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| 05:03.10 | freetown2 | hi all, i have a problem. Asterisk appears to be trying to hand off sip connections: http://pastebin.com/0XG3Q9F0 |
| 05:03.38 | freetown2 | but one peer is behind a nat so calls work in one direction but not the other |
| 05:04.05 | freetown2 | I have directmedia=nonat set in sip.conf...but asterisk is still not handling the entire call itself |
| 05:06.00 | [TK]D-Fender | That did not include SIP debug or the matching configs |
| 05:06.37 | freetown2 | coming |
| 05:21.29 | freetown2 | http://pastebin.centos.org/11301/ |
| 05:28.18 | [TK]D-Fender | I see no reinvite there... |
| 05:33.20 | freetown2 | oh... |
| 05:33.42 | freetown2 | so...the call connects, i answer but there is no sound from either side |
| 05:36.10 | [TK]D-Fender | Nothing tells me the rest of your networking is sane... |
| 05:36.59 | [TK]D-Fender | but the SIP comms look clean |
| 05:38.23 | freetown2 | calls from provider are fine. it's outgoing calls that are not. The call can get through but no sound and eventually gets drop. asterisk 1.8 here...anything to turn on to debug? |
| 05:41.59 | freetown2 | calls from provider has one extra bit...SIP/hkbnbrad03-000000b9 requested special control 20, passing it to SIP/ucm1-000000ba |
| 05:42.11 | freetown2 | does that make a difference? |
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| 05:43.22 | [TK]D-Fender | No. |
| 05:43.37 | [TK]D-Fender | I'd go look real close at all of your networking |
| 05:49.50 | freetown2 | [TK]D-Fender, when you say networking, you mean firewalls, ip connectivity or something asterisk related? |
| 05:49.58 | [TK]D-Fender | all of it |
| 05:58.54 | freetown2 | [TK]D-Fender, thanks...found the culprit...apparently a firewall rule for the two internal peers was not applied...i suppose the call was made with tcp and the media reverted back to udp which got dropped by the firewall... |
| 05:59.13 | [TK]D-Fender | Audio is always UDP |
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| 06:04.45 | freetown2 | [TK]D-Fender, thank you very much for your help |
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| 08:20.48 | kuraihikari | hi |
| 08:21.28 | kuraihikari | I'm trying to set up fax via asterisk, but I have hit a wall for the SendFAX and ReceiveFax applications |
| 08:21.28 | pa | hello |
| 08:21.44 | pa | you mean using hylafax? |
| 08:21.54 | kuraihikari | no, digium fax |
| 08:21.57 | pa | ah |
| 08:22.01 | pa | ok then i dont know |
| 08:22.31 | kuraihikari | I've been looking through the documentation, and the only way that I can see to set the destination number is to use a call file |
| 08:23.06 | kuraihikari | http://www.digium.com/sites/digium/files/fax-for-asterisk-manual.pdf |
| 08:23.18 | kuraihikari | here's the doc for their fax module... |
| 08:23.30 | kuraihikari | maybe you can look and see if there are some similarities with hylafax? |
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| 08:31.51 | kuraihikari | is there anybody who worked with DigiumFAX? |
| 08:35.37 | kleszcz | me |
| 08:36.20 | kuraihikari | could you tell me what's the best way to send a fax with digium? |
| 08:39.10 | kleszcz | on * CLI: fax show capabilities |
| 08:39.56 | kuraihikari | Registered FAX Technology Modules: |
| 08:39.57 | kuraihikari | Type : DIGIUM |
| 08:39.57 | kuraihikari | Description : Digium FAX Driver |
| 08:39.57 | kuraihikari | Capabilities : SEND RECEIVE T.38 G.711 MULTI-DOC |
| 08:39.57 | kuraihikari | 1 registered modules |
| 08:40.26 | kleszcz | great |
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| 08:40.53 | kleszcz | now: fax show settings |
| 08:41.15 | kleszcz | then, please show me extensions.conf |
| 08:41.23 | kleszcz | pastebin.com |
| 08:41.42 | kuraihikari | I have no extension set up for faxing now |
| 08:42.28 | kuraihikari | FAX For Asterisk Settings: |
| 08:42.29 | kuraihikari | <PROTECTED> |
| 08:42.29 | kuraihikari | <PROTECTED> |
| 08:42.29 | kuraihikari | <PROTECTED> |
| 08:42.29 | kuraihikari | <PROTECTED> |
| 08:42.29 | kuraihikari | <PROTECTED> |
| 08:42.30 | kuraihikari | FAX Technology Modules: |
| 08:42.32 | kuraihikari | DIGIUM (Digium FAX Driver) Settings: |
| 08:42.34 | kuraihikari | <PROTECTED> |
| 08:42.36 | kuraihikari | <PROTECTED> |
| 08:42.38 | kuraihikari | <PROTECTED> |
| 08:43.31 | kleszcz | great, now extensions.conf, your dialplan send/recive fax |
| 08:43.59 | kuraihikari | that's kind of what I'm asking directions for |
| 08:44.07 | kuraihikari | how to set it up |
| 08:44.54 | kleszcz | ok, i give you my conf, ok? |
| 08:45.01 | kuraihikari | ok |
| 08:45.09 | kuraihikari | thx |
