| 00:07.28 | sawgood | check this out ... since I got g729 to work on a SIP LAN to SIP LAN phone ... I then tried to call from the same SIP LAN phone to my cell phone (it failed) ... I never heard ringing, and the call disconnected in 2 seconds ... |
| 00:07.42 | sawgood | When I changed back to ulaw, the call was completed (ringing and talking) just fine! |
| 00:09.04 | sawgood | Its 'easy' to run sip show channels during a live call, but how would one do this for a dropped call which is no longer live (thus sip show channels does not work)? |
| 00:13.54 | ChannelZ | does your ITSP even support g729? |
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| 00:23.23 | sawgood | does the 'order' in which the allow statements are listed in sip.conf determine which codec would 'chosen' first? |
| 00:26.48 | sawgood | This is so cool ... I think I have G.729 in pass-through mode working all the way through to the PSTN (via an ITSP which supports G.729) |
| 00:28.17 | sawgood | <PROTECTED> |
| 00:30.35 | WIMPy | Yes, order matters, but the order in the phone maters more. |
| 00:30.44 | WIMPy | At least that's what I observed. |
| 00:32.33 | sawgood | so, if I have allow=all as the 1st statement in the [general] section of sip.conf (I wouldn't need to have any other entires) .. right? |
| 00:33.20 | sawgood | I just want everything to be used ... but use g729 if possible ... as the 1st choice |
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| 00:44.35 | carrar | allow = <codec> : Allow codecs in order of preference |
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| 00:44.54 | sawgood | I have allow = all (as the 1st entry now) |
| 00:47.06 | sawgood | Unable to find a codec translation path from 0x3f0004 (ulaw|jpeg|png|h261|h263|h263p|h264) to 0x100 (g729) |
| 00:47.28 | sawgood | I keep getting this WARNING in Asterisk with allow = all as the 1st entry in sip.conf |
| 00:48.14 | sawgood | Could this be from the SIP LAN phone trying to pass something other than g729 to Asterisk? |
| 00:48.50 | sawgood | If I put allow = ulaw (first) ... I do not think g729 will pass, but I can try |
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| 00:57.01 | sawgood | I'm show this codec format going out to the ITSP = 0x00 (g729) |
| 00:57.25 | sawgood | I see this codec coming IN from the ITSP = 0x3f0100 (g729) |
| 00:57.54 | sawgood | <PROTECTED> |
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| 02:16.55 | CaneToad | I've got a situation where a remote ATA is registering with my sever, and can successfully make calls to local extensions, but calls out from the server via a provider don't work...it makes the calls, but no audio both ways. Where should I be looking? |
| 02:18.28 | CaneToad | calls out from the local extensions work fine, just not calls from remote extensions |
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| 02:19.53 | ChannelZ | Same old sad song. Firewalls, NAT... |
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| 02:51.49 | VenomX | Hi. I wonder if it is possible to send/receive SMS using chan_mobile + Nextel/iDen ... I wonder if it is a matter of phone/asterisk setup or if it is not possible at all |
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| 03:51.53 | eyeballer | hey guys, i have asterisk up and running on my rt16 router with gv set up and linked to my pap2.. outgoing calls work perfectly, however with incoming the phone rings but then i see a seg fault in CLI |
| 03:52.03 | eyeballer | any thoughts on where to look? |
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| 05:42.39 | irroot | mornin |
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| 06:19.25 | schmidts | good morning |
| 06:21.44 | zkn | morning |
| 06:25.05 | zkn | is there a way to specify the context where "alternateexts" of users.conf will place these extensions in the dialplan ? |
| 06:25.47 | WiretapSeven | I think you misunderstand dialplans |
| 06:26.08 | zkn | well i can use include |
| 06:26.39 | zkn | but i suppose i want alternateexts to be created in some other context that default from the get go |
| 06:26.50 | zkn | than* |
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| 06:51.36 | zkn | i'm usin "alternateexts" in users.conf file, it will create all the necessary entries to [default] context and they work as long as that particular user is online.. suppose alternateexts=cellphonenumber, now what I want to achieve is that when that cellphone number is dialed and the peer whom that alternateext belongs to is not online, patternmatching will take over and dials the same initial extension (cellphone number) out to trunk.. |
| 06:52.18 | zkn | but that does not work currently because Goto application changes the extension to peername so patternamtching will not be able to match against the alternateext anymore |
| 06:54.57 | zkn | any thoughts/tips how i could overcome this? |
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| 07:02.25 | jacc0 | Hi all |
| 07:04.01 | jacc0 | I'm looking for a good sip client for my iphone |
| 07:04.18 | jacc0 | Any sugestions? |
| 07:05.28 | WIMPy | zkn: It sould be possible with regcontext, but I haven;t tried that. |
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| 07:10.20 | zkn | ok, gonna try it out, that answers my first question, and I think I have a solution to my second question too... need to test |
| 07:10.27 | zkn | thanks |
| 07:11.03 | kaldemar | ~users.conf |
| 07:11.03 | infobot | [~users.conf] users.conf is a flaming pile of sh1t that takes the fine control of several perfectly usable asterisk config files and reduces them to the lowest common denominator and makes your system behave like a "toaster grade" PBX system. |
| 07:11.22 | zkn | :D |
| 07:12.36 | zkn | i love the alternateexts feature it provides |
| 07:12.48 | zkn | any way i can have that in sip.conf ? |
| 07:13.17 | kaldemar | you can have it in extensions.conf. |
| 07:13.22 | zkn | sigh |
| 07:17.55 | zkn | the beauty of alternateexts is that they get created dynamically, when the user is removed also these extensions will be removed automatically ... easier to manage |
| 07:19.12 | zkn | can you give me an exampe where users.conf renders my Asterisk to a toaster ? :) |
| 07:20.14 | WIMPy | What is alternateext? |
| 07:20.45 | WIMPy | So it's the same as regcontext? |
| 07:20.59 | zkn | right now I don't see it... i can bring over everything from sip.conf that i have defined for peers, pluss gain a few extra features that otherwise requires me to manually write hours of long and dull dialplan |
| 07:22.39 | zkn | WIMPy, alternateexts will create several extensions for the peer that will all ring the device |
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| 07:45.25 | cneb3000 | hmmm... northing like starting your day at the gym with 20 minutes in a jacuzzi :)# |
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| 07:46.53 | irroot | skip the gym and go for 1hr jacuzzi with a couple hotties and bubbly :P |
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| 08:09.38 | cneb3000 | irroot: in my dreams ¬_¬ |
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| 08:30.01 | resmo | hi |
| 08:30.44 | *** join/#asterisk whtsup (hello@WimaxUser3773-134.wateen.net) |
| 08:30.49 | whtsup | hello |
| 08:32.36 | whtsup | i am using 2 asterisk boxes using iax trunk mode the delay is about 800ms b.w them wil jitter buffer will make call quality better |
| 08:32.38 | whtsup | ? |
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| 08:34.24 | WIMPy | If you have jitter, yes. |
| 08:34.34 | WIMPy | But it will make the delay worse. |
| 08:35.10 | WIMPy | Which shouldn't matter if it way above the acceptable level. |
| 08:35.22 | WIMPy | s/it/it is/ |
| 08:41.36 | kaldemar | whtsup: what is wrong with the call quality? |
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| 08:51.47 | schmidts | whtsup 800ms sounds more like a bandwith problem. which connection do you use between these hosts? satelites? |
| 08:52.31 | WIMPy | Erm, where is the connection between delay and bandwidth? |
| 08:52.46 | WIMPy | Unless the delay increases forever, that is. |
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| 09:09.57 | E-bola | Do anybody have time to look at my problem here: http://pastebin.com/djNCfLLh |
