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| 00:35.18 | AgroTemp | If I receive a sound input through AGI, is there an easy way to get the tone frequency? |
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| 01:06.30 | metabsd | Anyone have probleme with meet-me + asterisk 1.8.11 + freepbx 2.10 ? |
| 01:06.48 | metabsd | when i page someone sometime the channel dont close |
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| 01:19.17 | metabsd | anyone can help me ? |
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| 01:27.02 | voipnation | hola |
| 01:28.09 | voipnation | Does anyone have any idea the comparison of FFA vs Spandsp in terms of reliability goes? |
| 01:30.44 | voipnation | I am running Asterisk 1.8.4 on a Dual, Quad Core Xeon 3.0 if that helps. I just need to find a setup that will be as compatible as possible with most fax providers. |
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| 07:16.43 | jww_ | Hello everyone. |
| 07:17.07 | jww_ | if anybody have any experience with using a2billing with asterisk, I'll very glad to hear about it. |
| 07:17.28 | jww_ | I'm tryin to setup it, but there is few documentation, and I'm a noob. |
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| 08:49.32 | BeeBuu | hello,all. |
| 08:55.35 | BeeBuu | is there a limit on ami login? I want to write a programm that make multi connect to AMI in a same time. |
| 09:02.19 | kaldemar | BeeBuu: using the same credentials? |
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| 09:04.07 | BeeBuu | kaldemar: same user |
| 09:04.31 | BeeBuu | and same password |
| 09:04.38 | BeeBuu | :P |
| 09:05.22 | BeeBuu | socket multi-thread |
| 09:05.29 | kaldemar | BeeBuu: just make sure you don't have allowmultiplelogin=no in manager.conf. |
| 09:06.31 | BeeBuu | kaldemar: i make sure there isn't |
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| 09:09.13 | hariom | I am using no-ip service to connect a sip softphone to the asterisk running on my system. Can anybody guide me how to achieve that? |
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| 09:12.26 | BeeBuu | kaldemar: is there any limit on it? |
| 09:12.43 | BeeBuu | or anyone tell me please? |
| 09:14.59 | hariom | what entries are required for allowing my asterisk system to accept calls from remote host name? |
| 09:15.30 | bulkorok | hi... I have segfaults. I have coredumps and gdb... can anyone help to "read" the output, or should I just put it in issues.asterisk.org? |
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| 09:17.23 | kaldemar | hariom: what calls? |
| 09:17.35 | kaldemar | BeeBuu: doubt it. test and see. |
| 09:18.42 | hariom | kaldemar: I am testing to call from a sip softphone installed on Windows machine to a asterisk machine installed on Linux. I am using no-ip.org so that I can ask my friends to try the sample application I have. What settings I need to do in sip.conf? |
| 09:19.10 | hariom | kaldemar: friends staying in different cities. |
| 09:21.06 | kaldemar | hariom: depends on your network setup and what you want to offer to your friends. |
| 09:21.35 | hariom | kaldemar: basically I want to allow my fiends to be able to dial into my asterisk system and listen the hello-world message. |
| 09:22.35 | kaldemar | ~book |
| 09:22.35 | 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 |
| 09:22.44 | BeeBuu | kaldemar: no any document about that? |
| 09:23.00 | kaldemar | BeeBuu: about what? |
| 09:23.57 | kaldemar | BeeBuu: if the connections were limited by asterisk, it would be the first unconfigurable software limit i've ever come across with asterisk. |
| 09:24.18 | hariom | kaldemar: How to setup? Can you suggest the chapter number ? |
| 09:25.44 | kaldemar | hariom: http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-DeviceConfig.html |
| 09:26.24 | kaldemar | if nat is involved on asterisk's or the client's end, see this: |
| 09:26.29 | kaldemar | ~sipnat |
| 09:26.29 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, canreinvite, externhost or externip, and localnet. |
| 09:26.39 | BeeBuu | kaldemar: thanks. maybe i test that myself |
| 09:43.05 | hariom | Does anybody has experience of using no-ip or dydns like service to redirect softphone incoming request to the asterisk system? |
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| 09:53.22 | kaldemar | BeeBuu: just out of curiosity, my test asterisk started to choke on too many open files around 9XX manager connections. |
| 09:53.52 | Chainsaw | kaldemar: Sounds like the 1024 FD limit to me. |
| 09:54.05 | kaldemar | Chainsaw: and that it was. |
| 09:54.27 | Chainsaw | kaldemar: I think I have mine set to 4096. |
| 09:54.41 | kaldemar | the point being that "many" conncetions are not an issue. :) |