| 08:45.49 | kleszcz | you use some sip trunk or maybe E1 ? |
| 08:46.10 | kuraihikari | no, I want to send it through DAHDI |
| 08:46.56 | kleszcz | great |
| 08:50.22 | pa | WIMPy, you around? |
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| 09:18.43 | gsimpson1712 | Hey guys. Looking for a bit of advice if you would be so kind. I have a request to set up an asterisk system for a few small clients (10-20 users per site). I was thinking about hosting it on something like AWS. My 2 questions are: 1) Are there any show stoppers running in a virtual environment for production. 2) Is multi tenancy achieved by just using contexts or are there other settings required |
| 09:18.43 | gsimpson1712 | ? Many thanks |
| 09:25.36 | paolo_ | Hi, i am doing some tests with TLS and SIP. In sip.conf you can set tlscipher (wich is an SSL cipher string) and tlsclientmethod. I am asking my self for tlscipher is needed. AFAIK TLS have his own set of ciphers. For TLS there are cipher strings TLSv1.2, TLSv1, SSLv3, SSLv2. |
| 09:26.35 | paolo_ | So if tlsclientmethod is set to sslv2 and tlscipher is set to TLSv1 could be wrong |
| 09:35.29 | paolo_ | hmm.... what does ssl cipher even mean. can i only use TLSv1.2, TLSv1, SSLv3, SSLv2 ? Or can I use other ciphers too ? |
| 09:36.35 | paolo_ | s/ciphers/cipher strings |
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| 09:36.57 | Bonzor | hey guys, anyone around? |
| 09:37.16 | coppice | paolo: those aren't cipher options. They are versions of the SSL spec |
| 09:37.20 | zamba | paolo_: that's not ciphers.. that's the different protocol versions.. |
| 09:37.50 | paolo_ | zanmba, There are cipher string TLSv1.2, TLSv1, SSLv3, SSLv2 |
| 09:38.06 | paolo_ | zamba, http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS |
| 09:38.56 | paolo_ | zamba, these are protocols and cipher strings ;-) |
| 09:39.23 | zamba | paolo_: no, that's just confusing.. TLSvX.Y and SSLvX are NOT ciphers |
| 09:39.53 | zamba | but i see they've aliased the ciphers specific to the different versions of TLS/SSL by grouping them |
| 09:39.54 | paolo_ | zaba, i did'n said they are ciphers |
| 09:40.09 | zamba | ok, then we agree to agree :) |
| 09:40.23 | paolo_ | zamba, i said TLS have his owns set of ciphers |
| 09:40.29 | Bonzor | hey guys, i have a gsm gateway that i have setup as a trunk, Im trying to configure the channel to wait or idle for a small ammount of time after completeing a phonecall, anyone setup something like this? |
| 09:40.34 | paolo_ | zamba, OK :-) |
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| 09:42.12 | paolo_ | zamba, i was thinking: what if you choose a cipher string that have no cipher in common with tlsclientmethod=tlsv1, for example |
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| 09:43.57 | paolo_ | zamba, or can you use any cipher available in openssl for TLS ? |
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| 09:44.54 | zamba | paolo_: frankly i have no idea :) |
| 09:44.56 | zamba | paolo_: sorry |
| 09:48.27 | paolo_ | zamba, thanks anyway. It was nice to chat with you :-D |
| 09:51.15 | Bonzor | anyone know how to configure a trunk channel to to be unavailible or idle for a few seconds after a phone call? |
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| 10:32.07 | pa | WIMPy, i think you use lcr. what kernel version do you have? and do you have the module mISDN_l1loop available? |
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| 11:00.00 | D30 | hi all, i have a question regarding FXS ports on a regular TDM card by digium... |
| 11:01.03 | D30 | I'm trying to call using a regular analog phone by inserting it directly to a FXS ports on a digium card... while a regular POT line is inserted to a FXO port.. |
| 11:01.24 | D30 | will that directly work? or do i have to configure asterisk first? |
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| 11:14.29 | pa | actually you make me wonder a similar thing: are these digium cards still supported? http://www.digium.com/en/products/telephony-cards/digital/euro-isdn-bri |
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| 11:15.06 | pa | i can see that the chip is a cologne chip |
| 11:15.10 | pa | so.. |
| 11:16.02 | Chainsaw | D30: You need to have a running Asterisk instance with dial plan to make that possible. |
| 11:16.33 | D30 | Chainsaw: yes i do have a running asterisk already |
| 11:17.03 | Chainsaw | D30: Your question seems to be "is there auto pass through" to which I can give you a resounding no. |
| 11:17.21 | D30 | what you mean Chainsaw ? |
| 11:17.43 | D30 | do i need to create an extension for it too?? |