| 09:10.22 | E-bola | Its should be pretty simple, i hjave an issue with a gotoif line that works in 1.8.3 but not 1.6.2 |
| 09:13.12 | kaldemar | E-bola: you're getting the warning because ${DB(data/KIC-nightkey)} is empty and there is no "" around it. |
| 09:13.41 | WIMPy | I find the [:200] more interesting. |
| 09:13.47 | kaldemar | GotoIf($["${DB(data/KIC-nightkey)}" = "100"]?KIC,33123030,3:label) |
| 09:14.07 | WIMPy | But yes, if that gives an empty result you've got a problem. |
| 09:15.02 | E-bola | doh |
| 09:15.07 | E-bola | that makes sense :( |
| 09:15.28 | kaldemar | GotoIf(condition?[labeliftrue][:labeliffalse]) <-- that is the syntax, but [] is not literal. it just means that the insides are optional. |
| 09:17.18 | E-bola | It could be a bit more helpfull when encountering empty var's |
| 09:17.32 | E-bola | like give a warning if var is empty |
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| 09:22.35 | kaldemar | "Input: = 100" <-- that should be a pretty good indication. |
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| 10:27.55 | *** join/#asterisk sehh (~sehh@cust-224-67.on1.ontelecoms.gr) |
| 10:27.57 | sehh | hey people |
| 10:28.02 | cneb3000 | hey hey |
| 10:28.31 | sehh | is there some way to send * and # directly to the line, bypassing asterisk? currently, they are interpreted by asterisk. |
| 10:28.41 | sehh | my telco allows me to setup different line settings (forwarding, etc) by sending codes like "#20#" etc |
| 10:29.06 | sehh | but when I try them, the # or * are interpreted by asterisk all the time |
| 10:29.22 | WIMPy | Asterisk won't intercept anything unless you told it to do so. |
| 10:30.27 | sehh | well, I can't say I've "told" it anything in particular, since I don't know much about it anyway. How can I trace the problem and force asterisk to forward the * and # to the phone line? |
| 10:31.01 | WIMPy | Take a look at features.conf and your Dial() options. |
| 10:31.44 | cneb3000 | sehh: ^^^ yea, by default features.conf will interpret # (at least) as you trying to start a blind transfer. |
| 10:32.31 | sehh | hmm |
| 10:32.48 | WIMPy | But only if enabled in the Dial. |
| 10:36.31 | skrusty | morning |
| 10:40.58 | *** join/#asterisk Dovid (~Dovid@office.mypbxmanager.net) |
| 10:41.15 | Dovid | anyone here in Japan ? |
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| 10:59.34 | whtsup | hello |
| 10:59.49 | whtsup | i am using 2 asterisk boxes using iax trunk mode the delay is about 800ms b.w them wil jitter buffer will make call quality better |
| 11:00.19 | kaldemar | sehh: be more specific. are you pressing the keys when in a call or dialing a number with a prefix? what does your dialplan look like? |
| 11:00.20 | mchou | whtsup: no |
| 11:00.53 | whtsup | any solution to use peer with this latency |
| 11:00.54 | whtsup | ? |
| 11:01.03 | mchou | whtsup: 800ms is almost beyond redemption, jitter or no jitter |
| 11:01.32 | WIMPy | whtsup: You could try to put a time machine in between. |
| 11:01.43 | mchou | whtsup: jitter buffer is only going to make the latncy WORSE |
| 11:01.55 | mchou | latency* |
| 11:02.45 | mchou | whtsup: you need to find out what's causing the latency (easier said than done) |
| 11:02.48 | WIMPy | A jitter buffer might increase quiality, but nor usability. |
| 11:03.38 | mchou | whtsup: or just pay the big $$$ and get lower latency carriers |
| 11:04.09 | mchou | whtsup: how far georaphically are the to * servers? |
| 11:04.17 | mchou | s/to/two |
| 11:04.45 | whtsup | one server is in germany |
| 11:04.49 | whtsup | one is west africa |
| 11:05.03 | mchou | well shit |
| 11:05.17 | whtsup | sometime i get delay of 200 ms |
| 11:05.22 | WIMPy | And the connection is via USA? :-( |
| 11:05.31 | whtsup | but its vary alot |
| 11:05.33 | mchou | west afrca carriers are problably responsible for high latency |
| 11:06.11 | mchou | damn. new keyboard is really messing me up! |
| 11:06.37 | mchou | need to get used to these chiclet keys |
| 11:06.46 | WIMPy | mtr should give you some hints. |
| 11:07.54 | mchou | sometimes the internet aint so swell :) |
| 11:08.14 | mchou | might have to resort to old fashioned TDM |
| 11:13.04 | mchou | anyone know how to get the contents of the 's' (start) extension? |
| 11:13.32 | mchou | ${EXTEN} gives me 's' which isn't useful. |
| 11:14.39 | WIMPy | It can't be. |
| 11:14.58 | WIMPy | If a exten was available, you'd end up there and not at s. |
| 11:15.11 | mchou | no |
| 11:15.22 | mchou | this is an inbound |
| 11:15.49 | WIMPy | Every call starts as inbound. |
| 11:16.03 | mchou | it enters the context and thereafter is 's' |
| 11:16.32 | mchou | at least according to ${EXTEN} |
| 11:17.22 | WIMPy | EXTEN is waht it is. If you match n non-wildcard extension, you already know wht's in EXTEN. |
| 11:17.36 | mchou | lol |
| 11:17.51 | WIMPy | And if a call ends up at s, the caller didn't give any details. |
| 11:17.59 | mchou | WIMPy: bullshit |
| 11:18.13 | WIMPy | Bad luck. |
| 11:18.17 | mchou | WIMPy: you have now idea what you're talking about |
| 11:18.30 | mchou | s/now/no/ |
| 11:18.41 | WIMPy | If you know better, why do you ask? |
| 11:19.19 | mchou | WIMPy: because inbout calls always have deatil |
| 11:19.29 | mchou | detail* |
| 11:19.32 | WIMPy | No |
| 11:19.59 | mchou | it's only 's' if the dialplan places it into a context that start with 's' |
| 11:20.08 | WIMPy | If that was true, there wouldn't be an s extension as it could never match. |
| 11:20.15 | mchou | sigh |
| 11:20.32 | kaldemar | mchou: no they don't always have a number. |
| 11:20.34 | mchou | WIMPy: just ignore my question |
| 11:20.49 | mchou | kaldemar: I'm not after a "number" |
| 11:20.53 | WIMPy | Are you asking about the context the call arrives at or about some macro or something? |
| 11:21.14 | kaldemar | mchou: what are you after then? give some details. |
| 11:21.54 | mchou | you have some call arriving inbound from some trunk |
| 11:22.28 | kaldemar | what trunk? |
| 11:22.33 | mchou | in dialplan you say "exten => s, 1, Dial(SIP/myexten)" |
| 11:23.00 | mchou | s is a "generic" match |
| 11:23.16 | kaldemar | no, s is used when there is no other information. |
| 11:23.41 | WIMPy | s is not generic, it is indeed very specific, as kaldemar said. |
| 11:23.41 | mchou | the call always come in with information |
| 11:24.08 | kaldemar | mchou: no it doesn't. what kind of a trunk are you talking about? |
| 11:24.19 | mchou | it's only if the dialplan chooses not to match specifically it falls into 's' |
| 11:24.34 | kaldemar | for example if you have an FXO interface, you won't have anything. hence s will be used. |
| 11:24.41 | mchou | why are we even arguing about this? |
| 11:24.41 | WIMPy | No, s is not a fallback or catchall, either. |
| 11:24.56 | kaldemar | mchou: we're not arguing. why are you not answering questions? |
| 11:25.15 | mchou | I answered the question |
| 11:25.34 | WIMPy | You didn't tell us, HOW the call arrieved. |
| 11:25.36 | mchou | what does it matter what kind of trunk it is? it's a sip trunk |
| 11:25.54 | mchou | it doesn't matter how the call arives is my point |
| 11:26.08 | mchou | the call hasd info, and it's matched by 's' |
| 11:26.27 | WIMPy | No |
| 11:26.30 | kaldemar | mchou: i already told you. if you have FXO, you don't have any information. if you have SIP, the information you're after might be in a header that you have to parse in the dialplan yourself. |
| 11:26.36 | WIMPy | Either of those is not the case. |
| 11:26.48 | mchou | now I want to find out the content of 's' or more appropriatly, the "exten" |
| 11:27.26 | mchou | trust me, the info is there |
| 11:27.32 | kaldemar | mchou: s has no content, it is just s. did you pass an extension to the provider in your register statement? |
| 11:27.39 | mchou | oh god |
| 11:27.46 | WIMPy | Then it's in a place/format that Asterisk doesn't understand. |
| 11:27.50 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 11:27.54 | *** join/#asterisk Buklov (~Buklov@mail.sapsun.su) |
| 11:28.01 | puzzled | hi |
| 11:28.14 | irroot | lo there puzzled |
| 11:28.15 | kaldemar | mchou: show a trace and we'll see if the info is there. |
| 11:28.23 | puzzled | hi irroot |
