 19:34.40 | jkroon | mjordan, does he have suggested "fixes"? |
| 19:34.50 | kuku | what can I do to find out all the available channels for a given extension ? |
| 19:35.49 | mjordan | jkroon: what is the timing source in Asterisk? What position are the terminating resistors in on the card? |
| 19:37.23 | jkroon | mjordan, i'm 95% sure the terminating resistors are correct (can have that double checked in a few hours), dahdi is the slave (http://pastebin.com/bvYEAWEy) and span1 is the master channel. |
| 19:37.44 | jkroon | span 1 + 3 is working correctly (NT1 units), span 2 is connected to the NT2 and is causing problems. |
| 19:38.12 | WIMPy | Pretty interesting that you have different hardware there. |
| 19:38.22 | jkroon | WIMPy, don't get me started :( |
| 19:38.37 | jkroon | if I had my way it would all be NT1s |
| 19:39.29 | jkroon | mjordan, and users.conf: http://pastebin.com/A6CfC4V0 (which I prefer to use for dahdi due to the inheritance of groups) |
| 19:39.31 | WIMPy | I mean I've never seen explicit NT1 or NT2 hardware. |
| 19:39.47 | jkroon | Telkom refers to them as ISDN2 or ISDN2a |
| 19:39.58 | mjordan | jkroon: NT1/NT2 isn't a term I'm familiar with |
| 19:40.09 | jkroon | oh, then I might have my terminology wrong. |
| 19:40.24 | WIMPy | No, the terminology is correct. |
| 19:40.24 | mjordan | jkroon: I wouldn't say that :-P |
| 19:40.35 | jkroon | ISDN2 is the one without analog ports on the device, ISDN2a is the one with. |
| 19:40.38 | mjordan | jkroon: firmly willing to admit ignorance on that reference |
| 19:40.40 | sruffell | jkroon: I'm not bailing on you..but I have to run. I'll check back later. |
| 19:40.44 | jkroon | mjordan, ah ok :) |
| 19:41.03 | jkroon | sruffell, ok, i'll hang around, have other work that I should have done today which I'll try and sort quickly. |
| 19:41.34 | WIMPy | Ah, so it's about integrated terminal adaptors. |
| 19:43.05 | WIMPy | jkroon: The NTs should be intercahngable. |
| 19:43.15 | jkroon | my turn to get a blank look on his face |
| 19:43.23 | jkroon | WIMPy, indeed they should. |
| 19:43.40 | mjordan | jkroon: you get a +1 from rmudgett for using the newer config format |
| 19:43.52 | jkroon | my experience however states that get Telkom to get rid of the ISDN2a (NT2) and replace with ISDN2 (NT1) |
| 19:44.00 | newtonr | yeah the B410P is advertised to act as the S or T interfaces, so it can be the NT1 or NT2 effectively from what i understand |
| 19:44.06 | jkroon | mjordan, thanks :). i think that's a first. |
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| 19:44.10 | newtonr | s/act/provide |
| 19:45.05 | WIMPy | The config says ptmp for all ports, so technically they should all be NT2. |
| 19:45.29 | WIMPy | But I doubt it really refers to the technical terms. It's probably just a product name. |
| 19:46.34 | jkroon | could be that both are NT2's, but something is different between Telkom's ISDN2 and 2a, DAHDI likes ISDN2 and hates ISDN2a. I would LOVE to figure out what and get this sorted once and for all. |
| 19:47.09 | *** join/#asterisk sustav (~vpp@nat/digium/x-kshyaqoydhpogfoh) |
| 19:47.21 | jkroon | got a 25MB asterisk full log (pri debug on all channels, verbose 4 for dialplan) I'm more than willing to provide to digium. |
| 19:47.28 | mjordan | jkroon: just to confirm, its NT2 that has the analog lines? |
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| 19:47.47 | newtonr | jkroon: all you have to do is contact Digium support, they will support it with DAHDI.. |
| 19:47.52 | jkroon | mjordan, ISDN2a has the analog lines yes. They are not plugged in. |
| 19:48.07 | mjordan | that doesn't stop the device from grabbing TEIs |
| 19:48.22 | jkroon | newtonr, but I need the serial number which I don't have atm, already sent an email to support@digium.com no response. |
| 19:48.47 | jkroon | mjordan, indeed you're right come to think of it - do you have a suspicion? |