| 11:17.45 | Chainsaw | D30: "Will that directly work". If "that" is "auto pass through", then no. |
| 11:19.28 | D30 | Chainsaw: do i need to create an extension when using an analog phone directly connected to the FXS port? |
| 11:21.27 | Chainsaw | D30: Yes. |
| 11:21.57 | D30 | okay thanks... |
| 11:24.56 | D30 | i am confused how to write the dial plan for a fxs port hehe |
| 11:25.18 | D30 | or do i misinterpreted it |
| 11:27.01 | sekil | Dial(Dahdi/1/1) i.e |
| 11:27.49 | WIMPy | pa: You don't need that loop module. |
| 11:28.57 | pa | oh i see! i found some guide that said it's necessary to pass something to something else to asterisk |
| 11:29.09 | pa | but then it's really good news :) |
| 11:29.44 | WIMPy | No, it is (or was?) neccessary for use with the GSM stuff. |
| 11:29.59 | pa | WIMPy, can i safely "bridge" the two interface (asterisk and isdn i mean) in lcr? |
| 11:30.13 | pa | so that i can avoid myself doing routing and stuff |
| 11:30.41 | WIMPy | Probably. I still use the old version whe Asterisk is a "remote application". |
| 11:31.34 | WIMPy | With the old version it's easier to route to different interfaces from within Asterisk. |
| 11:31.50 | pa | oh i see |
| 11:31.56 | pa | thanks then i will give it another try :) |
| 11:32.02 | pa | i feel less depressed suddenly :) |
| 11:33.39 | pa | then brb from 14.04 |
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| 11:43.06 | pa | WIMPy, when i run lcr (lcr start), before it says "waiting for calls" it complains that "ERROR Port name 'mISDN_l1loop.1' not found, did you load loopback interface?" |
| 11:43.29 | WIMPy | No. I don't have it. |
| 11:44.13 | WIMPy | Do you have some junk in your interfaces.conf? |
| 11:46.12 | pa | no, my interfaces.conf is pretty neat: http://paste.ubuntu.com/7969720/ |
| 11:47.36 | WIMPy | Err, yes. |
| 11:47.55 | WIMPy | MAybe it's a new thing. |
| 11:48.37 | WIMPy | Maybe because of the new way Chan_lcr creates an interface? |
| 11:49.00 | pa | could be.. let me dig a lttle into it, i will come back soon :) |
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| 12:10.50 | pa | WIMPy, with debug on, this is what happens when LCR receives a call: http://paste.ubuntu.com/7969899/ . itseems that asterisk connects to LCR but the call is not forwarded to it |
| 12:11.10 | pa | line 32 and 33 are a little suspicious |
| 12:15.00 | WIMPy | Unfortunately I don't know what it's supposed to look like with the new interface style configuration. |
| 12:15.24 | WIMPy | Do you see asterisk connected in 'lcradmin state'? |
| 12:15.45 | WIMPy | Have you tried turning up verbose and debug in Asterisk? |
| 12:16.06 | pa | yes i ran asterisk with -cvvvvvvvvvvvvvvvvv....v |
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| 12:16.14 | pa | and 0xffff debug on lcr |
| 12:16.24 | WIMPy | Add some -d |
| 12:16.25 | pa | usually asterisk speaks when a call arrives |
| 12:16.30 | pa | ok |
| 12:16.38 | WIMPy | But it looks to me as if chan_lcr might not be loaded. |
| 12:17.10 | pa | well lcr says: REMOTE APP registers app name=asterisk |
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| 12:17.40 | pa | maybe i should have compiled that with debug |
| 12:17.47 | pa | coz it does not speak much |
| 12:17.58 | pa | i mean chan_lcr |
| 12:18.31 | WIMPy | But it is connected? |
| 12:18.53 | pa | actually |
| 12:18.59 | pa | it works from asterisk to ISDN |
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| 12:19.09 | pa | but it seems not to work the other way around |
| 12:19.12 | WIMPy | Ok, that's a good start. |
| 12:19.16 | pa | or maybe it simply does not say anything |
| 12:19.28 | WIMPy | Then maybe it just doesn't find the extension. |
| 12:19.41 | pa | could be.. i could add some debug extension |
| 12:19.43 | WIMPy | But you should be able to see that with debug enabled. |
| 12:19.49 | pa | like "write the CID to a file" |
| 12:20.06 | pa | WIMPy, can it be because i compiled chan_lcr without enabling debug? |
| 12:20.52 | WIMPy | IIRC that becomes very verbose. |
| 12:21.03 | pa | gm |
| 12:21.04 | WIMPy | Usually Asterisks debug should tell you something. |
| 12:21.10 | pa | right |
| 12:21.44 | pa | right now, in the other direction i only get "sending MESSAGE_XX to socket" |
| 12:22.19 | WIMPy | So does it work or doesn't it? |
| 12:22.31 | pa | from asterisk to isdn it does |
| 12:22.48 | pa | but i cant see anything when i receive a call |
| 12:23.07 | pa | but probably it could be some extension problem |
| 12:23.09 | WIMPy | Ok, so at least we know that everyting needed is available and working. |
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| 12:23.40 | WIMPy | Maybe the extension matching has changes as well. |