| 11:28.29 | irroot | haver you hugged your * today |
| 11:28.41 | mchou | kaldemar: no, it's clear you have no idea what I'm even talking about |
| 11:28.43 | kaldemar | mchou: now you're just shouting "give me" but not telling what you want. |
| 11:28.59 | mchou | I'm not shouting anything |
| 11:29.02 | puzzled | irroot: heh lemme put that on the TODO list |
| 11:29.07 | kaldemar | mchou: have it your way then, have fun. |
| 11:29.09 | mchou | a sip call arrives inbound |
| 11:29.13 | WIMPy | mchou: Then wait for more people telling you the same as we did. |
| 11:29.48 | mchou | dialplan says "exten => s,1,Dial(me)" |
| 11:30.04 | mchou | you thelling me s has no info what so ever? |
| 11:30.11 | mchou | telling* |
| 11:30.17 | WIMPy | Indeed |
| 11:30.26 | mchou | cause that's clearly bullshit |
| 11:30.32 | WIMPy | I told you at the beginning. |
| 11:30.46 | kaldemar | s is s. s is not something else. it is not a variable or a pointer to some information. |
| 11:30.47 | irroot | passes bucket of popcorn |
| 11:30.56 | WIMPy | If there's destination information available, you don't get to s. |
| 11:31.18 | mchou | cause if I say "exten=>s/1234,1, Dial(me)" how the hell does asterisk match that? |
| 11:31.56 | WIMPy | scratches head |
| 11:31.58 | mchou | lol |
| 11:32.01 | irroot | Wimpy what if DNID is set and it goes to s this happens ;) |
| 11:32.03 | mchou | QED |
| 11:32.03 | irroot | exten => s,1,Goto(${CALLERID(DNID)}) |
| 11:32.05 | kaldemar | mchou: when clid number is 1234 but there is no destination number. |
| 11:32.05 | irroot | love exceptions |
| 11:32.22 | resmo | How do I define callgroups ringing in extension.conf? |
| 11:32.25 | resmo | Any hints? |
| 11:32.29 | WIMPy | irroot: Hmm. How do you get that? |
| 11:32.36 | irroot | resmo see app_queue |
| 11:33.03 | WIMPy | resmo: Or dimply dial multiple channels (dest1&dest2) |
| 11:33.14 | mchou | kaldemar: so now you admit 's' carries CALLERID |
| 11:33.18 | irroot | WIMPy i get it from a particullarly retarded PBX that comes into s with DNID set |
| 11:33.44 | WIMPy | irroot: What technology? |
| 11:33.51 | irroot | mchou depends on the underlying channel driver |
| 11:33.59 | mchou | irroot: bingo! |
| 11:34.02 | irroot | WIMPy DAHDI PRI |
| 11:34.03 | WIMPy | mchou: Are you asking about callerid all the time??? |
| 11:34.03 | kaldemar | mchou: no. you're mixing things. |
| 11:34.07 | mchou | irroot: someone with a clue |
| 11:34.25 | WIMPy | irroot: That sounds like a bug to me. |
| 11:34.46 | mchou | lol |
| 11:34.48 | irroot | E1/PRI yeah but not on the * so much as the interop with the switch |
| 11:34.54 | resmo | irroot, hmm what is callgroup in sip.conf for then? |
| 11:35.05 | WIMPy | Or is it a redirected call with an oriignal destionation forwarded to none? |
| 11:35.23 | irroot | resmo call pickups the *8 function in features.conf |
| 11:35.27 | WIMPy | irroot: E1? DSS1? |
| 11:35.29 | irroot | to map a phone |
| 11:35.41 | irroot | E1 ETSI [but not so much ;)] |
| 11:35.50 | irroot | had to put immediate on |
| 11:35.58 | WIMPy | irroot: I'd like to see a trace of such a call. |
| 11:36.21 | irroot | dont have access to that site ATM |
| 11:36.48 | irroot | started running DISA on the chan till i spotted DNID was set |
| 11:36.56 | WIMPy | Does Asterisk act as TE or NT? |
| 11:37.24 | irroot | it was both bridging NT on 2 ports TE on 2 |
| 11:37.36 | irroot | the NT side was where the problem was |
| 11:37.51 | WIMPy | With immediate=yes, you might need a s,1,WaitExten. |
| 11:37.58 | WIMPy | Even if it maybe doesn't have to wait. |
| 11:38.27 | WIMPy | That would indeed usually need the WaitExten. |
| 11:38.39 | irroot | without immediate it was not completing the number with or without enbloc on the switch |
| 11:38.47 | irroot | even messing with the idd / pdd |
| 11:39.52 | WIMPy | Yes, dialing a number gan be quite a PITA with technologies that allow different methods of passing dialing information. |
| 11:40.22 | WIMPy | Which reminds me that I need to experiment with the Incomplete() app, I stumbled across the other day. |
| 11:40.41 | WIMPy | Maybe I can finally reformat dialed numbers that way. |
| 11:40.44 | irroot | have a [ddi] context that caters for all the oddness i have seen so far it is filled in the horror section at the library |
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| 11:46.10 | resmo | ic queue, nice but a bit of an overhead for now, just want to call a few phones at once. so I define an exten 200 wich dials 201.202, 203 right? |
| 11:46.58 | WIMPy | resmo: For example, yes. |
| 11:48.21 | resmo | hmm but how would I make them ringing at the same time? |
| 11:48.56 | WIMPy | Dial(sip/phone1&sip/phone2&...) |
| 11:49.19 | resmo | ahh thx |
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| 12:01.52 | resmo | asterisk is really ... awsome |
| 12:02.36 | cneb3000 | resmo: Have you played with telephony much before? |
| 12:03.22 | zkn | so when I receive a call via IAX2 channel to a SIP extension, IAX2 channel will not be able to react on DIALSTATUS responses? |
| 12:03.58 | WIMPy | zkn: Please elaborate. |
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| 12:05.20 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:07.56 | zkn | so, I want the call to failover according to DIALSTATUS, for example when a call via IAX2 channel to a local SIP device failed |
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| 12:09.25 | WIMPy | Well, conversion of failure information between channeltypes can be lossy, but generelly it should work. |
| 12:09.38 | resmo | cneb3000, I used to use askozia (which is a asterisk distro for embedded devices) but this is very low featured |
| 12:09.46 | *** join/#asterisk fish-bulb (~qcstewart@nat/digium/x-idveywjwzvgjpldg) |
| 12:09.46 | WIMPy | I'd assume that it should at least work as far as DIALSTSTUS goes. |
| 12:10.10 | resmo | cneb3000, now going the hard way learning asterisk plain naked :) |
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| 12:13.55 | irroot | wiseman says max ip in DHCP scope should never be more than broadcast - 1 .... |
| 12:13.57 | irroot | bang head on desk search for techie to molest |
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| 12:15.01 | WIMPy | knows an ISP who failed on that one. |
| 12:15.25 | WIMPy | That was ppp, not dhcp, but was fun anyway. |
| 12:19.17 | cneb3000 | resmo: good on yer! |
| 12:19.34 | zkn | Wimpy, but if the call fails in a specific context then I cannot send that call to another one to continue trying there, can I ? |
| 12:19.35 | cneb3000 | resmo: never heard of askozia.. will have a look! |
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| 12:20.10 | zkn | WIMPy, e.g. same => n,GotoIf($["${DIALSTATUS}" = "CHANUNAVAIL"]?failovertrunk,${EXTEN},1) |
| 12:21.16 | zkn | if i do include => failovertrunk in the previous context where a call is Dialed then it seems to continue trying |
| 12:21.32 | WIMPy | You can do everything you do otherwise. |
| 12:21.39 | *** join/#asterisk l2trace99 (~jr@rrcs-71-43-104-238.se.biz.rr.com) |
| 12:22.02 | WIMPy | Maybe a timeout thing? |
| 12:22.48 | zkn | hmm.. |
| 12:25.51 | irroot | exten => _XXX.,n,GotoIf($[${HANGUPCAUSE} != 88]?s-${DIALSTATUS},1) |
| 12:25.53 | irroot | exten => _XXX.,n,Dial(Local/${FWDNUM}@ftrunkproxy,${TOUT},wgt) |
| 12:25.55 | irroot | exten => _XXX.,n,Goto(s-${DIALSTATUS},1) |
| 12:26.24 | irroot | example of how to check for a incompat ISDN line and run it through a proxy channel ie local |
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| 12:41.21 | hudony | Hi, I have few general questions about installing an asterisk server with POT. I've already installed one with ip phones but never with POT. Anyone willing to answer them? Thank you! |
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| 12:46.03 | WIMPy | ~ask |
| 12:46.03 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
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| 12:50.33 | hudony | ok, here I go |
| 12:51.02 | hudony | We actually have about 15 phones lines |
| 12:51.20 | hudony | Do I need a T1 to replace them? |
| 12:51.29 | hudony | Or is there alternatives |