| 10:01.44 | Chainsaw | kaldemar: Oh don't mind me, I only fell into the conversation halfway. |
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| 11:07.56 | [sr] | WIMPy: howdy |
| 11:08.15 | WIMPy | Hi [sr] |
| 11:08.22 | [sr] | were u on vacation? :p |
| 11:08.39 | WIMPy | Last month. |
| 11:08.48 | WIMPy | Special vacation. |
| 11:09.19 | [sr] | i see |
| 11:09.32 | [sr] | home it was something good at least |
| 11:10.22 | WIMPy | Somehow. I found out some things and lost 5kg. |
| 11:11.14 | [sr] | well, dont need to say anything more, i may guess what you are trying to say |
| 11:12.22 | WIMPy | So how's things going in your area? |
| 11:12.57 | [sr] | portugal? not so good... |
| 11:13.27 | [sr] | seems to be a sinking boat |
| 11:14.31 | WIMPy | So just like everywhere, I guess. |
| 11:15.03 | WIMPy | Do you have a Pirate Party in PT? |
| 11:15.18 | [sr] | what is that? |
| 11:15.31 | [sr] | not familiar with that name |
| 11:15.55 | WIMPy | http://en.wikipedia.org/wiki/Pirate_party |
| 11:17.03 | [sr] | oh i get it |
| 11:17.18 | [sr] | we do have something like that, but it's almost unspoken |
| 11:20.27 | WIMPy | PP is becoming very popular in DE. 9% and 7% in the last land elections and predictions of 13% for the nation wide elections next year. |
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| 11:41.22 | [sr] | WIMPy: in here i guess won't be even 1% |
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| 11:50.07 | jww_ | good afternoon #asterisk. |
| 11:50.20 | jww_ | does anybody every heard about a2billing, exept from me ? :) |
| 11:52.15 | WIMPy | Yes, but that's the whole story already. |
| 11:53.27 | jww_ | doh :\ |
| 11:54.32 | WIMPy | But you know, there's that thing about |
| 11:54.37 | WIMPy | ~ask |
| 11:54.37 | 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. |
| 11:55.12 | jww_ | I tried to be a lot more precise when I asked first. but it was days ago. |
| 11:55.30 | jww_ | so I became more generic days on days ;) |
| 11:55.46 | jww_ | still I'll rephrase. |
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| 12:00.02 | jww_ | I have troubles with asterisk 1.8 and a2billing 1.9.4 . when I add users in a2billing and generate the additional sip conf file, this one is empty and when a call come it's dropped. |
| 12:01.14 | [TK]D-Fender | jww_, a2billing isn't supported here. |
| 12:02.00 | jww_ | do you know where I could ask questions about it ? |
| 12:02.22 | [TK]D-Fender | jww_, Go check their project page for a list of other resources; mailing lists, forums, etc |
| 12:02.51 | jww_ | thanks for the advise. |
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| 12:33.24 | aberrios | I have a strange issue on a PRI since updating dahdi to 2.6 and * to 1.8.11. I keep getting this error: http://pastebin.com/C0g3W6ni but no alarms on the cards... any ideas? |
| 12:34.43 | WIMPy | You and your peer don't seem to talk the same language. |
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| 13:09.17 | aberrios | WIMPy: langue=signalling type? |
| 13:09.24 | aberrios | *language |
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| 13:13.07 | aberrios | this has only started since upgrading to 1.8 and dahdi 2.6 |
| 14:01.31 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 14:01.31 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.3.0 (2012/03/29), 1.8.11.0 (2012/03/29), dahdi-linux 2.6.0 (2012/01/04), dahdi-tools 2.6.0 (2012/01/04), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 14:03.55 | like_a_horse | bulkorok, i see in his faxout context he has "send" and "h". Which would apply when the call file is read? Is it both? |
| 14:04.51 | bulkorok | like_a_horse: when the fax is sent, the call will be hungup. The the h-extension can be used to get the FAXOPT data |
| 14:05.12 | bulkorok | send is for sending, h for getting the status after sending... |
| 14:05.38 | like_a_horse | ok bril - i'll tinker a bit |
| 14:05.39 | like_a_horse | thanks |
| 14:06.37 | asilva | Hey, Little Help. After upgrading my asterisk from 1.8.5.0 to 1.8.11.0 when a call is picked up(using builtin asterisk feature) the phone where the call got picked up is not showing the missing call anymore, any thoughts ? |
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| 14:23.14 | asilva | anyone ? |
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| 14:30.51 | asilva | hello, anyone alive here able to help me out ? |
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| 14:31.31 | blitzrage | asilva: revert back and see if the change is caused by asterisk -- if so, then start stepping through and try to narrow down which commit broke it, and then report it |
| 14:31.53 | aberrios | how do i change which span is master for timing in dahdi? |