| 19:49.31 | mjordan | jkroon: rmudgett says he's had problems with that exact device before (prior to that version of Asterisk/libpri), but can't remember what it was |
| 19:49.45 | WIMPy | Off course it will, but that doesn;t seem interesting to me. |
| 19:49.46 | mjordan | jkroon: rmudgett says its possible that the device might be causing the CRC errors |
| 19:50.04 | jkroon | mjordan, thanks for the feedback. can I switch off CRC checking on DAHDIs side as a test? |
| 19:50.37 | mjordan | jkroon: probably better to ask sruffell for that :-P |
| 19:50.45 | mjordan | sorry, but I won't claim to be a DAHDI expert |
| 19:50.51 | jkroon | mjordan, also, would it help if I quickly log a ticket with digium support - even if using a serial number for another of my B410P cards? |
| 19:51.00 | mjordan | newtonr is checking |
| 19:51.02 | jkroon | mjordan, we can't be experts in all things :) |
| 19:51.10 | newtonr | jkroon: yeah im checking right now |
| 19:51.34 | jkroon | thinks digium's karma just gained a few points in his books. |
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| 19:52.45 | mjordan | jkroon: the acting suspicion (and its only a hunch from rmudgett) is that the NT2 box may be stepping on bits it shouldn't be, which is causing the CRC errors |
| 19:53.38 | jkroon | would not be surprised at all ... busy scanning mod-info for ways of switching off the CRC stuff as well. |
| 19:54.12 | WIMPy | Isn't that in system.conf? |
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| 19:55.10 | jkroon | WIMPy, for PRI I know it is, but with the PRI if your peer has it on, you need it on too otherwise dahdi won't send crc checksums causing HDLC errors as well. |
| 19:55.28 | jkroon | what we want is to simply *ignore* the CRCs from the peer |
| 19:55.41 | WIMPy | jeah |
| 19:56.12 | WIMPy | But do you have CRC errors as well? |
| 19:57.12 | jkroon | hmm, no just HDLC Abort. |
| 19:57.21 | kuku | what can I do to find out all the available channels for a given extension ? |
| 19:57.58 | jkroon | kuku, what exactly are you looking for? the question doesn't make much sense for me. |
| 19:57.58 | WIMPy | kuku: That question doesn't make sense. What are you looking for? |
| 19:58.09 | jkroon | this is the error I'm seeing in *: PRI got event: HDLC Abort (6) on D-channel of span 2 |
| 19:58.16 | citywok | mjordan: i set up the debugging and it hasn't crashed since i turned it on. argh. |
| 19:58.30 | jkroon | dmesg isn't reporting anything (at least not at default log levels) |
| 19:58.54 | citywok | jkroon: last time i had similar HDLC errors and the T1 kept dropping i had a bad card, ended up replacing it. |
| 19:59.28 | jkroon | citywok, swapped spans 1 and 2 around, HDLC errors moved from span 1 to 2. |
| 19:59.43 | citywok | jkroon: easy way to prove that one! lol |
| 19:59.52 | citywok | have you called your carrier yet? |
| 20:00.10 | mjordan | jkroon: just to double check - are established calls getting dropped? |
| 20:00.14 | WIMPy | Have you tried swapping the lines before the NTs? |
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| 20:00.51 | jkroon | mjordan, correct. |
| 20:01.08 | jkroon | WIMPy, please define "before" |
| 20:01.10 | mjordan | jkroon: rmudgett's response: That's certainly not good. |
| 20:01.23 | jkroon | mjordan, :) |
| 20:01.24 | WIMPy | Swapped the telco lines. |
| 20:01.41 | jkroon | WIMPy, that's actually not a bad idea ... wish I had physical access right now. |
| 20:01.44 | mjordan | jkroon: what's confusing there is that the ISDN session is established and media only should be sent. |
| 20:01.53 | jkroon | jip. |
| 20:01.53 | mjordan | unless the switches are restarting the link |
| 20:02.02 | WIMPy | When you did the install, did you move the box? I.e. has it been re-cabled? |
