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| 00:19.05 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.16.0 (2017/05/30), 11.25.1 (2016/12/08), Standard: 14.5.0 (2017/05/30); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 06:21.56 | TazzNZ | does anyone know if the comm. support from digium will cover asterisk 13.1 ? (the version that ships with Ubuntu 16.04) - I suspect it would |
| 06:23.37 | drmessano | Are you referring to one-off support or contract? |
| 06:24.02 | TazzNZ | one-off |
| 06:24.14 | drmessano | I've never known it to matter |
| 06:24.19 | drmessano | If you have an issue, call |
| 06:24.29 | drmessano | Lines are open to take your money |
| 06:25.03 | TazzNZ | cool - I'll sort it out "tomorrow" |
| 06:25.11 | TazzNZ | t38 killed my brain :( |
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| 11:24.51 | xochilpili | hi all |
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| 11:33.30 | xochilpili | i have a new ITSP, but the trunk i have configured in sip.conf as [out-route], works of a period of time, then yesterday i just restart eth interfaces, and works again, but now it doesnt |
| 11:33.37 | xochilpili | any idea what could be happening? |
| 11:33.54 | xochilpili | i think about a expirtation time of the register |
| 11:34.27 | xochilpili | is it different to make a register=> and make a [out-route] as an extension ? |
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| 11:49.51 | xochilpili | is there a way to debug only registers? |
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| 12:30.49 | pawiecki | xochilpili: Your ITSP should give you details on how to configure your Trunk properly. If you are not sure, ask them or look for information ot their website for example. |
| 12:31.37 | Samot | xochilipili: sip set debug on |
| 12:31.44 | Samot | Make REGISTER attempt. See results. |
| 12:31.59 | Samot | Or watch for REGISTER attempts. |
| 12:38.01 | xochilpili | Samot, thanks for answer, the itsp give me the details, i could connect, but for an hour, then i got banned or something, because i lost connection |
| 12:38.38 | Samot | I highly doubt you were banned. |
| 12:38.46 | Samot | Is your WAN connection static? |
| 12:39.33 | xochilpili | Samot, yes, that's why i did not use register=> but i think i have to |
| 12:40.07 | xochilpili | now i have wired disconnected, i can make more test later, when personal arrive to the office |
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| 12:42.48 | xochilpili | Samot, i use register=> then how do i need to declare a context for outgoing? |
| 12:43.27 | [TK]D-Fender | Yes you need a peer to call out properly |
| 12:44.00 | xochilpili | im a little confused about it, a register=> .../sample_extension ? do i need to use [sample_extension] << as a trunk ? |
| 12:44.18 | [TK]D-Fender | no |
| 12:44.29 | [TK]D-Fender | they have nothing to do with one another |
| 12:44.45 | [TK]D-Fender | register => user:pass@host/exten |
| 12:45.03 | xochilpili | exten is for incoming, right ? |
| 12:45.07 | [TK]D-Fender | they will be told to dial "exten" when calling you so that is what you should expect to match in the dialplan |
| 12:45.12 | [TK]D-Fender | ^ |
| 12:46.03 | xochilpili | [TK]D-Fender, but for incoming or for outgoing? |
| 12:46.08 | xochilpili | or both? |
| 12:46.23 | [TK]D-Fender | ^<[TK]D-Fender> they will be told to dial "exten" when calling you so that is what you should expect to match in the dialplan <--- |
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| 12:47.59 | xochilpili | [TK]D-Fender, english issues (also), in other words ^ means? |
| 12:49.07 | xochilpili | "they" just gave me a set of DID's but only one is "the head" that is the "exten" you mentioned? |
| 12:49.22 | [TK]D-Fender | -> WHEN CALLING YOU <- |
| 12:49.41 | [TK]D-Fender | THEY will call "EXTEN" when calling YOU |
| 12:49.47 | [TK]D-Fender | that is the NUMBER you request them to DIAL |
| 12:50.00 | [TK]D-Fender | They may mor may not actually FOLLOW this request |
| 12:50.05 | [TK]D-Fender | you'll have to look at the call and see |
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| 12:54.42 | pawiecki | xochilpili: You may want to read a bit. There a free book online about asterisk that might help you. Try it! http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html/asterisk-book.html#DeviceConfig_id283201 and here: http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html/asterisk-book.html#OutsideConnectivity_id291268 |
| 12:57.57 | xochilpili | oh i see what you meant [TK]D-Fender : "Next, you will need to create a peer entry in sip.conf " ... (sic) |