| 14:32.59 | WIMPy | asilva: Maybe you should reprhrase your statement. I don't understand the isse. |
| 14:34.24 | WIMPy | aberrios: The 2nd parameter of span=. |
| 14:35.02 | asilva | WIMPy: Example: A call to B and C pickup de call from builtin asterisk feature, in B should be a message saying missing call N and that is no longer hapenning after upgrade from 1.8.5.0 to 1.8.11.0 |
| 14:35.07 | *** join/#asterisk _Corey_ (~chatzilla@mail1.infradapt.com) |
| 14:35.25 | blitzrage | asilva: that is a function of the phone to determine whether to display a missed call or not |
| 14:35.40 | WIMPy | asilva: Why should b show a missed call when it wasn't missed? |
| 14:35.44 | blitzrage | and that |
| 14:35.53 | blitzrage | the functionality as it is happening now is what I would expect |
| 14:35.54 | asilva | blitzrage: i'm doing the version checking as you suggested. nothing on the phone has changed. |
| 14:36.12 | blitzrage | honestly the way it worked before in 1.8.5.0 sounds like a bug |
| 14:36.13 | asilva | WIMPy: before the upgrade i was showing. |
| 14:36.20 | blitzrage | what you're experiencing now sounds liek the correct behaviour |
| 14:36.38 | asilva | I thought so too, just trying to figure it out what to say to users here!!! |
| 14:36.40 | WIMPy | asilva: Sounds like a bug having been fixed to me. |
| 14:36.49 | blitzrage | WIMPy: same |
| 14:36.57 | blitzrage | asilva: tell them it was broken before and now it is fixed |
| 14:37.15 | blitzrage | a picked up call is not missed |
| 14:37.17 | blitzrage | it was answered |
| 14:37.34 | asilva | blitzrage: yeah.. just going to determine which version whas changed and which files got changed so i can understand better what happened before and what is happening now! |
| 14:37.49 | blitzrage | look for changes to either chan_sip or features.c |
| 14:37.56 | blitzrage | if you want to narrow it down |
| 14:38.07 | blitzrage | look at the commit logs to see if you see anything that makes sense |
| 14:38.13 | blitzrage | then try before and after the commit |
| 14:38.36 | asilva | i did.. could find the exaclty information .. let me narrow down the versions so i can look again! |
| 14:38.45 | WIMPy | IIRC there's an option to Dial() to always tell the call was missed. But I don;t know if that would work with features. |
| 14:42.34 | biberao | instead of a fxs or fxo cant i use a normal modem? |
| 14:43.00 | WIMPy | biberao: You don't want to. |
| 14:43.10 | RZero | Hi, I have set up that goes like this Incoming call -> Asterisk -> sip switch -> Asterisk-Dahdi -> ISDN , Im getting DTMF issue, I can see DTMF going all the way through and looks like it being sent across the dahdi card, but the external end does not receive the DTMF, if I register myself on the asterisk and make the same call, DTMF works, I have tried dtmfmode=auto , rfc2833 and INFO but none of these work. |
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| 14:45.00 | biberao | why WIMPy ? |
| 14:45.28 | aberrios | WIMPy: If timing is off would this cause the issue i paste binned? |
| 14:45.41 | aberrios | WIMPy: when i say off,... i mean its not right, i dont mean turned off |
| 14:46.02 | WIMPy | aberrios: No. Bad timing wil cause more trouble. |
| 14:46.15 | aberrios | WIMPy: like crackling lines? |
| 14:46.23 | *** join/#asterisk bchia (~Adium@nat/digium/x-bzhfcvxttawtchlf) |
| 14:46.28 | aberrios | WIMPy: because that is also happening |
| 14:46.47 | WIMPy | aberrios: Have you tried to find out when it's happening? Should be possible to find out what feature is causing it. |
| 14:47.07 | WIMPy | In the worst case you would get audio issues as well, yes. |
| 14:47.10 | aberrios | WIMPy: it seems to be very intermittent on the E1 lines... |
| 14:47.45 | WIMPy | If you have bad timing you should see HDLC abort messages. |
| 14:48.05 | aberrios | Yes i do |
| 14:48.17 | aberrios | Could i set span=1,0,0? |
| 14:48.22 | aberrios | in system.conf |
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| 14:49.05 | aberrios | I ahve an anoloque card in the machine and previously it seemed this was MASTER but now Span 2 on the PRI card is MASTER.... |
| 14:49.07 | WIMPy | That interface is connected to your telco? Is it only one interface? |
| 14:49.26 | aberrios | theres two E1 interfaces to two different telcos |
| 14:49.38 | aberrios | Theres also an anologue card installed |
| 14:50.07 | WIMPy | As far as I understand, the Digium cards don't support that configuration. |
| 14:50.50 | aberrios | WIMPy: you mean two different telcos on one card? |
| 14:51.12 | WIMPy | To have two independant timing sources, you need two cards or maybe other cards do support that configuration, but I'm not sure on that. |
| 14:51.28 | WIMPy | yes |