| 12:53.47 | ollii | hey, i have a problem with dahdi 2.4.1.2 (latest from svn), dahdi-extras (latest from tzafrir), libpri 1.4.11.5 and a hfc-s card. everything is loaded correctly. its signalling is pri_cpe_ptmp as it is connect to another isdn card from sangoma which has signalling NT...its used as an bridging internal s0. if i drop a call on hfc card it always says: Unable to receive TEI from network in state 3(Establish awaiting TEI)! |
| 12:54.00 | *** join/#asterisk jetsor (~jstorch@dynamic-acs-24-101-74-112.zoominternet.net) |
| 12:54.37 | ollii | i forgot to tell which asterisk is used: 1.8.4 |
| 12:55.33 | ollii | is there a known problem with libpri 1.4.11.5 or dahdi 2.4.1 or dahdi-extras (oslec, zaphfc) if a hfc-s card is used? |
| 12:55.56 | WIMPy | hudony: That would definitely make sense. |
| 12:56.53 | hudony | WIMPy: ok, I cannot just use a 1meg/sec standard cable connection? |
| 12:56.58 | WIMPy | ollii: Does the Sangoma support power saving? If so try to disable that. |
| 12:57.10 | WIMPy | That has caused many issues so far. |
| 12:57.39 | WIMPy | hudony: You can also go for an ITSP instead of an own PSTN connection. |
| 12:58.47 | WIMPy | ollii: Or do you have the other side as ptp, prehaps? |
| 12:58.48 | ollii | WIMPy: it also tried to connect it directly to telco line ... same behavior |
| 12:59.47 | WIMPy | ollii: Have you checked with pri debug if the card receives ANYthing? |
| 13:02.03 | WIMPy | Err, and teh telco line is ptmp? |
| 13:02.40 | ollii | i tried to also with ptp to our ptp telco line... |
| 13:03.03 | *** join/#asterisk Sertys (~sertys@89.252.247.42) |
| 13:03.37 | Sertys | guys, which is the most powerful/usable free billing/lcr manager for asterisk? |
| 13:04.42 | WIMPy | ollii: That looks like a problem at a lower layer. Does the card generate IRQs? |
| 13:05.27 | hudony | WIMPy: Ok and would it be better to change all our pot phones for ip phones so I dont have to buy analog cards? |
| 13:08.32 | WIMPy | hudony: That's a matter of taste and budget. Analog phones don't offer much comfort/functuonality. |
| 13:09.49 | WIMPy | Actually that may be true for sheap SIP phones as well. |
| 13:10.01 | WIMPy | s/ssheap/cheap/ |
| 13:10.06 | WIMPy | damn |
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| 13:10.24 | irroot | hudony comparing cost there is no win really a good ana phone + DAHDI cost +/- same as average IP phone per port |
| 13:10.27 | Sertys | !bash |
| 13:11.32 | ollii | WIMPy: maybe i startet wrong...is it a good idea to use latest libpri stable with asterisk 1.8.4 with dahdi 2.4.1 and dahdi-extra patches? |
| 13:11.51 | WIMPy | If you want it cheap, but ISDN phones. But you might have trouble obtaining them in NA. |
| 13:11.52 | thalesfp | Hello, I get problems while installing g.729? I dont have an interface named eth0 and the "register" returns Make sure that you have eth0 enabled. |
| 13:12.18 | WIMPy | ollii: I'd guess so. But it has been some weeks since I last tried dahdi. |
| 13:12.40 | cneb3000 | thalesfp: the g729 stuff needs to talk to Digiums servers over the internet. |
| 13:12.50 | cneb3000 | (..i think?) |
| 13:13.00 | cneb3000 | thalesfp: Does your box have access to the internet |
| 13:13.01 | cneb3000 | ? |
| 13:13.01 | Sertys | thalesfp: |
| 13:13.13 | Sertys | just strace the script |
| 13:13.20 | thalesfp | cneb3000: yes, but my interface is venet0:0 |
| 13:13.47 | WIMPy | thalesfp: Can't you rename or alias it? |
| 13:14.20 | Sertys | http://forum.openvz.org/index.php?t=msg&goto=35005& |
| 13:14.25 | WIMPy | Does it really need internet access or is it just comparing MAC adress? |
| 13:14.32 | thalesfp | WIMPy: I try it using vzctl whitout successfull |
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| 13:14.49 | Sertys | do u have access to the openvz switch? |
| 13:16.23 | WIMPy | thalesfp: ip link set ... name ...? |
| 13:16.39 | thalesfp | Sertys no |
| 13:16.53 | *** join/#asterisk gray_ (~Gray@unaffiliated/remnant13) |
| 13:17.01 | thalesfp | WIMPy i will try it |
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| 13:17.36 | Sertys | thalesfp: be careful, renaming your main interface may drop u out of your vps |
| 13:18.03 | Sertys | go to the register dir and grep eth0 * |
| 13:18.25 | Sertys | to check whether the interface option is controllable via some config option |
| 13:19.04 | zkn | is there a way I can specify what the default context is in extensions.conf ? |
| 13:19.42 | WIMPy | zkn: No you do that in the channels configs. |
| 13:21.06 | zkn | ok, but users.conf, it just automatically assumes I want to create alternateexts to [default] |
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| 13:22.11 | WIMPy | I don't know if that applies to the alternates as well, but there is an option in users.con to tell it the context under which to create extensions. |
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| 13:23.52 | WIMPy | Or at least I think so. It's been quite a while since I've had a users.conf. |
| 13:24.15 | WIMPy | I might mix that up with the regcontext. |
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| 13:35.14 | ollii | wimpy: i just recompiled from another source ( http://sourceforge.net/projects/dahdi-zaphfc/ ) and i got a working call |
| 13:35.28 | ollii | WIMPy: now i just have to include oslec again |
| 13:36.01 | WIMPy | Ja, too many versions of zaphfc :-( |
| 13:36.11 | WIMPy | I used the one from google code. |
| 13:36.35 | *** part/#asterisk sekil (~sekil@80.93.247.26) |
| 13:38.53 | ollii | yes, indeed |
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| 13:39.21 | k3asd` | hi |
| 13:39.29 | ollii | different opinions how to integrate an hfc-s card with zaphfc |
| 13:39.45 | cneb3000 | he k3asd` |
| 13:39.49 | WIMPy | And even more without :-) |
| 13:39.50 | ollii | hi |
| 13:39.51 | cneb3000 | hey* |
| 13:40.00 | ollii | oh yeah |
| 13:40.06 | ollii | chan_lcr,misdn and so on |
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| 13:40.15 | ollii | open source rocks! |
| 13:40.16 | ollii | ;) |
| 13:42.29 | irroot | uses mISDN v1 with 2.6.38 small change to asterisk configure.ac and change the header files :) |
| 13:43.13 | WIMPy | uses v2 with LCR. |
| 13:44.17 | irroot | needs to harass WIMPy perhaps when i try v2 again it uses sntp right ?? |
| 13:44.20 | WIMPy | I'd still like to get hold of an HFC-E1 card. |
| 13:44.22 | tzafrir_laptop | hmm.. my kernel git tree is called "linux-2.6" |
| 13:44.49 | WIMPy | sntp? |
| 13:44.58 | irroot | tzafrir_laptop see that 2.6.40 is now linux 3.0 |
| 13:45.01 | ollii | beronet is using hfc e1 chips |
| 13:45.25 | irroot | WIMPy like sangoma with sockets net proto ?? |
| 13:45.32 | WIMPy | I know, but the HFC-E1 cards get more expensive than the Digium ones. |
| 13:45.33 | tzafrir_laptop | Anybody has that? |
| 13:45.52 | tzafrir_laptop | There should be DAHDI drivers for it (cwain) |
| 13:46.08 | WIMPy | irroot: It uses sockets, yes. But I doubt any similarities with the Sangoma stuff. |
| 13:46.42 | WIMPy | dahdi supports HFC-E1? Or what do you say? |
| 13:47.02 | irroot | yeah it will support it via mISDN im sure |
| 13:47.23 | WIMPy | Yes, but that's not dahdi. |
| 13:48.04 | irroot | ah libpri support for HFC should be able to if it does not |
| 13:48.15 | tzafrir_laptop | WIMPy, Junghanns wrote DAHDI drivers for it (CWAIN: Channen [driver] Without An Interesting Name) |
| 13:48.23 | WIMPy | libpri doesn't do hardware. |
| 13:48.32 | irroot | yeah getting there |
| 13:48.48 | irroot | backporting the mISDN stuff should be probable |
| 13:49.03 | tzafrir_laptop | This is the simple HFC-[2]E1 card. Not any embedded system built on top of it |
| 13:49.06 | WIMPy | Ok, so HFC-E1 would be a good idea as you get both driver options. |
| 13:49.19 | irroot | talking to the HFC E1/BRI should be similar |
| 13:49.23 | WIMPy | irroot: Backport? |
| 13:49.52 | WIMPy | E1 != BRI |
| 13:49.58 | irroot | look at the diff in the hfc drivers for mISDN and backward engineer the dahdi based on zaphfc |
| 13:50.31 | irroot | the chip does the heavy lifting the interface is quite simple |
| 13:50.42 | WIMPy | Still not sure, what you're trying to suggest. |
| 13:50.52 | WIMPy | The chip doesn't do much. |
| 13:51.29 | irroot | take the mISDN kernel drivers to work out the diff between the hfc-s and e1 |
| 13:52.06 | WIMPy | The data sheets from CCD are available for their chips. |