| 12:23.52 | WIMPy | You could try to make a wildcard extension. |
| 12:24.57 | pa | right.. but with this file http://paste.ubuntu.com/7969720/ what would be the number of the extension that asterisk receives? my home number? the caller id number? |
| 12:25.06 | pa | i'm asking since i can't see anything in the log |
| 12:25.48 | WIMPy | Erm, I think it's a configuration issue. I think you have to configure the Asterisk context in the interface definition. |
| 12:26.00 | WIMPy | The called number. |
| 12:28.23 | pa | right.. might be that i miss the context |
| 12:29.53 | pa | ouch.. Error in /usr/local/etc/lcr/interface.conf (line 4): unknown parameter: 'context'. |
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| 12:31.49 | pa | i think it's easier if i try to rebuild with debug on |
| 12:31.55 | pa | at least i might get some hints |
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| 12:34.19 | WIMPy | Hmm. The context has to be defined somewhere, obviousely. |
| 12:35.00 | WIMPy | And in the new version it has to be in interfaces.conf. |
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| 12:40.13 | pa | what looks suspicious though, is CHANNEL SELECTION (port not available, skipping) port -1 position 0 |
| 12:40.57 | WIMPy | Definitely |
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| 12:43.57 | WIMPy | But you definitely need a context somewhere. Have you checked 'lcr interface' for help? |
| 12:44.37 | pa | not now, i can re-check but i think i didnt find context , yesterday, when i was looking at it |
| 12:45.10 | pa | i remember right, it's not there |
| 12:45.50 | WIMPy | Oh, wait. Could it be the interface name is used as context or something? |
| 12:45.58 | pa | http://paste.ubuntu.com/7970113/ |
| 12:46.00 | WIMPy | definitely prefers the old way. |
| 12:46.21 | pa | hm.. it could be.. would be a little weird though |
| 12:46.35 | pa | i can try |
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| 12:49.54 | pa | same error. now with interface name = asterisk context |
| 12:50.09 | pa | port not availablbe |
| 12:50.18 | pa | portname |
| 12:50.22 | pa | maybe that is what i have to use |
| 12:51.18 | pa | maybe not |
| 12:51.22 | pa | it says about isdninfo |
| 12:52.37 | pa | maybe i can't do bridging |
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| 12:55.03 | RadJackson | Hello , i need to integrate a speech recognition system in my PBX , i was using Google Speech Recognition , but seems like google have stopped this API. And the new api is limited to 50 requests per day , i've seen on forums people talking about Sphinx , LumenVox , Julia ... All i need is to detect two words , no more. so its a two words recognizer ,what would you suggest ? |
| 12:55.04 | RadJackson | thanks |
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| 12:57.09 | tparcina | Macro exits non-zero - http://pastebin.com/z9C13hkD |
| 12:57.17 | tparcina | how can I find out what's the reason? |
| 12:58.20 | [TK]D-Fender | Get a better log. |
| 12:59.04 | tparcina | [TK]D-Fender: Why didn't I think of that? :D |
| 12:59.16 | tparcina | [TK]D-Fender: Thank you. |
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| 12:59.43 | WIMPy | pa: I think Jolly said you can use different contexts by defining multiple interfaces. But it was actually a discussion about why he changed it. |
| 13:02.28 | Katty | morning |
| 13:05.42 | pa | hm.. still it should work, right? |
| 13:05.57 | pa | i mean at least with interface name = context name |
| 13:06.02 | pa | i can look at the code |
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| 13:06.05 | pa | i have it, after all |
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| 13:07.33 | WIMPy | Well, as I said, I haven't tried the new way, yet. It would make things more complicated the way I use it. |
| 13:11.10 | pa | WIMPy, maybe it's really a bridge problem. because it says CHANNEL SELECTION (found given interface) interface lcrasterisk, but then it says CHANNEL SELECTION (port not available, skipping) port -1 position 0 |
| 13:11.31 | pa | and this happens when: if (!ifport->mISDNport) |
| 13:11.50 | pa | so it does not find a mISDNport in interface asterisk |
| 13:12.09 | WIMPy | Sounds to me like the interface just does not work. And I can only imaginethere's something missing. |
| 13:12.43 | WIMPy | Hmm. So maybe it needs the loopback interface for that? |
| 13:12.59 | pa | cant i do routing without loopback? |
| 13:13.14 | pa | like i would like to route everything from mISDN to lcrAsterisk |
| 13:13.16 | pa | and vice-versa |