| 14:51.59 | aberrios | I see. Could I set the Analogue card to do all the timing, I believe this is how we were previously before the dahdi upgrade |
| 14:52.16 | WIMPy | No. |
| 14:52.31 | WIMPy | You always have to use the telcos timing. |
| 14:57.33 | aberrios | WIMPy: its strange it all worked fine until the dahdi upgrade, even with two telcos |
| 15:01.33 | WIMPy | You might want to ask Digium. |
| 15:06.09 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 15:38.35 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 15:38.35 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.3.0 (2012/03/29), 1.8.11.0 (2012/03/29), dahdi-linux 2.6.0 (2012/01/04), dahdi-tools 2.6.0 (2012/01/04), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
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| 15:42.40 | aberrios | WIMPy, [TK]D-Fender: we've downgraded dahdi to 2.5.0.2 and the crackling issues have gone it seems. still getting the paste binned errors though |
| 15:45.03 | asilva | blitzrage: from 1.8.8 the "problem" occurs ahahah.. trying to figure it out what change made that!! |
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| 15:45.43 | WIMPy | aberrios: yes, that's libpri or chan_dahdi. |
| 15:45.52 | WIMPy | What span configuration are you using now? |
| 15:46.54 | asilva | blitzrage: could be this - 2011-09-07 13:26 +0000 [r334682] Stefan Schmidt <sst@sil.at> |
| 15:46.54 | asilva | * main/features.c: Adding the Feature to sent a Reason Header in a |
| 15:46.55 | asilva | <PROTECTED> |
| 15:46.55 | asilva | <PROTECTED> |
| 15:48.38 | aberrios | 1,1,0 and 2,2,0 |
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| 15:51.47 | WIMPy | That would require both telcos tobe in sync. |
| 15:52.11 | WIMPy | I've also done that once and it worked, but you need some luck. |
| 15:52.21 | WIMPy | And you never know how long it lasts. |
| 15:53.32 | aberrios | WIMPy: WHy would 2.6.0 have an issue but 2.5 no? |
| 15:58.04 | WIMPy | It's probably something else that goes wrong. |
| 15:58.33 | WIMPy | I had completely garbled audio with 2.6.0. |
| 15:59.04 | WIMPy | But interestingly only after connect. Tones and announcements were clear. |
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| 16:32.28 | aberrios | WIMPy: how do i tell chan_dahdi to shut up with these errors, dont want it filling up the log file while its fixed. |
| 16:33.16 | WIMPy | Find out why they are there. |
| 16:33.26 | WIMPy | It's probably somethin that can be diasbled. |
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| 16:33.57 | aberrios | WIMPy: well yeah XD but at the moment everyone's knackered and everything sounds okay so its being left until we've had a break... |
| 16:35.44 | aberrios | WIMPy: suppose i could just stop errors going to file in logger.conf |
| 16:36.19 | WIMPy | Just enable pri debug and see what's going on. |
| 16:36.50 | aberrios | WIMPy: its late here, its being left until after the weekend. Calls are fine. |
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| 16:38.18 | sruffell | aberrios: Did you always get crackling audio on all your calls with 2.6.0? |
| 16:38.22 | sruffell | or only on some calls? |
| 16:38.40 | aberrios | sruffell: some, about 90% |
| 16:39.04 | WIMPy | sruffell: Is that still open? I didn;t really look in to it. |
| 16:39.24 | aberrios | sruffell: and only our side. Customer heard no crackling only our agents |
| 16:39.52 | sruffell | aberrios: And I take it was E1 spans on a dual or quad span card? |
| 16:40.06 | aberrios | sruffell: correct. 2 E1 spans on a dual card |
| 16:40.13 | aberrios | sruffell: digum card too |
| 16:40.34 | sruffell | aberrios: I bet you'll need 2.6.1. It was tagged yesterday, and I was just waiting for a few signatures before moving it to downloads. |
| 16:40.57 | Qwell | can solve that |
| 16:41.35 | sruffell | aberrios: Is it possible you were hitting https://issues.asterisk.org/jira/browse/DAHLIN-275 ? |
| 16:42.08 | sruffell | Qwell: thanks. |
| 16:43.04 | sruffell | WIMPy: do you have any B410P cards that you can test with BRI spans from your telco? |
| 16:43.14 | WIMPy | I found it interesting that it only happened after connect. That way I missed the isse at first. |
| 16:43.28 | sruffell | nods |
| 16:43.33 | WIMPy | sruffell: No, only OctoBRIs. |
| 16:43.35 | sruffell | yeah..it's when the ec is enabled on the bchannel. |
| 16:43.37 | Qwell | aberrios: I just typed my password 14 times in a row, just for you. |
| 16:44.04 | sruffell | WIMPy: ok, thanks. I was going to see if you could test something out for me but alas... |
| 16:45.02 | WIMPy | Only PRI hardware from Digium. |
| 16:45.45 | WIMPy | And after I had that experience with that old HWEC module, I found out that I find the HWEC on the b410p rather useless. |
| 16:45.52 | WIMPy | Why does it only do 64ms? |