| 13:52.22 | tzafrir_laptop | WIMPy, IIRC the HFC-E1 only works with E1 ISDN (PRI) and handles the HDLC encoding |
| 13:52.27 | irroot | as they done by bero (c richter) it is good ref |
| 13:52.44 | tzafrir_laptop | So in a way it is kind of like BRI |
| 13:52.48 | irroot | yeah euro only im fine with that :) |
| 13:52.52 | WIMPy | Yes, it might have a HDLC framer, but that's about it. |
| 13:53.18 | tzafrir_laptop | In that there's not much to choose from |
| 13:53.19 | irroot | yip its a bri of sorts with 30 channels ;) |
| 13:53.40 | WIMPy | ... called PRI. |
| 13:54.20 | WIMPy | And the PRI is easier as it doesn't support ptmp. |
| 13:55.40 | WIMPy | And I haven't heard about L1 deactivation on PRIs, either. |
| 13:56.15 | irroot | HDLC error is similar to L1 loss ?? |
| 13:56.21 | tzafrir_laptop | PtP / PtMP is at a higher level |
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| 13:56.49 | tzafrir_laptop | HDLC error is often (normally?) due to a Layer 1 error |
| 13:57.20 | WIMPy | tzafrir_laptop: Sure, but if you only want PRI support, you don;t have to care about that. |
| 13:57.28 | tzafrir_laptop | With other channels you normally don't care about a missing bit or 2. With the D channel you see it right away |
| 13:57.29 | irroot | L1 Loss will always cause HDLC but not always other way round |
| 13:57.44 | WIMPy | HDLC abouts can also be caused by IRQ handling issues. |
| 13:58.11 | tzafrir_laptop | WIMPy, It might be caused by a host of potential issues |
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| 14:14.53 | hudony | To link 16 POTS to an asterisk server, Should I use a voip gateway or put pci cards into the asterisk server? Which one is better and why? |
| 14:15.07 | *** join/#asterisk bl0b` (~vsfdfsdfs@host98-89-dynamic.53-79-r.retail.telecomitalia.it) |
| 14:15.15 | bl0b` | hi |
| 14:16.13 | WIMPy | hudony: That's a big one. |
| 14:16.21 | hudony | :( |
| 14:17.21 | WIMPy | External units are easier to replace and can be located elsewhere. PCI cards might be more user friendly as they allow overlap dialing. |
| 14:17.33 | bl0b` | Someone know why the H264 file created by VoiceMail did not get recognized by anysoftware converter\player ? (like mplayer, ffmpeg.. and the "file" command which return "msg0000.h264: data" instead of "JVT NAL sequence, H.264 video, baseline @ L 13" |
| 14:18.01 | *** part/#asterisk Joe_CoT (~joecot@pdpc/supporter/active/joe-cot) |
| 14:18.19 | irroot | hudony i have sites with 24 port cards there power concerns and cableing |
| 14:18.33 | irroot | also have sites with SPA-8000 linksys 8 ports |
| 14:18.48 | irroot | put in middle of desks easier to wire |
| 14:19.15 | irroot | have site with 16 spa 2102's in same area as they lightning prone |
| 14:19.25 | irroot | replace couple a year |
| 14:19.40 | irroot | hope that helps |
| 14:19.56 | hudony | ok but basically, I mean regarding performance : cards and gateways are almost identical? |
| 14:20.12 | hudony | Regarding echo cancellation etc. |
| 14:20.37 | WIMPy | You can get good and bad ones either way. |
| 14:20.52 | irroot | if you use a ITSP with G.729 its a bounus as you dont need licence and transcoding is done out of core |
| 14:21.27 | irroot | echo cancelation is only needed when there is a hybrid/balun in the path |
| 14:22.00 | hudony | ok |
| 14:22.25 | hudony | While comparing gateways and cards, I also found channel bank |
| 14:22.32 | hudony | Seems just like a voip gateway to me |
| 14:23.10 | hudony | oh...seems to connect to the pbx using t1 and not rj-45 cable |
| 14:23.16 | hudony | Is it that? |
| 14:23.52 | irroot | the rhino channel bank rocks costs more as it needs T1 but worth it for multi floor enviroments |
| 14:24.09 | irroot | put one on each floor run one T1 cable to server room |
| 14:24.44 | Rholk | Hi, i have a little problem compiling chan_dahdi, for an unknown reason, all my "can use" elements or going to "depends on" in menuconfig. I already runned the pre-req script. Why all "can use" are considered as "depends on"? |
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| 14:25.59 | hudony | irroot: ok I see |
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| 14:33.41 | hudony | I'm looking at the Mediatrix 4116 MGCP voip gateway and it says 16 fxs ports....but it allow up to 24 simultaneous calls....What is it that I dont get ? :( |
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| 14:37.42 | WIMPy | Maybe calls on hold? |
| 14:40.20 | hudony | It could be I guess....They also have 3 products 8, 16 and 24 fxs ports |
| 14:40.35 | hudony | maybe they are using the same hardware behind it so it can always handle 24 ports |
| 14:40.46 | hudony | altough limited by the number of fxs ports |
| 14:40.49 | hudony | weird |
| 14:40.59 | WIMPy | Possibly. Do they all state 24 calls? |
| 14:41.55 | hudony | yes |
| 14:41.58 | hudony | so that must be it |
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| 14:42.39 | hudony | btw, have a question about max simultaneous call you can make : if using one T1, you have a max of 24 right? |
| 14:42.48 | hudony | because its 24 pairs of copper wires |
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| 14:43.54 | hudony | Am i right? |
| 14:43.56 | irroot | yeah it uses the ports with no D channel as far as i remember |
| 14:44.10 | irroot | all fxs not pri signalling |
| 14:44.31 | Tech_Travis | hudony: A T1 is only 2-pair of copper wires. |
| 14:44.48 | Tech_Travis | hudony: but there are 24 DS0 channels on a T1 |
| 14:45.53 | hudony | oh ok |
| 14:46.54 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 14:47.05 | hudony | But what about ip phones? Lets say you have 1meg/sec cable connection... and 24 ip phones...I know it will depend of the voice codecs used but in a general way, do you only have to calculate how much kb the used codec takes per sec * number of phone |
| 14:47.23 | hudony | So have an approx. of how many person will be able to talk at the same time |
| 14:47.26 | hudony | ? |
| 14:47.45 | irroot | voip bandwidth calculator <- google is your friend |
| 14:47.59 | hudony | Ok but I mean its really like that...nothing else... |
| 14:48.04 | hudony | just a math calculation |
| 14:48.28 | WIMPy | Well, yes and no. |
| 14:48.31 | hudony | :( |
| 14:48.54 | WIMPy | As for the bandwidth: yes, but you also want to make sure that bandwidth is avaliable at all times. |
| 14:49.32 | hudony | ya ok |
| 14:49.59 | hudony | So of course, you have to get a plan that gives you enough bandwith quota so you don't pay extra fees at the end of the month |
| 14:50.04 | WIMPy | Or use a dedicated line only for voip and no other internet traffic. Otherwise you need a QoS/TC/whatever capable router. |
| 14:50.16 | hudony | for T1, do you have to take bandwith consideration into account or do you only pay a monthly fee? |
| 14:50.33 | WIMPy | That might even be an issue on the LAN. |
| 14:51.40 | WIMPy | Depends on your plan, obviousely. |
| 14:51.47 | WIMPy | Plus that of an ITSP. |
| 14:51.53 | hudony | ok |
| 14:51.57 | WIMPy | Anyway: BBL |
| 14:52.56 | irroot | cheers WIMPy im home soon too |
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| 15:16.44 | bl0b` | Someone know why the H264 file created by VoiceMail did not get recognized by anysoftware converter\player ? (like mplayer, ffmpeg.. and the "file" command which return "msg0000.h264: data" instead of "JVT NAL sequence, H.264 video, baseline @ L 13" |
| 15:19.59 | engrxyz | anyone for any howto on how to make asterisk work with ultramonkey? |
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| 15:41.34 | puzzled | engrxyz: google does not work for you? www.danielaliaman.com/blog/files/ultramonkeyasterisk.pdf |
| 15:43.16 | cneb3000 | engrxyz: What are you trying to achieve with UltraMonkey? |
| 15:43.27 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 15:44.13 | engrxyz | puzzled, : that i am reading now but doesn't work |
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| 15:44.27 | engrxyz | puzzled, : the virtual ip can't be ping |
| 15:45.07 | engrxyz | cneb3000, : two asterisk server that will receive sip traffic from the load-balanced ultramoneky/heartbeat/ldirectord |
| 15:45.34 | cneb3000 | engrxyz: what does it say when you try to ping the virtual IP? |