| 13:13.28 | WIMPy | Maybe you should downgrade to 1.13? |
| 13:13.40 | pa | i have 1.13 i think? |
| 13:13.51 | pa | loopback i dont think i need.. the other way works fine |
| 13:13.55 | WIMPy | Not master? |
| 13:14.09 | pa | no i have 1.13 |
| 13:14.31 | pa | can double check |
| 13:14.41 | WIMPy | Oh, I didn't think that already had the new way. |
| 13:14.57 | WIMPy | But it might just be work in progress, I guess. |
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| 13:15.43 | pa | but did you just change the download page? |
| 13:15.48 | pa | i see now it's 1.14 there |
| 13:16.52 | WIMPy | Where do you see that? |
| 13:16.59 | Bonzor | anyone know how to configure a trunk channel to to be unavailible or idle for a few seconds after a phone call? |
| 13:17.33 | WIMPy | What kind of trunk? |
| 13:17.42 | Bonzor | its a gsm gateway |
| 13:18.38 | Bonzor | i just want the hangup to process, then have the line in "cool down" so to speak, for a few secs |
| 13:19.09 | pa | WIMPy, yes 1.13 is what i use, and has no context |
| 13:19.29 | pa | i try a newer version |
| 13:19.48 | pa | 1.14 does not exist |
| 13:20.03 | WIMPy | Yes, mine is based on 1.12. So that does have the old way. |
| 13:20.06 | pa | (http://voice.yeti.dk/Asterisk_vs_ISDN/6) |
| 13:20.21 | pa | would 1.12 work on a 3.13 kernel? |
| 13:20.24 | WIMPy | oops? |
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| 13:20.50 | pa | <PROTECTED> |
| 13:20.55 | WIMPy | Where did you see 1.14? |
| 13:21.06 | pa | in your page |
| 13:21.12 | WIMPy | I'm on 3.15. |
| 13:21.20 | pa | git clone git://git.misdn.eu/lcr.git/ -b 1.14 lcr-1.14 |
| 13:21.40 | pa | oh.. then i can try the one you use, sure |
| 13:21.55 | WIMPy | Hmmm. yes, that should be 1.13. |
| 13:22.12 | WIMPy | Maybe I should also list 1.12 for the old style. |
| 13:22.34 | pa | would be useful i think |
| 13:23.07 | pa | but in theory, do i need loopback if i simply want to route everything from mISDN to asterisk ? it seems to work the other way around without loopback (but with bridging) |
| 13:23.38 | WIMPy | I don't know the requirements for the new way. |
| 13:24.07 | WIMPy | But as you said it moans about not finding the loopback interface, there's a chance it is required now. |
| 13:24.08 | pa | maybe i should try to contact the author.. the howto is kind of outdated |
| 13:24.36 | WIMPy | You can either do so directly or on the I4L mailing list. |
| 13:26.28 | pa | anyhow, i will first try 1.12 |
| 13:26.31 | *** part/#asterisk Zogot (~Adium@D4B2620B.static.ziggozakelijk.nl) |
| 13:26.56 | pa | but it's not tagged in the git |
| 13:27.16 | WIMPy | Oh? That's evil. |
| 13:28.00 | WIMPy | True. There's a gap between 1.11 and 1.13. |
| 13:28.46 | pa | hm.. |
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| 13:29.19 | Stefan27 | What's the deal with two underscores added before a variable name? e.g. setting __TRANSFER_CONTEXT=testcontext vs TRANSFER_CONTEXT=testcontext in [globals] in extensions.conf (in Ast 12.3) or __DYNAMIC_FEATURE vs DYNAMIC_FEATURE. http://www.voip-info.org/wiki/view/Asterisk+config+features.conf suggested using "__TRANSFER_CONTEXT" but my old asterisk uses "TRANSFER_CONTEXT" and both seem to |
| 13:29.19 | Stefan27 | work the same way?? |
| 13:29.43 | WIMPy | wonders if he needs to put his version online... |
| 13:30.41 | pa | wouldnt hurt :) |
| 13:32.09 | WIMPy | Maybe I need to officially fork it? |
| 13:32.14 | WIMPy | Hmmm. |
| 13:32.33 | WIMPy | didn't really intend to do so. |
| 13:33.24 | pa | maybe 1.13 works , it's just me who cant properly configure it |
| 13:33.40 | pa | im trying to write a routing conf now |
| 13:33.49 | pa | but the syntax is not exactly straightforward |
| 13:34.02 | WIMPy | Or it just needs the loop module. |
| 13:34.02 | pa | i see the author loves goto's both in the code and in the config file |
| 13:34.25 | pa | but why would it work one way? |
| 13:35.55 | WIMPy | http://voice.yeti.dk/lcr-1.12w1.tar.xz |
| 13:36.03 | pa | also, could i do directly within LCR something like i do in asterisk with "exten => MYISDNNUMBER,1,System(/root/bin/INSERTCALL ${CALLERID(num)})" ? |
| 13:36.19 | pa | if i could do that, then one way can be enough for me |
| 13:36.33 | pa | (thanks) |
| 13:37.02 | WIMPy | I do all from within Asterisk, but I'm pretty sure I did see a shell command. |
| 13:39.12 | WIMPy | But if you only want call infos, you might as well take a look at the logging utilities. |
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| 13:40.19 | pa | well i was planning to set up an answering machine, and eventually use it for pbxing, but i could live a little more like i live now, where isdn is used only to call landlines, and receive call infos |