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| 16:46.17 | sruffell | WIMPy: I don't know…and I hear what you're saying.... |
| 16:46.42 | sruffell | WIMPy: although, the Hx8 can take a better hwec (just sayin...) |
| 16:46.57 | coppice | the HWEC on the B410P is the only one that doesn't generate complaints |
| 16:47.40 | sruffell | depends on your line conditions. |
| 16:47.59 | coppice | BRIs don't normally encounter long echoes |
| 16:48.26 | WIMPy | No, but the person you're talking to might use VOIP and exceed the 64ms. |
| 16:48.34 | coppice | the EC on the BRI card is a different make, and its a lot less troublesome than the octasic ones |
| 16:48.45 | WIMPy | Ecen the 128ms ones fail regularly :-( |
| 16:49.00 | WIMPy | s/Ece/eve/ |
| 16:49.03 | aberrios | sruffell: it could be that issue, he describes a loud noise where we would describe it as crackling,, like the static mentioned in the thread |
| 16:49.09 | sruffell | But I haven't heard any complains against the VPMADT032 since 1.25 firmware (DAHD-Linux 2.4.0) nor the VPMOCT032 released in DAHDI-Linux 2.5.0 |
| 16:49.16 | coppice | the far end using VoIP is irrelevant. If they don't keep their echo off the PSTN, there is generally little you can do about it |
| 16:49.53 | aberrios | Qwell: whoop |
| 16:50.10 | coppice | the octasic ecs, both on digium and sangoma cards, cause a lot of people problems, like screwing up DTMF that's passing through the card to an IVR |
| 16:50.19 | WIMPy | If you touself use non-local VOIP phones, you're also most likely exceeding 54ms. |
| 16:50.24 | WIMPy | 64 |
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| 16:50.50 | sruffell | aberrios: ok..2.6.1 will be on the downloads later today……so when you pick that up again, feel free to update (and sorry about that…it was my mistake…I do appreciate when people update to the latest and feel bad when I make trouble for them) |
| 16:50.53 | WIMPy | Might be a point for SWEC. It's easier to replace. |
| 16:50.57 | WIMPy | And a lot cheaper, off course. |
| 16:50.58 | coppice | domestic lines do not encounter long echoes. you needs to be treated as a telco to see a long echo |
| 16:52.09 | aberrios | sruffell: No worries, we spend quite a bit of time trying to get it working assuming it was some other hardware that was in the call route. But eventually just decided to downgrade again,,, in retrospect probably should have downgraded straight away lol. |
| 16:52.43 | WIMPy | In theory it shouldn't matter as conversation it disturbed by the delay as well, but unfortunately it's todays reality we have to live with :-( |
| 16:53.57 | WIMPy | sruffell: On aberrios' topic: Can you give a definite asnwer if it's possible to have multiple clock sources on one card? |
| 16:54.40 | sruffell | WIMPy: no, you cannot |
| 16:55.24 | aberrios | WIMPy: well thats that then =D |
| 16:55.26 | sruffell | If you need to recover clock from two different spans (i.e, you have two providers who aren't using a synched clock) you will have timing problems on the span that isn't recovering the clock for the entire card. |
| 16:55.26 | WIMPy | Ok, so now I know for sure instead of prety sure. |
| 16:56.00 | sruffell | aberrios: are you in Brazil? |
| 16:56.02 | WIMPy | Yes, unfortunately I'd call that the standard situation. |
| 16:56.09 | aberrios | It seems to work with Virgin Media provided E1 and BT provided E1,,,so they must be synced..... no sruffell ,, UK |
| 16:56.33 | sruffell | most providers *should* be syncing their clocks from their master….it's only when they don't do that where I've seen problems. |
| 16:57.37 | aberrios | well folks I gotta scram. Thanks for all your input and help. Just have to solve this chan_dahdi Error that keeps popping up sometime. |
| 16:59.05 | coppice | aberrios: even if the operators don't sync properly, their own timing should be derived from an atomic clock |
| 16:59.32 | aberrios | if anyone wants to look at it, (for whatever crazy reason) I paste binned it here: http://pastebin.com/C0g3W6ni |
| 16:59.54 | aberrios | and thats the BT E1 span |
| 17:00.17 | WIMPy | aberrios: That's not saying much without a pri debug. |
| 17:00.32 | aberrios | WIMPy: righto, well I'll grab some debug next week. |
| 17:00.42 | aberrios | k. Im off... had no lunch :( and its tea time! |
| 17:00.45 | aberrios | byeeeeee |
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| 17:08.48 | WIMPy | sruffell: Are you into the documentation? |
| 17:08.55 | WIMPy | Or manual to be precise. |
| 17:09.30 | sruffell | not really…but I have it on my todo list to flesh out a DAHDI section on wiki.asterisk.org. |
| 17:10.02 | WIMPy | That would be a very good idea. |
| 17:10.31 | WIMPy | Just came across a little misnomer in the Hx8 manual. |