| 15:46.30 | engrxyz | just don't reply |
| 15:47.10 | cneb3000 | engrxyz: Copy and paste exactly what it says when you ping it. |
| 15:47.29 | cneb3000 | engrxyz: to pastebin.com, if it's more than 3 lines :) |
| 15:48.07 | mchou | what's there to paste? He told you ping timed out |
| 15:48.29 | cneb3000 | mchou: no, he said 'ping didn't reply'. that could mean a few things. |
| 15:51.10 | cneb3000 | engrxyz: also.. make sure you follow page3 carefully. it's the important bit! |
| 15:51.22 | engrxyz | what page |
| 15:51.40 | engrxyz | ah that link |
| 15:51.47 | cneb3000 | engrxyz: page 3 of this document you said you read ---> http://www.danielaliaman.com/blog/files/ultramonkeyasterisk.pdf |
| 15:52.23 | engrxyz | yeah i did that |
| 15:52.31 | engrxyz | modules were loaded fine |
| 15:52.39 | cneb3000 | how about the last page? about modifying some of the NIC config? |
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| 15:59.48 | engrxyz | the floating ip seems to be pingable now |
| 16:00.06 | engrxyz | but when i do "telnet <floating_ip> 5060" |
| 16:00.11 | engrxyz | it won't respond |
| 16:00.24 | emsLinux | Good morning, i have a question about hardware... How important is the processor in the Asterisk process, could be this AMD Athlon II X2 Neo K325 Dual Core 1.3 GHz processor, enough for a big asterisk central, maybe with a good amount of memory? |
| 16:00.57 | cneb3000 | engrxyz: did you do anything to get it to responding to ping normally? |
| 16:01.18 | engrxyz | i even tried registering a softphone using "<floating_ip>:5060" |
| 16:01.19 | psilikon | emsLinux, sure that will work but it depends on what you want to do with the system. Google 'asterisk dimensioning'. |
| 16:01.37 | engrxyz | cneb3000, : well nothing at all |
| 16:01.44 | engrxyz | i just stop and restart both the heartbeat |
| 16:01.48 | engrxyz | and the ldirectord daemons |
| 16:02.02 | mchou | engrxyz: there's no need to get so complicated |
| 16:02.15 | mchou | engrxyz: ucarp works just fime |
| 16:02.19 | mchou | fine* |
| 16:02.46 | engrxyz | but the asterisk or any sip servers won't register any client |
| 16:02.58 | engrxyz | as is on page 11 |
| 16:03.07 | engrxyz | i have asterisk, opensips, and even freeswitch |
| 16:03.14 | mchou | lol |
| 16:03.20 | mchou | overkill |
| 16:03.26 | engrxyz | if i register a sip softphone directly in any of those |
| 16:03.27 | cneb3000 | ^^^ yea.. complicated! |
| 16:03.28 | engrxyz | it will register |
| 16:03.44 | engrxyz | not really complicated just one at at time sip server |
| 16:04.02 | engrxyz | i tested the setup in all the three, seems not working though |
| 16:04.11 | mchou | lol |
| 16:04.24 | mchou | that's the definition of complicated |
| 16:04.42 | engrxyz | is it? |
| 16:04.47 | mchou | 3 instances NOT working |
| 16:04.57 | engrxyz | not 3 instances |
| 16:05.02 | engrxyz | just 1 at at time |
| 16:05.50 | engrxyz | if the <floating_Ip> works as it shoud, it should respond to sip ports |
| 16:05.56 | mchou | get 1 pair failover working first |
| 16:06.05 | engrxyz | failover is working |
| 16:06.24 | engrxyz | ip addr sh eth0 |
| 16:06.39 | engrxyz | ldirectord ldirectord.cf status |
| 16:06.50 | engrxyz | ipvsadm -L -n |
| 16:07.08 | engrxyz | and even the LVSSYNCDaemon Swap |
| 16:07.25 | engrxyz | all seems to be working fine |
| 16:07.32 | mchou | no |
| 16:07.51 | engrxyz | if i stop the heatbeat daemon on load balancer 1, the floating ip failback to load balancer 2 |
| 16:07.52 | mchou | when I say working I mean registering to virtual ip |
| 16:08.13 | engrxyz | well in that case, it doesn't work |
| 16:08.19 | engrxyz | maybe i need some hint from you guys |
| 16:08.22 | mchou | regardless where the real servers are |
| 16:09.08 | engrxyz | is the virtual configured as "virtual=<floating_ip>:506"? |
| 16:09.17 | emsLinux | I have another stadistical question, how many user can handle a 8 line trunk with normal use times, not too much traffic, just normal... |
| 16:09.22 | engrxyz | or 5060 |
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| 16:37.26 | cneb3000 | late shift tonight.. tum tee tum.. |
| 16:38.39 | orn | jesus... i just can't figure out what's wrong... AMI isn't outputting any events until I try to enter some data (or press enter)... I have this exact asterisk deployment all over the place, running on the same asterisk version, and this is only a problem in this specific install |
| 16:38.54 | orn | i've even tried upgrading asterisk to another version and it's still like this |
| 16:39.00 | orn | and i've run out of ideas |
| 16:39.19 | orn | manager.conf is identical to other places where it works |
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| 16:40.09 | orn | i'm accepting any ideas from anyone at this point |
| 16:40.50 | Qwell | orn: How are you connecting? |
| 16:41.14 | orn | i'm using telnet at the moment, but the script that wasn't working (thus alerting me to the problem) is just opening a socket |
| 16:41.37 | Qwell | and if you telnet from the same client to a different server, it works? |
| 16:41.50 | orn | let me check |
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| 16:44.38 | orn | yes, if i telnet from this particular server to an AMI on another server it works |
| 16:44.49 | orn | so it seems it isn't the client on the server |
| 16:44.50 | Qwell | So it's the server |
| 16:46.05 | orn | seemingly, and i would guess that it is a configuration issue |
| 16:46.19 | orn | but manager.conf is exactly the same as on multiple other servers |
| 16:46.53 | orn | and it's consistently broken across two different asterisk versions (both of whom have working AMI) |
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| 16:47.30 | Qwell | are you using any event masks? |
| 16:47.36 | Qwell | setting events: on? |
| 16:47.42 | Qwell | (2 different questions) |
| 16:48.20 | orn | yes, setting events: on |
| 16:48.26 | orn | what are event masks? |
| 16:48.33 | Qwell | You'd know if you were using them |
| 16:48.47 | orn | also, in case you didn't see above, as soon as i try to enter any commands or enter an empty newline, all the events appear |
| 16:48.50 | orn | so they're buffering up for some reason |
| 16:49.02 | Qwell | What versions of Asterisk have you tried, exactly? |
| 16:49.07 | orn | 1.6.1.11 and 1.6.2.11 |
| 16:49.25 | Qwell | upgrade :p |
| 16:49.55 | orn | well, i already upgraded from 1.11 to 2.11, and i'm reluctant to go farther as i haven't tested it and this particular installation is on a rather large live installation :) |
| 16:50.06 | Qwell | 1.11 to 2.11 is a VERY BIG upgrade |
| 16:50.19 | Qwell | much much larger than 1.6.2.18 would be |
| 16:50.26 | irroot | orn 1.6-> 1.8 is not too bad a path |
| 16:50.39 | irroot | 1.4->1.6 lots of changes were required |
| 16:50.53 | orn | but i'm already running 1.6.2.11 in many locations and i've confirmed it to work well |
| 16:51.06 | orn | 1.8 however has a bug that isn't being addressed with UTF8 characters in the From header |
| 16:51.17 | orn | as far as i know it hasn't been addressed at least |
| 16:51.28 | Qwell | Have you reported it? |
| 16:51.28 | irroot | 1.8 has some deadlocks most seem resolved ... |
| 16:51.31 | orn | yes |
| 16:52.15 | irroot | any one use TLS with snom ?? |
| 16:52.29 | orn | the funny thing is that there was a discussion as these particular changes were being made (escaping characters when they aren't part of the ascii alphabet), and there was a guy pointing out these problems all along the way but the changes went through anyway :) |
| 16:54.42 | orn | actually, thinking on it, the bug was introduced before, but is only encountered when pedantic = yes, and pedantic = yes is default in 1.8, which is why i encountered the bug |
| 16:55.09 | irroot | set pedantic = no in your local config ?? |
| 16:55.22 | orn | irroot: yeah, i know, but this isn't the problem :) |
| 16:55.53 | orn | irroot: there was also a patch fixing this bug submitted already, but it hasn't made its way into the codebase yet |
| 16:56.28 | orn | Qwell: I'm going to try to upgrade to 1.8, if only to confirm for a brief moment whether the weird AMI behavior is still there |
| 17:01.45 | irroot | what is the # for the bug ?? |