| 20:02.13 | jkroon | also, sporadically I'll get a cause code 27 (destination equipment out of order) when trying to make a call - which is super weird. |
| 20:02.20 | jkroon | WIMPy, yesterday. |
| 20:02.55 | WIMPy | I still think it might be the cabling/termination. |
| 20:03.04 | jkroon | re-used the exact same cabling that came from the Telco to the NTUs, as well as the same cables from the NTUs to the B410P, and used the same B410P card |
| 20:03.26 | jkroon | WIMPy, can you motivate that somehow? |
| 20:03.43 | WIMPy | Hmm. That actually indicates a bad line. You usually get OOO when the fault is between the CO and te NT. |
| 20:04.12 | WIMPy | the |
| 20:04.22 | mjordan | jkroon: was that call coming towards the asterisk box when you got the 27? Or outbound from asterisk? |
| 20:04.33 | jkroon | that's interesting and makes sense, but why did it work with mISDN? |
| 20:04.41 | jkroon | mjordan, that particular one was outbound. |
| 20:05.00 | WIMPy | Oh, in that case forget what I said. |
| 20:05.24 | mjordan | which wouldn't be us sending the 27 then. The switch could be really hating that line at that point however. |
| 20:05.31 | WIMPy | You could also try to use the analog ports and see if they work ok. |
| 20:05.55 | jkroon | mjordan, jap, the debug shows the 27 coming from the line. |
| 20:06.04 | mjordan | jkroon: have to bounce to a meeting. newtonr is looking into the support thing. I'll check back in a bit. |
| 20:06.12 | jkroon | WIMPy, yes, I can get the client to do that. thanks mjordan |
| 20:06.42 | newtonr | jkroon: they should be calling you here in a minute |
| 20:07.00 | *** join/#asterisk vlad_starkov (~vlad_star@77.41.88.210) |
| 20:07.14 | jkroon | thanks newtonr |
| 20:07.24 | jkroon | got root access ready ... |
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| 21:09.41 | ska | Is there a way for asterisk to store voicemail on a remote server? |
| 21:10.06 | Qwell | several |
| 21:10.21 | Qwell | nfs mounted FS, odbc, imap |
| 21:12.57 | jkroon | odbc works quite well. |
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| 21:16.08 | kuku | what can I do to find out all the available active channels for a given extension ? |
| 21:16.41 | WIMPy | kuku: We still have no idea what that means. |
| 21:21.15 | citywok | jkroon: silly question but did you swap the cables on the PBX side or on the carrier side? |
| 21:21.39 | citywok | cables going bad isn't completely unpossible, although quite rare |
| 21:21.48 | Qwell | citywok: he says he replaced them, IIRC |
| 21:22.01 | citywok | Qwell: Ah, i scrolled up and was reading but must have missed that. |
| 21:22.10 | jkroon | Qwell, no, same cables that still worked the day before. |
| 21:22.23 | jkroon | swapped them between spans 1 and 2 right at the card. |
| 21:22.24 | Qwell | guess not |
| 21:22.37 | citywok | jkroon: swap them on the other side and see if it still follows |
| 21:22.45 | jkroon | so here is the kicker: reboot. can't reproduce the HDLC errors reliably any more. |
| 21:22.51 | jkroon | citywok, will have to do that tomorrow. |
| 21:23.08 | jkroon | i'm just afraid the HDLC errors is going to be coming back in the morning .... |
| 21:23.57 | jkroon | just still stumped that misdn worked and dahdi not. utterly and totally stumped. |
| 21:24.15 | jkroon | will get a technician out there tomorrow if need be ... |
| 21:25.15 | citywok | i fought with a carrier in the philippines, total outage was 27 days on one of 4 bonded trunks before i finally convinced them to go out and figure out wtf was wrong. |
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| 21:25.48 | citywok | it would just bounce and error intermittently and every time they scheduled a truck roll, they would test right before for 5 minutes and it would be clean for 5 minutes and they would cancel. arghhhh. |
| 21:25.50 | jkroon | damn |
| 21:25.55 | citywok | they asked me to swap cables at least 5 times |