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| 17:11.14 | sruffell | if you email it to me, I'll find out who the right person to forward it to is, and then do so... |
| 17:12.00 | sruffell | or if it's small..you can just tell me here. : |
| 17:12.01 | sruffell | :) |
| 17:12.09 | sruffell | and save opening up your mail client. |
| 17:12.36 | WIMPy | Actually I'm not sure what's going on there. I think I better finish reading first. |
| 17:12.58 | WIMPy | But I read 8 ports an see 7 on the picture. |
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| 17:15.55 | WIMPy | Ok, the ports are called RJ-11, but they must have 4 conductors obviousely. So are they 4P4C or 6P4C? |
| 17:17.52 | malcolmd | 6P4C |
| 17:18.09 | Qwell | WIMPy: The 8th port is hidden under the metal |
| 17:18.17 | Qwell | I think. |
| 17:18.19 | WIMPy | But at least the chapter on termination is correct, which wasn;t the case in the b410p manual. |
| 17:18.26 | malcolmd | well..that's not entirely true...there are 6 positions and there are 6 contacts in the jack itself, but only 4 of those contacts are actually wired to anything on the card |
| 17:19.38 | WIMPy | Oh. Page 59 talks about a "delta channel". That should surely be "data channel". |
| 17:19.54 | malcolmd | delta channel is actually fine terminology |
| 17:20.17 | WIMPy | Delta between what? |
| 17:21.36 | sruffell | ...or is it phonetic for 'D'? as opposed to greek delta? |
| 17:22.19 | malcolmd | in that case, the bearer channels would be bravo channels, but they're not, they're bearers |
| 17:22.31 | sruffell | worth a try. :) |
| 17:22.36 | WIMPy | I have seen "delat" befor, but not in any place I'd trust. |
| 17:22.53 | WIMPy | bad typing :-( |
| 17:23.17 | WIMPy | But the echo channel would fit that idea :-) |
| 17:23.25 | Qwell | malcolmd: What do they bear? |
| 17:23.35 | malcolmd | Qwell: burdens :( |
| 17:23.36 | WIMPy | Payload |
| 17:26.17 | WIMPy | Still no power supply :-( |
| 17:26.36 | malcolmd | nope |
| 17:42.25 | adeel|work | if i have a realtime peer, can i override a single SIP value via a flat file? |
| 17:43.02 | [TK]D-Fender | adeel|work, no |
| 17:43.24 | adeel|work | hmmm...interesting |
| 17:55.42 | seanbright | would anyone know the per-interaction or per-minute cost for lumenvox's usage-based licensing? |
| 17:57.24 | paulc | I don't, but I'd be interested in the answer too.. |
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| 18:15.18 | _Corey_ | seanbright: Good luck, Lumenvox sales has been a blackhole for a while |
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| 18:18.50 | pabelanger | _Corey_: fun thing you say that, I had a friend ask me about Lumenvox the other day, never used them but told him they sound like an okay company |
| 18:18.54 | pabelanger | I might have to update him |
| 18:19.22 | _Corey_ | Well, they've had an ASR product and excellent Asterisk support for years |
| 18:19.51 | _Corey_ | Sometime last fall, they fired a bunch of people and I got this weird e-mail from their CEO and Chairman saying he was my new account rep |
| 18:20.05 | pabelanger | OIC |
| 18:20.06 | _Corey_ | I've been waiting on pricing for something now for like 3 weeks |
| 18:20.13 | _Corey_ | so, it's a mystery |
| 18:20.21 | pabelanger | ya, that doesn't sound good |
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| 18:26.30 | jaytee | God I hate AT&T customer service. It's such an oxymoron. |
| 18:29.21 | keiths_ | I hope AT&T is better to deal with then the Canadian providers :) |
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| 18:32.57 | FLeiXiuS | Is it possible to originate a call through AMI using an extension that starts with 0? |
| 18:34.12 | FLeiXiuS | It appears to be getting ripped off every time I try. |
| 18:34.29 | WIMPy | Why should that matter? |
| 18:35.08 | FLeiXiuS | I didnt think it did. |
| 18:35.24 | WIMPy | I'm certain it doesn't. |
| 18:36.22 | seanbright | _Corey_: yeah... just got off the phone... pretty awful |
| 18:37.51 | seanbright | they claim to have usage-based and burst licensing, but he can't tell me how much those cost?? |
| 18:37.54 | seanbright | oh well |
| 18:38.40 | blitzrage | _Corey_: ya someone I know who has worked there for years recently left(?) and started another voice related business |
| 18:38.55 | *** topic/#asterisk by sruffell -> #asterisk he Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.3.0 (2012/03/29), 1.8.11.0 (2012/03/29), DAHDI-linux 2.6.1 (2012/04/20), DAHDI-tools 2.6.1 (2012/04/20), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 18:40.58 | _Corey_ | blitzrage: I think something financial (and significant) must have happened... I've heard of other turnover like that after the firings in the fall |
| 18:41.08 | g_r_eek | any freelancer in here that can code a 4 question survey dialer? |