| 13:40.39 | *** part/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 13:49.56 | pa | ok tried, does not work, still.. maybe you are right, i need the loopback, donno.. I try your version now :) |
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| 13:54.26 | pa | WIMPy, is it normal that if i run lcr interface on your version, context is still missing? |
| 13:54.55 | WIMPy | Yes. It's using the old style Asterisk integration. |
| 13:55.38 | pa | right |
| 13:55.47 | WIMPy | So you don't have an Asterisk interface. You send calls to Asterisk via a line in routing.conf. |
| 13:56.03 | pa | ah |
| 13:56.21 | pa | ok i check the routing.conf in the package, there are probably examples |
| 13:56.52 | WIMPy | Something like: |
| 13:56.57 | WIMPy | remote=asterisk : remote application=asterisk context=in-dtag |
| 13:57.36 | WIMPy | The condition isn't really neccessary. I just have it so I can do fail-over if Asterisk should not be running. |
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| 13:58.14 | WIMPy | That's the only line you need under [main]. |
| 13:58.57 | pa | but if i want to route everything from "mISDN" interface to "lcrasterisk" interface, what would that look like? remote=lcrasterisk: remote application=asterisk context=lcrasterisk ? |
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| 14:01.18 | pa | hm.. that seems wrong. it says "Condition item 'remote' must have at least one value, '=' expected, and not a space." |
| 14:01.38 | WIMPy | The part before the colon are conditions. So in my example it just means to do it only if Asterisk (chan_lcr) is connected. |
| 14:02.31 | WIMPy | And the name of the remote application is fixed. You just change the context to what you have in your extensions.conf. |
| 14:03.21 | WIMPy | The Asterisk interface doesn't exist in that version, yet. |
| 14:03.46 | pa | ah i see |
| 14:03.49 | pa | so i have to remove it |
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| 14:06.10 | pa | ok, the asterisk -> isdn still works |
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| 14:07.03 | pa | now the other way too :) |
| 14:07.16 | pa | asterisk just fails with some timeout :) |
| 14:07.22 | WIMPy | Great |
| 14:07.24 | pa | WIMPy, thanks a lot! |
| 14:07.32 | pa | it was a challenge :) |
| 14:08.07 | WIMPy | Yes, looks like the situation has become even worse than it was. |
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| 14:08.46 | pa | ideally dahdi_hfcs should be updated |
| 14:08.50 | pa | or chan_capi |
| 14:09.01 | pa | an additional daemon should not be necessary |
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| 14:10.02 | WIMPy | I know that Karsten Keil wrote an chan_misdn2, but never finished it. So it's not available anywhere :-( |
| 14:10.21 | WIMPy | And it definitely was for an rather outdated Asterisk version. |
| 14:13.06 | Qwell | if it's as old as I expect, it would have been when he was at Novell. I doubt they would have let him contribute it back. |
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| 14:17.43 | WIMPy | Well, they have always been a little special in many ways. |
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| 15:00.55 | sekil | remembers novell |
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| 15:19.06 | pa | by the way, do you have an idea why ithe call gets to asterisk, but it timeouts and i can't hear anything? it does this : http://paste.ubuntu.com/7971173/ |
| 15:19.38 | pa | i have the same chain (answer; mp3player, hangup) on a different extension that i can call directly through sip, and it still works fine |
| 15:20.46 | gsimpson1712 | Hey guys. Looking for a bit of advice if you would be so kind. I have a request to set up an asterisk system for a few small clients (10-20 users per site). I was thinking about hosting it on something like AWS. My 2 questions are: 1) Are there any show stoppers running in a virtual environment for production. 2) Is multi tenancy achieved by just using contexts or are there other settings required |
| 15:20.47 | gsimpson1712 | ? |
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| 15:23.26 | pa | well, nevermind, doesnt matter anyway. it does what it should :) |
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| 15:27.17 | [TK]D-Fender | gsimpson1712: Contexts define what happens when devices place calls. |
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| 15:29.19 | gsimpson1712 | @Fender. Yeah is that the wrong way to do it? |
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| 15:31.23 | [TK]D-Fender | gsimpson1712: Dialplan is one part of your system |