| 21:26.17 | jkroon | that sucks |
| 21:26.34 | citywok | being 5,000 miles away makes things difficult. |
| 21:27.10 | ska | Qwell: I'm planning to run on an appliance that has no storage, but boots from compact flash. I might be able to put on an MTA. |
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| 21:44.14 | Dovid | anyone here use mplayer with Asterisk? |
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| 21:57.38 | libryder | i have unimrcp and cepstral setup, and i compiled unimrcp using the —with-swift flag. when i start the unimrcpserver, i get: |
| 21:57.41 | libryder | 2012-09-27 21:57:31:266766 [WARN] Failed to Load DSO: ../plugin/mrcpcepstral.so: cannot open shared object file: No such file or directory |
| 21:58.25 | libryder | i did find / -name mrcpcepstral.so with no results |
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| 22:06.07 | navaismo | libryder, wrong channel? |
| 22:06.50 | libryder | i'm not sure where to go for cepstral+asterisk help |
| 22:09.13 | navaismo | try in the maillist/forum of unimrcp, clearly it doesn't create you module and check the real path to the swift app |
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| 22:12.04 | libryder | okey, thanks |
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| 22:23.46 | jkroon | *sigh* HDLC is back :( |
| 22:25.51 | sruffell | it left? |
| 22:26.44 | sruffell | So I take it you're still having problems. Anything new clues before I go back and read everything from before? |
| 22:26.46 | jkroon | yes, for a few hours after a reboot |
| 22:27.12 | jkroon | pvt? got a case number for you |
| 22:27.16 | sruffell | You ran misdn on this same system before? Or completely new install with the same B410P? |
| 22:27.23 | sruffell | kk |
| 22:27.28 | jkroon | completely new install |
| 22:28.02 | jkroon | ok, case number should get you access to loads more info, including full logs. |
| 22:29.08 | *** join/#asterisk Lantizia (~lantizia@cpc6-stok15-2-0-cust8.1-4.cable.virginmedia.com) |
| 22:30.07 | Lantizia | hey does asterisk now include support for broadsoft sip access? (e.g. tell handsets that support it to answer or hold calls from asterisk) |
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| 22:30.50 | Lantizia | all I've found is this... https://issues.asterisk.org/jira/browse/ASTERISK-11159 |
| 22:31.00 | jkroon | Lantizia, i've been linking broadsoft to asterisk for four years now ... what's the problem? |
| 22:31.44 | jkroon | nm, SLA stuff. |
| 22:31.49 | Lantizia | well we've a customer who wants to use some middleware with the asterisk AMI - they want to sent sip notify events to aastra peers to answer calls and hold them |
| 22:31.49 | jkroon | not familiar with that. |
| 22:32.15 | Lantizia | nah neither am I really |
| 22:34.02 | jkroon | ok, but there is a sip notify framework in asterisk - won't that suffice for your needs? |
| 22:35.58 | jkroon | sruffell, any ideas? debugging to enable? |
| 22:36.31 | sruffell | yeah…only two places in the driver that will generate those since they are coming out of the interface chip directly (hardhdlc) looking for the debug flags now... |
| 22:37.59 | Lantizia | jkroon, I don't know - when I key in 'sip notify' and tab - all I get for autocomplete is things to reboot particular makes of phones |
| 22:38.05 | mjordan | Lantizia: IIRC, the Broadsoft SLA stuff cannot be licensed under GPLv2. |
| 22:38.14 | Lantizia | i'd be nice to know if I can program my own notifies |
| 22:38.27 | Lantizia | mjordan, is this really SLA though? it's just the part that sends two sip notifications |
| 22:38.54 | jkroon | debug=64? |
| 22:39.06 | jkroon | sruffell, on wcb4xxp module? |
| 22:39.31 | sruffell | yeah…and spanfilter is already set to 0xff so that should be enough. You can change spanfilter to limit it to just the problem port if too much spam. |