| 12:58.07 | xochilpili | pawiecki, thanks! |
| 12:58.11 | xochilpili | both |
| 12:59.23 | [TK]D-Fender | yes |
| 12:59.32 | [TK]D-Fender | you always have to match your caller |
| 12:59.36 | [TK]D-Fender | that isn't different |
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| 12:59.55 | xochilpili | [TK]D-Fender, yes i think i understand |
| 13:00.23 | xochilpili | btw, another question, if i want to use TLS, is it possible to create a self-signed certificate ? |
| 13:00.48 | xochilpili | or something like starssl needs to be generated? |
| 13:01.34 | xochilpili | the ITSP needs to have my cert right? |
| 13:01.55 | xochilpili | what does ROW means? |
| 13:02.21 | Samot | First your ITSP needs to support TLS |
| 13:02.38 | Samot | Second, if they do they'll tell you what the settings are because it's their certs. |
| 13:03.16 | xochilpili | Samot, all right! thanks! |
| 13:03.34 | Samot | Third, the amount of ITSPs that do TLS is very small |
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| 13:32.46 | navaismo | xochilpili like the aztec god? |
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| 13:51.41 | xochilpili | navaismo, exactly |
| 13:53.23 | navaismo | nice! |
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| 16:01.10 | CrowX- | If I want to set in the context to record a call, do I have to Answer() the call too? |
| 16:02.05 | CrowX- | I mean, do I have to Anser() the call before staring the recording? |
| 16:02.41 | Samot | Yes. |
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| 16:05.25 | CrowX- | Samot, so, after I answer, set to recording, can I set the call to from-pstn? |
| 16:05.38 | Samot | What? |
| 16:07.03 | CrowX- | I'll ask you on that again, but can you please let me know what does 'n' stand for in "exten => _X!,n,Goto(from-pstn,${EXTEN},1)"? |
| 16:07.22 | Samot | next |
| 16:07.37 | CrowX- | ok, thanks |
| 16:07.55 | Samot | What are you trying to do? |
| 16:09.34 | CrowX- | I'm trying to put the call in the context directly from the trunk, put it to an ARI stasis app, subscribe to it via ari, continue the call back into the context from ARI, start recording from the context, and then send it to dialplan, using 'exten => _X!,n,Goto(from-pstn,${EXTEN},1)' |
| 16:09.55 | CrowX- | would this work? |
| 16:10.01 | CrowX- | I currently have this working without the recording part |
| 16:10.03 | Samot | I would need to see your dialplan. |
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| 16:16.45 | DivideBy0 | CrowX-: I don't know if you ever told us your final goal of what you're trying to do |
| 16:16.52 | DivideBy0 | rather than just this part |
| 16:17.10 | [TK]D-Fender | There is no "context to record a call" |
| 16:17.21 | [TK]D-Fender | Your terms or concepts are a little scrambled |
| 16:17.28 | [TK]D-Fender | Call comes IN to your server. |
| 16:17.32 | [TK]D-Fender | You can start at any time |
| 16:17.33 | CrowX- | I'll paste you my current context |
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| 16:17.54 | Samot | Well I was hoping that seeing the dialplan would shed more light. |
| 16:17.54 | [TK]D-Fender | This may have no relation to calling ANOTHER device at all |
| 16:17.59 | [TK]D-Fender | <Samot> Well I was hoping that seeing the dialplan would shed more light. <- that assumes it's correct for his intent |
| 16:18.04 | [TK]D-Fender | it could become even MORE misleading |
| 16:21.01 | CrowX- | Samot, my dialplan is set by Freepbx |
| 16:21.24 | CrowX- | A call comes to the server and it goes to the trunk settings, the trunk sends it to my context. |
| 16:21.27 | CrowX- | Here is my context: https://dpaste.de/WbAX |
| 16:23.05 | [TK]D-Fender | multiple failures there |
| 16:23.12 | CrowX- | go on |
| 16:23.23 | [TK]D-Fender | you have have TWO priority 1 extens there for one extension |
| 16:23.24 | [TK]D-Fender | and |
| 16:23.25 | [TK]D-Fender | exten => s,n,Set(MONITOR_FILENAME=/var/spool/asterisk/recording/${TIMESTAMP}-to_${CONNECTEDLINE(num)}-from_${CALLERID(num)}) |
| 16:23.33 | [TK]D-Fender | this will never get used period |
| 16:23.42 | CrowX- | what should I change? |
| 16:23.55 | [TK]D-Fender | it has an "n" without a "1", and "s" has no relationship to a numbered target being sent inbound |
| 16:24.00 | [TK]D-Fender | ~book |
| 16:24.00 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 16:24.01 | [TK]D-Fender | ^^^ |
| 16:24.09 | Samot | Yeah... |
| 16:24.11 | Samot | Wow. |
| 16:24.12 | [TK]D-Fender | If you're doing custom dialplan you need to understand how it anctually works |
| 16:24.15 | [TK]D-Fender | Get reading |