| 15:31.39 | [TK]D-Fender | gsimpson1712: Voicemail, SIP, etc, all to be considered |
| 15:33.00 | kos_tom | hello. As I said yesterday, I'm writing a custom DAHDI driver, which talks over SPI to an ISDN chip. However, we have performance issues because it's really hard to do all the SPI communication that needs to be done within one millisecond (which is the timer frequency used to send/receive 8 bytes) |
| 15:33.05 | gsimpson1712 | Sure. I get you. My main concern was dealing with overlapping dial plans. You know haw everyone likes having 10XX ;) |
| 15:33.19 | kos_tom | what are the consequences of increasing DAHDI_CHUNKSIZE to 32, and therefore get an interrupt only every 4 ms ? |
| 15:33.30 | kos_tom | according to the thread at http://lists.digium.com/pipermail/asterisk-ss7/2010-March/003544.html it doesn't seem to be trivial |
| 15:33.50 | kos_tom | and according to http://www.sangoma.com/assets/docs/misc/2009_10_09_How_to_Reduce_Asterisk_System_Loads.pdf it means you have to give up on software echo cancellation |
| 15:33.56 | gsimpson1712 | What about the AWS idea. Any worries about running a production system on something like that? |
| 15:33.58 | kos_tom | but that was back in 2009/2010, what about today ? |
| 15:34.31 | [TK]D-Fender | gsimpson1712: https://www.google.ca/#q=running+asterisk+on+AWS |
| 15:36.00 | gsimpson1712 | yep. I have done that. Was more wondering about if any of you guys had any experience doing it. I have been running tests for a few weeks and all seems ok. |
| 15:36.46 | gsimpson1712 | Its always good to get get real feedback from people rather than reading blogs |
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| 15:37.10 | [TK]D-Fender | gsimpson1712: Confirmation from those who've already posted articles, plus having experience youself should tell you what you need to know... |
| 15:39.11 | gsimpson1712 | Fair enough. I wasn't asking for someone to tell me how to do something. I guess I just like bouncing ideas of real people. Thanks anyway! |
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| 18:44.52 | kos_tom | what is the appropriate mailing list to post questions about DAHDI ? |
| 18:44.58 | kos_tom | (developer questions) |
| 18:45.08 | Qwell | kos_tom: probably asterisk-dev |
| 18:45.17 | kos_tom | is it asterisk-dev ? (but looking at the archives, it seems to really be focused on asterisk itself, and not really on dahdi) |
| 18:45.54 | Qwell | There are dahdi discussions there occasionally. |
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| 19:09.24 | pa | WIMPy, are you still around? i have one last problem with LCR :) |
| 19:10.00 | pa | it seems that international numbers caller id comes without 00/+ .. do you know whether it's possible to make LCR leave it on? |
| 19:10.19 | pa | also in LCR logs the 00 or + is stripped away |
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| 19:47.55 | WIMPy | pa: That's normal. Those prefixes aren't part of the nnumber. |
| 19:48.45 | WIMPy | You can find the Type Of Number with CALLERID(ton) or you can so search/replace in LCR with screen-in rules. |
| 19:49.28 | WIMPy | If you do the later, make sure to reset the TON to "unknown". |
| 19:49.48 | pa | oh, i see! thanks |
| 19:50.00 | pa | i think asterisk was giving me the 00 by default on CID |
| 19:50.06 | pa | with zaphfc i mean |
| 19:50.19 | WIMPy | screen-in national % unknown 0% |
| 19:50.24 | WIMPy | screen-in international % unknown 00% |
| 19:50.52 | WIMPy | Add that to your interface, if you want the prefixes in the number. |
| 19:51.13 | pa | wow thank you very much! : ) |
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| 20:16.01 | *** join/#asterisk Kattyroo (61577062@gateway/web/cgi-irc/kiwiirc.com/ip.97.87.112.98) |
| 20:16.13 | Kattyroo | hello my asterisk does not work at all how to fix plz |
| 20:18.24 | *** join/#asterisk SirLouen (~pickupdog@95.63.247.9) |
| 20:18.48 | SirLouen | hello |
| 20:19.07 | SirLouen | has anyone tryied 11.11 webrtc support? |
| 20:19.17 | Kattyroo | hello, SirLouen! |
| 20:19.22 | Kattyroo | i have not. |
| 20:19.29 | Kattyroo | but perhaps others have. been a bit quiet lately |
| 20:19.47 | SirLouen | Kattyroo have u used kamailio before? |
| 20:20.02 | Kattyroo | no i have not. |
| 20:20.08 | SirLouen | oook |
| 20:20.22 | SirLouen | i'm trying various solutions, but now i'm a bit lost |
| 20:20.29 | SirLouen | kamailio is a little bit complicated |
| 20:20.34 | Kattyroo | yes that tends to happen with you try things out |
| 20:20.42 | Kattyroo | pesky ol learning curve! |
| 20:20.45 | Kattyroo | you will get it tho! |