| 17:04.14 | orn | for the UTF-8 URI encoding bug? |
| 17:04.51 | orn | 18298 |
| 17:07.00 | irroot | ah and you have tested this patch ?? |
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| 17:08.07 | orn | yes |
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| 17:09.52 | orn | Qwell: Same issue with AMI on 1.8.4.1 |
| 17:10.09 | orn | This _has_ to be some configuration issue somewhere, but I really can't think of where |
| 17:10.24 | orn | Isn't manager.conf the only configuration file pertaining to AMI ? |
| 17:10.46 | irroot | well then apply it to your path ;) |
| 17:12.13 | orn | irroot: Haha, well, duh :) But as I said, this is an unrelated issue... it was the reason I wasn't going to upgrade to 1.8, but then I remembered the patch |
| 17:12.29 | orn | irroot: But as I just said, the AMI issue I'm having is still there in 1.8 so there's no reason to do so anyway |
| 17:12.43 | *** part/#asterisk ollii (~risker@2001:470:1f15:1384:62eb:69ff:fe31:b4) |
| 17:13.02 | orn | irroot: At least it's not a fix for my current problem ;) |
| 17:14.30 | irroot | ah |
| 17:23.45 | zkn | orn, u sure you haven't set noload to some importand module or smth... when you compare with "module show" on your servers, do you see some differences ? |
| 17:25.32 | zkn | at this point i would do: make uninstall ; make distclean ; make clean ; ./configure ; make menuselect (check all needed) ; make ; make install .. and see if that fixed the issue |
| 17:27.36 | orn | zkn: yes, same modules loaded |
| 17:27.58 | zkn | i once had a somewhat similar issue with Gosub and related functions never appeareing even though app_stack was installed and loaded |
| 17:28.18 | zkn | ended up reinstalling |
| 17:31.28 | orn | The operating system or Asterisk? |
| 17:31.47 | zkn | asterisk |
| 17:32.07 | orn | I've tried on three different asterisk versions already :/ |
| 17:32.34 | orn | This seems so much like a configuration issue, but I just can't find anything that would explain this behavior |
| 17:34.28 | zkn | although, with problems that personally I am experiencing with my Asterisk - i think i will be reinstalling my operating system, too, soon - i have also started from earlier versions and now try to use 1.8.4 but every time I restart it, I need to rotate astdb, otherwise I will get Segmentation Fault |
| 17:35.19 | zkn | and also, other installations have no such issue |
| 17:35.50 | zkn | configuration issue as well... i dont know... |
| 17:50.01 | heffer | hi there. i have a problem with my setup consisting of an asterisk 1.8 server with a budget HFC-S PCI card (vzaphfc) |
| 17:50.15 | Qwell | Problem with a budget card? Say it ain't so! |
| 17:50.35 | heffer | well i'm afraid it is so :) |
| 17:50.45 | Qwell | Never thought I'd see the day. |
| 17:51.54 | heffer | ;) |
| 17:52.14 | Qwell | No, but seriously, "a problem" isn't descriptive in the least. |
| 17:52.18 | WIMPy | heffer: Multiport HFC cards are supported by wcb4xxp now. |
| 17:52.41 | heffer | i was trying to word my problem a bit more just now :) |
| 17:53.06 | WIMPy | "make it so!" |
| 17:54.09 | heffer | basically it seems to be inducing noise into my NTBA because at some random point callers can't understand me when calling from a phone connected to the PBX that is connected to the same NTBA as the Asterisk server |
| 17:54.30 | heffer | however, callers can understand me if calling from the Asterisk server |
| 17:54.49 | heffer | calling from SIP to PSTN that is |
| 17:55.06 | WIMPy | Define "some point". |
| 17:55.34 | heffer | anything from 2 hours to 2 days |
| 17:55.40 | WIMPy | But as I already sdaid, you shouldn't need vzaphfc any more. |
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| 17:56.09 | heffer | that would mean i'd need to get a multiport HFC-S Card, right? |
| 17:56.11 | WIMPy | But you need a little patch if you want to use NT mode on a jumperless card. |
| 17:56.59 | WIMPy | Oh, sorry I thought it was one. |
| 17:57.07 | heffer | i'm using TE mode right now. no plans for using NT anytime soon |
| 17:57.27 | heffer | no, i have one of these cheap cologne chip ones from ebay :) |
| 17:57.29 | heffer | sorry ;) |
| 17:57.40 | WIMPy | Well, for the single port cards, you find versions of the zaphfc module on google code or at |
| 17:57.47 | WIMPy | http://sourceforge.net/projects/dahdi-zaphfc/ |
| 17:58.52 | heffer | cool, i didn't know that link yet |
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| 17:59.01 | WIMPy | Personally, I prefer to use the standard kernel drivers (misdn v2) with LCR. |
| 17:59.38 | heffer | i'm using tzafrir's repository at the moment. but it has a bug that always needs to be patched when doing a fresh clone |
| 18:00.17 | heffer | WIMPy: i felt the misdn + lcr combo to be a weird construct |
| 18:00.38 | heffer | i like how dahdi directly integrates with asterisk |
| 18:00.45 | cmw | i'm having a problem with gtalk on asterisk 1.8.2.3 where it randomly doesnt pickup the call. I see it coming in and being answered by asterisk but the dialing party just hears ringing until googlevoice goes to voicemail. Any ideas? |
| 18:00.52 | WIMPy | Yes, I thought so too, but it has served me well on two locations. |
| 18:02.23 | WIMPy | No more patching, neither for single nor multi port cards. But you need LCR from GIT with 1.8. |
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| 18:04.36 | heffer | i thik i'll give it a try then |
| 18:04.49 | heffer | thanks for your help WIMPy |
| 18:05.13 | WIMPy | Dahdi has the killer feature however: It supports ECT. |
| 18:05.37 | WIMPy | But that's probably less interesting if only using one TE port. |
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| 18:29.29 | heffer | WIMPy: i think i found out what my problem is |
| 18:29.30 | heffer | http://stkn.techmage.de/archives/200 |
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| 18:31.47 | nny | for all my fllow polycommunists out there, I am working on enabling buddy watch and persistent volume in 3.3.1's config revisions. I have http://pastebin.com/r7k4nEqF (sip.cfg snippet) Can anyone check and see if this is what the admin guide intends as far as format and placement? |
| 18:31.53 | nny | fllow/fellow |
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| 18:59.35 | cneb3000 | hm... |
| 18:59.48 | cneb3000 | dont suppose anyone has an archive of speaker videos from astricon 2008? |
| 18:59.58 | *** join/#asterisk zsasz (~zsasz@unaffiliated/zsasz) |
| 19:00.09 | cneb3000 | in particular the video that went with this ---> why-cluster-an-introduction-to-asterisk-clustering-and-database-integration-astricon-2008.pdf |
| 19:01.06 | russellb | cneb3000: was that leif madsen? |
| 19:01.16 | Qwell | russellb: I'm sure |
| 19:01.17 | russellb | the database chapter in ATDG is probably a lot of the same content |
| 19:01.19 | russellb | ~newbook |
| 19:01.19 | infobot | Please see ~thebook for more information about Asterisk: The Definitive Guide |
| 19:01.22 | russellb | ~thebook\ |
| 19:01.24 | russellb | ~thebook |
| 19:01.24 | 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://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 19:01.55 | cneb3000 | ^^^ thanks guys. I've read that. Thought it'd be interesting to have it explained in a new way. But if he did both then it's all the same. |
| 19:01.56 | cneb3000 | Thanks |
| 19:03.35 | cneb3000 | talking of Astricon. It looks fantastic. I wish I could afford to go! |
| 19:04.08 | cneb3000 | Amoocon is my next closest bet (England > Germany). Which doesn't look all that bad. |
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| 19:16.42 | heffer | is hangupdetect needed on BRI links? |
| 19:17.06 | Qwell | no |
| 19:17.49 | heffer | because the channel isn't hung up after the call is disconnected |
| 19:17.59 | heffer | thus causing distortion on the line |
| 19:18.04 | *** join/#asterisk USCooler25 (~Cooler@75-145-181-193-Illinois.hfc.comcastbusiness.net) |
| 19:22.37 | heffer | is there a way to specifically close a dahdi channel after hanging up |
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| 20:05.51 | heffer | WIMPy: i finally found the cause for my noise on the line |
| 20:06.23 | heffer | it's just that asterisk never really closes the channels |