| 22:39.33 | jkroon | Lantizia, you can add your notifies |
| 22:39.48 | Lantizia | jkroon, how? are they text files somewhere? |
| 22:39.50 | jkroon | ok, just echo 64 info live running paramters? |
| 22:40.02 | jkroon | Lantizia, yes, in /etc/asterisk, sip_notify.conf iirc. |
| 22:40.45 | Lantizia | hmm i wonder if it's that simple |
| 22:40.49 | sruffell | Do you have physical access….I've not ever used those debug modes... |
| 22:40.55 | sruffell | just in case you need to reboot |
| 22:40.55 | jkroon | sruffell, echo 64 > /sys/module/wcb4xxp/parameters/debug good enough? |
| 22:41.05 | jkroon | no, not at the moment unfortunately. |
| 22:41.07 | sruffell | I think so….but I can't personally vouch that it's not without risk |
| 22:41.09 | sruffell | hmm... |
| 22:41.38 | jkroon | you should have root credentials in the case ... so if you'd like access to the box ... |
| 22:42.11 | jkroon | it be good, stuff starting to appear in dmesg |
| 22:42.33 | sruffell | heh….brave…I was going to say I should try and get it setup on one of the test machines to see. |
| 22:43.16 | sruffell | So you'll either see "HDLC Abort" or "odd, zlen less than 3?" Then I can go from there... |
| 22:43.22 | jkroon | client is ... furious as it stands anyway. |
| 22:43.36 | sruffell | :/ |
| 22:44.08 | jkroon | less than 3 :) |
| 22:44.22 | jkroon | http://pastebin.com/3UDTgies |
| 22:44.42 | sruffell | goes back to the code. |
| 22:45.23 | jkroon | wishes he understood more of the BRI protocol and the DAHDI drivers in general |
| 22:45.26 | Lantizia | mjordan, according to one post I've read it should be as simple as adding events talk and hold to sip_notify.conf - but apparently they need to be sent in a 'new dialogue' |
| 22:45.47 | Lantizia | the blog i'm reading is spanish translated to english mind you lol |
| 22:46.04 | Lantizia | they apparently need to be sent in the ongoing dialogue |
| 22:47.20 | jkroon | sruffell, would you mind committing a spelling fix on that less then ? should be less than if my english isn't totally packing up on me :) |
| 22:47.32 | jkroon | not urgent, but if code is going in anyway. |
| 22:48.01 | Lantizia | mjordan, where did you read it can't be licensed under GPLv2? I really don't mind that as the answer - it'll mean I don't have to do the work for the client :P |
| 22:48.06 | WIMPy | "at least"? |
| 22:48.08 | sruffell | Heh…kk |
| 22:51.05 | *** join/#asterisk TimeRider (~steve@188-220-34-144.dsl.cnl.uk.net) |
| 22:52.06 | jkroon | sruffell, ok, if I understand the code correctly there always has to be at least three bytes in a frame? what is the implication if there are fewer? should we just wait for more, or is it an actual issue? or is it perhaps safe to just ignore them? |
| 22:53.00 | sruffell | Yeah…reading through the data sheet for the framer now. Probably not safe to just ignore… |
| 22:53.20 | sruffell | tzanger: might have a clue…. |
| 22:53.55 | jkroon | tzafrir is also usually somewhat clued up on dahdi in my experience. |
| 22:54.16 | jkroon | tzafrir_laptop, you around? |
| 22:54.22 | sruffell | tzanger wrote that part of wcb4xxp. |
| 22:54.30 | jkroon | even better ... |
| 22:57.40 | jkroon | do you have a good reference at hand for the framing that you can provide me a link to? google is providing some information but I'm not sure how reliable the details are. |
| 22:58.14 | WIMPy | framing or framer? |
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| 23:00.21 | jkroon | semantics? ie, framer sounds like a worker whereas framing describes what the frames look like? |
| 23:00.33 | sruffell | framer…the zlen has to do with internal buffer management on the http://www.colognechip.com/hfc-8s-4s.pdf(which this conversation is veering away from what is probably appropriate in this room) |
| 23:00.52 | sruffell | bad link insertion there... |