| 20:21.51 | *** join/#asterisk slav3_kitten (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 20:22.15 | Kattyroo | slav3_kitten: hello, cupcake! |
| 20:24.13 | slav3_kitten | yo? |
| 20:24.52 | Kattyroo | how're you dear? |
| 20:24.56 | slav3_kitten | uhhh |
| 20:25.01 | slav3_kitten | not great? |
| 20:25.10 | Kattyroo | no? why not? |
| 20:25.17 | slav3_kitten | network problems |
| 20:26.20 | Kattyroo | what a bummer :< |
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| 20:34.34 | pa | WIMPy, also, zaphfc used to add a 0 for landlines and no 0 for mobile lines (which is in effect the actual dialed number). LCR strips all leading zeros.. is that also some ISDN parameter or is LCR who strips? |
| 20:34.59 | WIMPy | Where? |
| 20:42.33 | pa | on the caller id |
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| 20:43.22 | litn | in AMI how do I show ... what extensions are associated with a channel? For example I am detecting music on hold events, and I get the channel, but how do I then find out who all is on that channel? |
| 20:43.40 | WIMPy | pa: We just told it to add the 0s,didn't we? |
| 20:44.09 | pa | yes but it does so on landlines and mobiles |
| 20:44.21 | pa | and in practice mobiles do not need one |
| 20:44.28 | WIMPy | I don't understand. |
| 20:44.37 | pa | ok so, example |
| 20:45.21 | pa | landline number 06123456 -> becomes 6123456 im LCR |
| 20:45.43 | pa | mobile number 33912345 -> becomes 33912345 in LCR |
| 20:46.04 | tparcina | Asterisk doesn't write CDR information to remote (on another server) MySQL database. |
| 20:46.10 | tparcina | How can I troubleshoot this? |
| 20:46.18 | pa | if we add 0 to both, the mobile number becomes incorrect |
| 20:46.19 | WIMPy | Don't mix it up. The 0s at the beginning are not transitted in the PSTN. LCR doesn't change anything unless you set up rules to do so. |
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| 20:46.36 | pa | hm.. are you sure? |
| 20:46.55 | WIMPy | yes |
| 20:47.07 | WIMPy | So you don't need a 0 to call mobiles? |
| 20:47.14 | pa | exactly |
| 20:47.19 | WIMPy | Is it like calling a local number? |
| 20:47.25 | pa | so is it possible to know if the call is from mobile? |
| 20:47.36 | WIMPy | No |
| 20:47.41 | pa | hm.. |
| 20:47.43 | WIMPy | You only get the TON. |
| 20:47.55 | WIMPy | And that should hopefully be correct |
| 20:47.57 | pa | but then how did zaphfc do? |
| 20:48.52 | WIMPy | The mappings are also configured in chan_dahdi.conf. Just that they don;t reset the TON IIRC, so it's more of a musfeature there. |
| 20:49.31 | pa | ah i see |
| 20:49.34 | Kattyroo | tparcina: the CLI should give you some information when the CDR fails to be recorded in the database |
| 20:49.48 | Kattyroo | tparcina: so set your verbosity to 10 and run a test (= |
| 20:49.52 | WIMPy | You can see the TON in the logs. |
| 20:50.03 | tparcina | Kattyroo: Thank you, I'll try that. |
| 20:50.38 | keiths_ | have a good night folks |
| 20:50.45 | Kattyroo | g'night! |
| 20:52.49 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 20:53.03 | pa | WIMPy, LCR logs? cant find it.. |
| 20:54.40 | WIMPy | calling_pn type= |
| 20:54.43 | keiths_ | <PROTECTED> |
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| 20:57.01 | pa | WIMPy, so yes i have 3 different types for : national landline, national mobile, foreign mobile (dont have a foreign landline to try) |
| 20:57.02 | Kattyroo | [TK]D-Fender: fenderbender! |
| 20:57.20 | [TK]D-Fender | Kattyroo: Mew. |
| 20:57.32 | Kattyroo | how're you dear? |
| 20:58.03 | tparcina | Kattyroo: I don't get nay error message at CLI. Like Asterisk isn't trying to write anything to the database. |
| 20:58.07 | WIMPy | pa: Ok, so you can have different rules. |
| 20:58.33 | [TK]D-Fender | Kattyroo: Doing OK. Found some malicious PHP in my company's website today. Stuff getting cleaned up and getting ready to host it ourselves... |
| 20:58.38 | Kattyroo | tparcina: well the, looks like you need to make sure it's enabled and all of the details are there! |
| 20:58.59 | Kattyroo | [TK]D-Fender: ouch. can't say i blame you there |
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| 21:39.18 | ashkanull | hello |
| 21:39.59 | ashkanull | I'm new to asterisk and I'm lookin to develope saynumber() in my language, can anyone hook me with a startin point ? |
| 21:40.19 | ashkanull | plz :D |
| 21:41.00 | WIMPy | say.conf |
| 21:46.53 | ashkanull | WIMPy I'm lookin at https://github.com/rillian/asterisk-opus/blob/master/configs/say.conf.sample and en-base seems fine to me except for ther plural ruls millions, thousands ... . |
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