| 18:41.22 | blitzrage | _Corey_: ya not sure what happened... but seems like a major malfunction :) |
| 18:41.24 | _Corey_ | eh, we'll see... too bad there aren't a lot of other options on the ASR side |
| 18:41.59 | blitzrage | indeed |
| 18:42.08 | blitzrage | unless you want to screw with PocketSphinx |
| 18:42.11 | _Corey_ | g_r_eek: Send me a PM, we do that sort of thing |
| 18:42.22 | _Corey_ | blitzrage: gah, don't get me going on Sphinx |
| 18:43.05 | blitzrage | _Corey_: I promise nothing |
| 18:43.22 | _Corey_ | lol |
| 18:43.35 | _Corey_ | Maybe it's improved... it's been a few years since I messed with it |
| 18:43.49 | blitzrage | ya I tried for about an hour and threw it away because I didn't have anyone who needed it |
| 18:45.48 | _Corey_ | It's hard to sell speech recognition... I can't give it away sometimes |
| 18:46.07 | coppice | TTS and ASR companies have never shown much stability |
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| 18:46.50 | _Corey_ | Nuance seems to be doing just fine |
| 18:47.28 | coppice | but nuance is the amalgamation of many ASR and TTS companies that were sold to nuance for fire sale prices |
| 18:48.24 | _Corey_ | true enough... exception that proves the rule i guess |
| 18:48.58 | coppice | lumenvox were very uncooperative for a new player in the market |
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| 19:36.36 | aviationprinc3ss | Hey guys, would anyone know how Asterisk selects a channel to dial out on? |
| 19:37.03 | WIMPy | Yes, your dialplan. |
| 19:37.24 | WIMPy | The explanation is at the top of the sample extensons.conf. |
| 19:38.40 | aviationprinc3ss | Well, I've defined a group in extensions.conf |
| 19:39.08 | aviationprinc3ss | And I've got 4 licensed channels for faxing. |
| 19:39.13 | aviationprinc3ss | It always chooses to dial out on the highest numbered port |
| 19:39.36 | aviationprinc3ss | And doesn't seem to make much use of the other three. |
| 19:39.38 | WIMPy | Port of what? |
| 19:39.58 | WIMPy | And why is that bad? |
| 19:40.10 | aviationprinc3ss | I'm using 4-port Analog |
| 19:40.36 | WIMPy | See the options g, G, r, and R. |
| 19:41.00 | WIMPy | But it makes sense to have it do the opposite of your telco to avoid collisions. |
| 19:43.31 | aviationprinc3ss | Ah. I forgot about those options. That explains why it's choosing the highest-numbered channel. |
| 19:44.30 | aviationprinc3ss | I just don't understand why when I dump say like 100 call files, 90% of the time is only using that one channel. |
| 19:44.51 | aviationprinc3ss | I was hoping it would use all 4 channel simultaneously |
| 19:45.04 | aviationprinc3ss | Not quite sure what I'm doing wrong =/ |
| 19:49.53 | bchia | you're using the group in the call file and not the channel? |
| 19:50.51 | bchia | you could also check chan_dahdi.conf to make sure the right group is being applied to all the channels (with the syntax being 'sticky' from the top unless it's reset) |
| 19:52.01 | WIMPy | It will only use the first channels if you have 4 simultaneous calls. |
| 19:52.20 | WIMPy | How do you create the calls? |
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| 19:54.57 | raubvogel | On op-panel, where are the passwords for voicemail defined? Are they the ones in /etc/asterisk/voicemail.conf? |
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| 19:58.15 | WIMPy | I don't know what op-panel is, but that's where the VM passwords are stored, yes. |
| 20:07.20 | raubvogel | WIMPy: that would be something called flash operator panel (switchboard for asterisk). Can't find a channel for it so I am trying here |
| 20:10.42 | WIMPy | Ah, that one. Seemt to be referred to as FOP usually. |
| 20:10.46 | aviationprinc3ss | bchia: Yes, I'm using G2 in the call file. |
| 20:11.12 | aviationprinc3ss | In chan_dahdi.conf, I set channel to 1-4 |
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| 20:11.53 | aviationprinc3ss | WIMPy: What do you mean by the first channels? |
| 20:12.04 | aviationprinc3ss | I would expect it to use 4,3,2, then 1 |
| 20:12.12 | aviationprinc3ss | Which is actually...what it appears to be doing |
| 20:13.00 | aviationprinc3ss | But then I don't see the point in the other two channels, if they're not being used. |
| 20:13.24 | raubvogel | WIMPy: Aha. So, I know what the vm password is, but when I enter what is defined in voicemail.conf, I just get the login dialog box again. No error messages or anything. In fact if /var/log/op-panel/error.log, it reports nothing about my failed attempts. |
| 20:14.06 | WIMPy | aviationprinc3ss: It will always use the highest available channel, i.e. channel 1 would only be used if channels 2, 3 and 4 are all busy. |