| 23:00.56 | jkroon | however, it seems each individual frame contains 4 bits of the D channel. |
| 23:01.33 | jkroon | sruffell, if we need to take this elsewhere that's fine by me - just let me know where. |
| 23:01.53 | WIMPy | jkroon: framing as in how the data is transmitted in general vs. framer, the part of the chip. |
| 23:02.45 | sruffell | oh it's cool..we can talk about the details in the code in #asterisk-dev if you want….it's probably generally applicable here about how to trouble shoot HDLC aborts… but after I read the sections of the data sheet I can form a new hypothesis to test. |
| 23:04.06 | jkroon | WIMPy, makes sense :). so i'd like to understand the framing to see if what the chip does is correct. but I just realized that the debug output is already decoded, I think. |
| 23:05.14 | jkroon | sruffell, any chance of just using dchan instead of hardhdlc? ie, use the software framer? |
| 23:05.15 | WIMPy | As the chip dies the HDLC framing, it's the idea that you don't (have to) see the raw data. |
| 23:05.36 | jkroon | well, the chip is returning data, the module doesn't know what to do with it. |
| 23:05.39 | *** join/#asterisk Lepon (Lepon@60-241-109-208.static.tpgi.com.au) |
| 23:05.46 | sruffell | jkroon: no…not with the wcb4xxp…it doesn't spit that out (that I know of) on a raw channel. |
| 23:06.00 | *** join/#asterisk gg608f (~Adium@c-67-180-129-182.hsd1.ca.comcast.net) |
| 23:06.46 | jkroon | discussion earlier was that you probably actually want hardhdlc anyway. i'm guessing the same holds for PRI links ... which I realized today I always just used dchan ... will use hardhdlc for those too in future. |
| 23:07.46 | sruffell | I do not think I would recommend that….on some of the board the overhead on the PCI bus to deal with the hardware HDLC controllers outweighs the advantages of doing the decoding/encoding in hardware. |
| 23:07.47 | Lepon | Hey all, I'm using asterisk 1.8 and trying to record the SIP response from failed outgoing calls started by the asterisk management interface. I have put "${HASH(SIP_CAUSE,${CDR(dstchannel)})}" in my failed priority but there is never any entries for SIP_CAUSE in there. I can use it correctly in my dail plans after a failed Dial() call but not in the failed context from AMI generated calls. |
| 23:08.00 | Lepon | Any ideas would be great |
| 23:08.18 | jkroon | sruffell, ok, so just stick with dchan :) |
| 23:08.25 | Lepon | I also have storesipcause=yes in my sip.conf |
| 23:10.19 | jkroon | Lepon, that's new, haven't seen that before. |
| 23:10.52 | jkroon | try running DumpChan() on that channel to see if it's there and you're just missing it |
| 23:11.00 | *** join/#asterisk SeRi (~wtf@c-98-200-53-71.hsd1.tx.comcast.net) |
| 23:11.05 | Lepon | okay ill give that ago thanks |
| 23:11.54 | *** join/#asterisk jpsharp (jsharp@ohno.mrbill.net) |
| 23:12.32 | Lepon | unfortunately doesn't seem to be there |
| 23:12.42 | Lepon | maybe you can't use that in the way I am trying to? |
| 23:13.56 | jkroon | i dunno, just looked at the code, it set's the variable: MASTER_CHANNEL(HASH(SIP_CAUSE,%s)), where %s gets replaced with the owning channel's name. |
| 23:14.35 | jkroon | so perhaps try for ${MASTER_CHANNEL(HASH(SIP_CAUsE,${CDR(dstchannel)}))} ?? |
| 23:15.04 | Lepon | can't hurt to give ago |
| 23:15.21 | sruffell | jkroon: it's odd…that <3 check is explicitly disabled in the wctdm24xxp driver (which can have B400M modules) I think it might PTMP related. I could generate a patch but I don't think I'll be able to test it. |
| 23:15.39 | sruffell | and nothing in the data sheet referenced a minimum size (which is why I looked in the other driver)... |
| 23:15.46 | jkroon | ok, so what do I do, just comment it out? |
| 23:15.56 | sruffell | ok..if you're willing.../me looks |