| 20:06.53 | heffer | and it only started when i started using Asterisk 1.8 and libpri 1.4.12-beta3 |
| 20:07.12 | heffer | Asterisk 1.6 and libpri 1.4.11 are fine |
| 20:07.47 | heffer | and i thought it might have something to do with a bug you filed: https://issues.asterisk.org/view.php?id=17865 |
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| 20:54.43 | ruben23 | hi guys any idea how to create a dial plan for a france Sip trunk..? |
| 21:02.53 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 21:06.11 | ruben23 | hi anyone |
| 21:10.54 | *** join/#asterisk rue_work (~root@h24-207-19-104.cst.dccnet.com) |
| 21:11.10 | rue_work | is 1.8.4.1 'trouble free' ? |
| 21:11.23 | rue_work | should I be using 1.6 to be safe |
| 21:12.14 | AlexForster | really need suggestions of what to look at- audio *to* our sip trunk (going over the public internet) is choppy, but audio *from* the sip trunk is clear |
| 21:12.39 | AlexForster | just started today, channelstats shows no jitter and no dropped packets |
| 21:12.53 | Freeaqingme | rue_work: just consider that development of 1.6 has stopped some time ago, and that 1.8 is a successor of 1.6 |
| 21:13.12 | rue_work | Freeaqingme which means 1.6 prolly worked pretty well |
| 21:13.20 | Freeaqingme | and 1.8 even better |
| 21:13.32 | rue_work | ok, well the last system I did was 1.4, so I expect to be completely alienated again |
| 21:13.48 | AlexForster | timing test is fine, top shows very very low utilization, iftop also shows very low utilization |
| 21:14.29 | rue_work | did any of the config formats change since 1.4? |
| 21:14.36 | rue_work | to 1.8 |
| 21:15.40 | rue_work | ok, let me move on more, where is the best place for instructions on setting up a sagnoma card? |
| 21:15.50 | rue_work | I presume it dosn't use dahdi |
| 21:17.10 | RypPn | http://wiki.sangoma.com/wanpipe-linux-asterisk-install |
| 21:22.07 | rue_work | why does this site talk about zaptel? |
| 21:22.14 | Freeaqingme | <PROTECTED> |
| 21:22.19 | Freeaqingme | question being asked daily... |
| 21:23.00 | rue_work | zaptel went away with 1.2, this page regarding sagnoma is alking about zaptel... |
| 21:23.04 | WiretapWork | AlexForster, probably your upstream then |
| 21:23.12 | RypPn | 1.6 is EOL simples |
| 21:23.37 | rue_work | ok fine, I'm going 1.8, but I'm confused about the sagnoma thing |
| 21:23.55 | rue_work | it looks from this page I'z pointed at, like we use zaptel drivers |
| 21:24.03 | WiretapWork | rue_work, prepare to want to bash your brains out, Sangoma A cards are a nightmare |
| 21:24.06 | RypPn | use dahdi |
| 21:24.18 | rue_work | its a B600 |
| 21:24.30 | WiretapWork | not really much different |
| 21:24.33 | WiretapWork | same driver package |
| 21:24.42 | WiretapWork | which doesn't compile for debian/ubuntu kernels |
| 21:24.46 | rue_work | well it cnt be as much a nighmare as the digium card I tried last time |
| 21:24.59 | WiretapWork | it can be more |
| 21:25.05 | rue_work | hmm I can compile a new debian kernel |
| 21:25.23 | WiretapWork | you can |
| 21:25.28 | rue_work | the audio quality of the digium card is horrid, were sagnoma on this one |
| 21:25.30 | WiretapWork | but you can't compile sangoma drivers against one |
| 21:25.53 | rue_work | maybe I should look at whats on teh cd that came with the card...? |
| 21:26.00 | WiretapWork | good luck |
| 21:26.10 | WiretapWork | I have a machine with a Sangoma A500 in it for work |
| 21:26.18 | WiretapWork | I have basically given up on getting it working |
| 21:26.26 | WiretapWork | especially after I saw what we're paying for BRIs |
| 21:26.42 | rue_work | I have a pots card here |
| 21:27.28 | RypPn | sangoma wanpipe latest tarball won't build against any kernel newer than 2.6.36 |
| 21:28.06 | rue_work | debian, its only 2.6.32 |
| 21:28.29 | WiretapWork | I can assure you it won't compiel against anything newer than 2.6.28 :P |
| 21:28.40 | RypPn | your talking mince |
| 21:28.54 | spck | hi guys, I'm having trouble with outgoing DTMF on DAHDI channels, internally they work fine, and loadzone=us and defaultzone=us are set |
| 21:30.57 | ruben23 | hi guys any idea how to create a dial plan for a france Sip trunk..? |
| 21:31.05 | rue_work | wonders what wanpipe is |
| 21:31.31 | *** join/#asterisk ketas- (ketas@ketas6-sixxs.si.pri.ee) |
| 21:31.32 | WiretapWork | the drivers |
| 21:32.44 | rue_work | but its says its for dahdi |
| 21:32.51 | rue_work | so I need wanpipe and dahdi? |
| 21:33.28 | WiretapWork | <-- has never got a sangoma card to work, full stop |
| 21:33.45 | rue_work | thats ok, everyone else says they are less troubel than digium cards |
| 21:38.55 | rue_work | http://wiki.sangoma.com/sangoma-wanpipe-b600 shoudl I just see 'zaptel' as 'dahdi' ? |
| 21:41.58 | rue_work | the driver says its zaptel / dahdi |
| 21:44.38 | rue_work | hahahah I forgot how to build asterisk |
| 21:44.54 | rue_work | stares blankly at the list of files from teh tgz |
| 21:47.33 | beek | ~book |
| 21:47.33 | infobot | For more information about the Asterisk book, see ~thebook |
| 21:47.39 | beek | ~thebook |
| 21:47.39 | 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://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 21:47.48 | beek | rue_work: ^^^^^^^^ ;) |
| 21:47.56 | rue_work | yea yea I know |
| 21:48.33 | spck | no period on the book url |
| 21:57.26 | carrar | rue_work, start by logging out and calling your admin :) |
| 21:57.50 | rue_work | every time I try to call him my phone rings |
| 21:58.08 | carrar | trying leaving a extremely rude message |
| 21:58.35 | rue_work | I ot thru menuconfig |
| 21:58.45 | rue_work | so this is good |
| 21:58.58 | rue_work | see how I do with teh dahdi drivers |
| 21:59.18 | carrar | start with the README |
| 21:59.24 | rue_work | I did! |
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| 22:03.55 | *** join/#asterisk michael-i (~anonymous@204.11.230.58.static.etheric.net) |
| 22:05.58 | michael-i | Using Asterisk 1.8.4.1 and cdr_sqlite3_custom and the sampe config file, I'm getting a database full of doubled columns. One column has the field name "calldate" and the other just has an integer value for the field number "1". How can I prevent that? No reason to store things twice. |
| 22:06.41 | rue_work | mmm recompile the kernel eh? |
| 22:06.42 | rue_work | hmm |
| 22:07.03 | michael-i | Some sample output of "select * from cdr" http://pastebin.com/HeM45Dv7 |
| 22:11.55 | rue_work | does the sagnoma wanpipe replace or compliment the dahdi package? |
| 22:14.51 | rue_work | the asterisk dahdi package is hardware drivers for digium cards or its a protocol laye? |
| 22:21.45 | WiretapWork | rue_work, both AFAIK |
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| 22:31.59 | el3slave | who are you guys using for SIP terminationa and origination in USA? |
| 22:34.19 | rue_work | was a member of the name 'pvt' 'recently' removed from the dahdi-span struct? |
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| 22:49.35 | russellb | infobot: tell el3slave about itsplist-us |
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| 23:00.16 | rue_work | the |
| 23:00.24 | nix8n82 | infobot: tell nix8n82 about itsplist-us |
| 23:00.31 | rue_work | the wanpipe-3.5.20 driver compiled ok |
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| 23:11.23 | el3slave | russellb, thanks sir |
| 23:11.44 | russellb | np |
| 23:20.20 | WIMPy | heffer: No, the issue I reported is not related. |
| 23:21.11 | WIMPy | From the link you posted it seems to be a bug in libpri (or dahdi?). |
| 23:22.32 | WIMPy | The issue you pointed out only affects calls that are not torn down correctly because of L1 failure. |
| 23:28.15 | rue_work | OK ! asterisk compiled and isntalled on debian with 2.6.32-5 kernel thats debian stock! |
| 23:28.29 | rue_work | ls |
| 23:28.32 | rue_work | er |
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| 23:45.16 | nny | for all my fllow polycommunists out there, I am working on enabling buddy watch and persistent volume in 3.3.1's config revisions. I have http://pastebin.com/r7k4nEqF (sip.cfg snippet) Can anyone check and see if this is what the admin guide intends as far as format and placement? |
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