| 20:14.31 | raubvogel | Nothing in the apache error log, so I do not know what FOP is doing and why it is not letting me in. |
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| 20:14.39 | WIMPy | raubvogel: I'm not familiar with FOP. |
| 20:15.08 | raubvogel | WIMPy: I not either. Just hoping someone here is and can shed some light. :) |
| 20:18.41 | aviationprinc3ss | WIMPy: That makes sense. |
| 20:19.29 | aviationprinc3ss | But if I dump 4 call files, there are only 4 channels, shouldn't it try to use all 4? |
| 20:19.34 | WIMPy | aviationprinc3ss: If you want to use them one-by-one, use r/R instead of g/G, but as explained earlier, you probably don't want to. |
| 20:19.39 | WIMPy | yes |
| 20:21.31 | aviationprinc3ss | Hmmm |
| 20:22.19 | aviationprinc3ss | Ok. I guess my issue with it, is that it's doing each job one roughly one minute later. |
| 20:22.35 | aviationprinc3ss | (As opposed to simultaneous) |
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| 20:22.46 | WIMPy | So it just doesn't have that much to do. |
| 20:23.52 | raubvogel | On an unrelated note (diff setup altogether), what would be the best way to move from trixbox to asterisknow? Assuming you have them installed on different machines, can you export/move the configs and data (voicemail) between both? |
| 20:24.04 | Qwell | raubvogel: Reinstall. |
| 20:24.12 | Qwell | trixbox is not compatible with anything, including itself. |
| 20:24.32 | raubvogel | Qwell: nice to know. Any way I can at least save the voicemails? |
| 20:24.46 | raubvogel | Everything else I do not mind doing |
| 20:25.09 | raubvogel | Qwell: and that is the reason I want to leave trixbox |
| 20:30.04 | Qwell | raubvogel: You can basically copy /var/spool/asterisk/voicemail/ |
| 20:30.28 | raubvogel | Qwell: Nice |
| 20:30.30 | raubvogel | Thanks! |
| 20:30.42 | aviationprinc3ss | WIMPy: Thanks, I think I'll play around with the round-robin approach for now :] |
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| 20:44.13 | nicola_pav | hello. I am facing random call hangups |
| 20:44.21 | nicola_pav | i have a digium pri card |
| 20:44.37 | nicola_pav | calls get connected well but randomly, the disconnect |
| 20:44.45 | nicola_pav | i have enabled pri debug span |
| 20:45.01 | nicola_pav | but I can't read it well |
| 20:45.18 | WIMPy | pastebin it. |
| 20:48.15 | nicola_pav | here is the call flow: http://pastebin.com/UFZbdRBc |
| 20:48.59 | blitzrage | _Corey_: back to speech recognition stuff, I've been using the speech-to-text feature on my android phone a lot lately |
| 20:50.34 | _Corey_ | blitzrage: I presume that'd be using Google's engine? |
| 20:50.44 | WIMPy | nicola_pav: Nothing wrong there. The call seems to be ended by the SIP side. |
| 20:50.45 | blitzrage | _Corey_: indeed |
| 20:50.49 | blitzrage | or so I assume |
| 20:50.58 | blitzrage | it's built into android so I imagine so |
| 20:51.23 | _Corey_ | I'm generally happy with google's solution though the accuracy is sometimes laughably bad |
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| 20:51.48 | _Corey_ | It's on my list to do some testing with Twilio's voice to text thing |
| 20:52.16 | nicola_pav | WIMPy: thank u. anything else I can do to debug it deeper? |
| 20:52.37 | WIMPy | sip debug. |
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| 21:01.32 | nicola_pav | WIMPy: is this normal: q931_disconnect: Call xxxxx enters state 11 (Disconnect Request)? |
| 21:02.16 | WIMPy | that's while it's disconnecting. |
| 21:03.33 | nicola_pav | so next step is sip debug or even wireshark? |
| 21:04.54 | WIMPy | sip debug should be good enough. |
| 21:05.18 | WIMPy | Or maybe you can even turn up more debug. |
| 21:05.47 | nicola_pav | like what? |
| 21:06.04 | WIMPy | core set debug 9 |
| 21:07.18 | nicola_pav | ok, thanks for the tips, i will do it and see |
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| 21:10.09 | seanbright | _Corey_: any idea what licensing for nuance's speech server looks like? |
| 21:10.35 | _Corey_ | seanbright: Expensive is about all I could tell you... :) |
| 21:10.45 | seanbright | higher than lumenvox? |
| 21:10.57 | _Corey_ | certainly |
| 21:11.06 | seanbright | captive audience i guess |
| 21:12.21 | _Corey_ | Their customers for their speech platform are Fortune 100 companies, so they price things as such |
| 21:12.37 | seanbright | i want bank of america's VR |
| 21:12.46 | seanbright | that's the best i have dealt with as a user |
| 21:12.47 | _Corey_ | It's probably nuance |
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| 23:45.04 | pabelanger | flowroute question, how are peeps doing multiple registrations to flowroute? I can't figure out how to setup more then 1 SIP registration |
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