| 23:15.58 | Lepon | do the dial plans have a channel when there is no call in progress? maybe because it couldn't intiate the call it doesn't have a channel or something weird? (don't really know a great deal about how it all works yet unfortunately) |
| 23:16.23 | jkroon | is willing to risk it .... wonder whether I might not perhaps havea B410P card lying in the office he can quickly put into a machine ... |
| 23:16.46 | jkroon | will have to drive there but it's at least accessible ... |
| 23:16.48 | *** join/#asterisk classix (~salven@silenceisdefeat.com) |
| 23:17.39 | sruffell | I don't think it'll just be a comment out…I'll have a patch for you here with then next 20 minutes. |
| 23:17.53 | sruffell | argh….English…whatever... |
| 23:18.18 | Lepon | no such luck |
| 23:19.13 | jkroon | if (0 && zlen < 3) { <-- from the xhfc from wctdm24xxp driver ? |
| 23:19.44 | WIMPy | o.O |
| 23:20.29 | jkroon | yes, that looks odd, it basically just disables the zlen < 3 check, el cheapo. |
| 23:20.44 | kuku | wimpy: |
| 23:20.50 | kuku | Are you there wimpy ? |
| 23:20.50 | sruffell | Yeah….I just want to make sure the reset of the function works the way it would. So I don't think you can just comment out the check. |
| 23:21.46 | WIMPy | Yes, you could say it IS commented out. |
| 23:21.50 | jkroon | it does, those two code blocks are functionaly identical |
| 23:22.30 | jkroon | both just end up calling dahdi_hdlc_finish() if status byte is zero, if status byte is non-zero it does hdlc abort, everything else is debug code (printks) |
| 23:22.46 | kuku | I'm trying run chanspy on an extension - but sometimes it gets attached to the wrong channel. So Instread of syaing I wasn to spy on SIP/15 I need to specify SIP/15-xxxxxx ( the specific channel - so asterisk doesn't try to "guess" at which channel it should be ). The reason is that about 60% of the time it chooses the wrong channel for the extension. |
| 23:22.59 | sruffell | but there is that hdlc_signal_complete in the wctdm24xxp driver... |
| 23:23.18 | jkroon | which is exactly the code in the else portion from wcb4xxp :) |
| 23:24.05 | WIMPy | kuku: Have you tried to add the "-" to the channel name? |
| 23:24.12 | sruffell | ahh…yeah…ok…I see that. |
| 23:24.26 | sruffell | so…yeah..you don't even need a patch from me…just comment out the abort and give it a try. :) |
| 23:25.08 | kuku | WIMPy: I don't know the channel name. I'm trying to query for it while the call is in progress. |
| 23:25.34 | kuku | That is the question I was posting previously ( I'm sorry I wasn't around to reply to your question then ) |
| 23:25.58 | WIMPy | kuku: I mean, just change the "SIP/15" to "SIP/15-". |
| 23:26.43 | jkroon | this'll just be my luck ... as soon as we've got a probably fix I can't log in to the machine with the problem any more :( |
| 23:27.01 | kuku | Now I'm confused - specifying SIP/15- will fix it ? but thats not a complete channel name... |
| 23:27.10 | WIMPy | Murphy |
| 23:27.39 | sruffell | uhh ohh…hopefully it's not hosed because of debug output to the kernel logs... |
| 23:27.56 | WIMPy | kuku: It doesn't have to be complete. |
| 23:28.20 | WIMPy | sruffell: Do yu have any theory why it could happen? |
| 23:28.20 | jkroon | sruffell, well, at least I have *something* now, so a reboot will get back to a clean state. |
| 23:28.20 | kuku | But why adding the "-" will help ? |
| 23:28.57 | jkroon | because otherwise a bunch of other channels can match too, eg SIP/1555-123 |
| 23:29.20 | WIMPy | kuku: I don't know what your names are, but if you have apeer called 150 as well, for exaple, that would also match if you don;t add the "-". |
| 23:29.29 | kuku | aaaaaaaaaaaaaaaaaaaaaaaaaaaaa |
| 23:29.35 | kuku | is a dumbass |
| 23:29.46 | kuku | Thank you thank you thank you |
| 23:29.57 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-hsbcwvbzyxaylgmd) |
| 23:30.29 | WIMPy | At leat that was a easy one. |
| 23:30.48 | sruffell | jkroon: you doing it now? I'll wait to hear any results if so. |
| 23:31.26 | jkroon | sruffell, can't gain access to the client premises to go reset that machine ... so I'm stuck until they reboot it. |
| 23:31.40 | jkroon | could just as well be their internet or power as well that decided to kick |
| 23:31.50 | sruffell | ahh…:/ I thought when you said *something* you were able to get a login. |
| 23:32.01 | sruffell | ok…then I'll watch the ticket. |
| 23:32.07 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 23:32.44 | sruffell | I can check with creslin tomorrow about the need (or lack of) the check for 3 bytes in the buffer. He's knowledgeable about those framers too. |
| 23:33.10 | WIMPy | sruffell: Do you have any theory why it could happen? |
| 23:33.27 | WIMPy | I mean it doesn't seem to be a common issue. |
| 23:33.36 | WIMPy | Just wondering if there might be any way to reproduce it. |
| 23:33.43 | sruffell | yes…working hypothesis is that on PTMP links there might be other runt messages….the Hx8 was tested more heavily against PTMP links than the B410P was. |
| 23:33.45 | jkroon | sruffell, no, I meant I had something to try :). |
| 23:34.39 | WIMPy | Hmm. Doesn't make sense to me. |
| 23:34.44 | jkroon | bows to the master of the ISDN, thank you very much sruffell - assistance is HIGHLY appreciated. |
| 23:34.56 | sruffell | WIMPy probably knows more about ISDN than I do…. |
| 23:34.58 | jkroon | WIMPy, doesn't have to make sense - just has to work :) |
| 23:35.31 | jkroon | WIMPy, essentially wcb4xxp freaks out and aborts HDLC if it receives D-chan frames with zlen<3, even if the status byte is 0. |
| 23:35.36 | WIMPy | But I didn't dig in to the hardware. |
| 23:36.00 | jkroon | wctdm24xxp does not. |
| 23:36.03 | sruffell | I'm thinking it's commented out in the driver for the Hx8 cards because someone hit this while doing integration testing. |
| 23:36.35 | kuku | Ok guys - the "-" didn't work |
| 23:36.37 | sruffell | which is why I can probably poll some more people and get an answer before trying to decode the Cologne chip data sheet…which isn't what I would call a straightforward document. |
| 23:36.46 | WIMPy | I have never seen these messages and when I tried dahdi BRi I only did so on PTMP so far. And it doesn't happen on your other lines, either. |
| 23:37.18 | sruffell | Yeah…so what is the difference between N1 and N2 lines where you are? |
| 23:37.24 | WIMPy | So even it it may be correct, there must be something unusual. |
| 23:37.49 | kuku | Wimpy: chanspy is sitll attaching to the wrong channel |
| 23:38.19 | jkroon | WIMPy, NT1 vs NT2 ? who knows ... perhaps the NT2 is just buggy ... |
| 23:38.29 | jkroon | kuku, to what channel is it attaching? |
| 23:38.29 | WIMPy | kuku: Can you show us how you called it and what it gave you? |
| 23:38.53 | kuku | Let me do a write up on pastebin and then I'll paste the url. |
| 23:39.01 | jkroon | WIMPy, thanks to you as well for continuously asking questions regarding the setup. you managed to keep me on my toes at least. |
| 23:39.15 | jkroon | kuku, just c&p the cli output ... |
| 23:39.53 | WIMPy | jkroon: You can only learn from unusual occasions. |
| 23:40.08 | kuku | ok |
| 23:40.17 | kuku | grepping - lots of calls on the system now. |
| 23:40.43 | jkroon | WIMPy, that is only partly true, but unusual situations is indeed a good teacher. |
| 23:42.46 | jkroon | ok, i'm going to grab a few hours of sleep before friday strikes. |
| 23:43.12 | WIMPy | Too late here :-) |
| 23:46.21 | kuku | when I look at the cli output it says it attaches to right one... |
| 23:49.32 | WIMPy | Ok, so this could become aninterestig one as well. |
| 23:49.55 | *** join/#asterisk hackeron (~hackeron@gentoo/user/hackeron) |
| 23:50.10 | WIMPy | Unfortunatly I don't really have any ideas on this one. |
| 23:53.29 | kuku | hm |
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