| 16:24.29 | Samot | ARI is after you get the hang of dialplan. |
| 16:24.35 | Samot | Not first. |
| 16:25.02 | CrowX- | oh but the ari stuff are already set and working |
| 16:25.15 | CrowX- | it's just these dialplan stuff that are new to me |
| 16:25.37 | [TK]D-Fender | Chapter 5 <--- |
| 16:25.41 | [TK]D-Fender | Dialplan basics |
| 16:25.45 | [TK]D-Fender | learn 'em |
| 16:25.53 | Samot | This is a straight Asterisk install? |
| 16:25.57 | [TK]D-Fender | nope |
| 16:26.00 | Samot | FFS. |
| 16:26.10 | [TK]D-Fender | manual code before continuing to FreePBX constucts |
| 16:26.17 | CrowX- | yes |
| 16:26.18 | [TK]D-Fender | Fully legit |
| 16:26.23 | [TK]D-Fender | But he still has to have a clue |
| 16:26.25 | Samot | So the monitor file can be overwritten |
| 16:26.32 | Samot | If not careful. |
| 16:26.39 | CrowX- | FreePBX records to another place |
| 16:26.45 | CrowX- | another location I mean |
| 16:26.50 | Samot | It also sets up a file for itself. |
| 16:27.01 | Samot | That's all I'm saying. |
| 16:27.33 | Samot | When FreePBX realizes it someplace in the call/dialplan it should record, it's going to create a recording file based on it's logic. |
| 16:28.05 | Samot | so 1) Make sure your recording isn't overridden and 2) Make sure you're not double recording calls. |
| 16:29.58 | Samot | 12:26:17 PM T<[TK]D-Fender> Fully legit <-- Totally agree. It's the not owning or controlling 100% of the dailplan that the concern. |
| 16:30.56 | Samot | And therefore not understand or knowing what the dialplan is before trying to do stuff that could conflict or be ignored down the pipe. |
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| 16:41.47 | [TK]D-Fender | Or knowing what other processes may interfere |
| 16:41.52 | [TK]D-Fender | But .. whatever |
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| 18:15.25 | bimbo | hello |
| 18:15.46 | bimbo | I have a SIP trunk registered using pjsip, this trunk supports both g729 and alaw codecs |
| 18:16.02 | bimbo | in pjsip.conf, under the correct endpoint for the trunk, I'm using: |
| 18:16.22 | bimbo | allow=!all, g729, alaw |
| 18:16.45 | bimbo | and for the calling endpoint, I have it like this (calling endpoint doesn't support g729): |
| 18:16.53 | bimbo | allow=!all, alaw, ulaw |
| 18:17.17 | bimbo | however, when I try to make a call from this endpoint throught the SIP trunk, I always get this error: |
| 18:17.43 | bimbo | Unable to find a codec translation path: (g729) -> (ulaw|alaw) |
| 18:18.09 | Samot | Do you have licenses for g729? |
| 18:18.26 | bimbo | it seems asterisk is trying to do some transcoding from g729 to ulaw|alaw, but what I really want is for it actually use alaw when the calling endpoint doesn't support g729 |
| 18:18.58 | bimbo | Samot: no, that is why I don't want to transcode, I want to use alaw instead |
| 18:19.09 | Samot | Well the both channels need to use the same codec. |
| 18:19.17 | Samot | If they don't then there will be transcoding. |
| 18:19.56 | bimbo | yeah, but if I change the order in the SIP trunk for allow, to this: |
| 18:20.15 | bimbo | allow=!all, alaw, g729 |
| 18:20.26 | bimbo | then it works, it uses codec alaw for communication |
| 18:20.35 | bimbo | but this causes problems with clients who support g729 |
| 18:20.56 | bimbo | because I get a lot of warnings |
| 18:21.03 | bimbo | so I think there's something I'm missing here... |
| 18:21.37 | Samot | No. |
| 18:21.45 | Samot | Does your carrier support alaw? |
| 18:21.52 | bimbo | Samot: yes |
| 18:22.13 | Samot | Do the endpoints that do g729 allow alaw as well? |
| 18:22.28 | file | Asterisk currently negotiates between itself and an endpoint, not end to end, there are dialplan variables which allow you to control things more - SIP_INBOUND_CODEC and SIP_OUTBOUND_CODEC for chan_sip, PJSIP_MEDIA_OFFER for chan_pjsip |
| 18:22.28 | bimbo | Samot: yes |
| 18:22.42 | Samot | What file said. |
| 18:22.44 | file | and then the CHANNEL dialplan function has things to allow you to inspect a channel |
| 18:23.39 | Samot | When you get an inbound call to Asterisk from your provider, it's negotiating only for that channel. |
| 18:24.51 | bimbo | file: Samot: but what happens when I make the outbound call through the carrier? I don't think I'm understanding the fact that is asterisk the one that negotiates the connection instead of the endpoints |
| 18:25.07 | Samot | Asterisk is a B2BUA |
| 18:25.11 | file | it's independent, it will offer what you have configured and choose the preferred from that |
| 18:25.25 | Samot | Endpoint (phone) calls Asterisk, it deals with the codec for that channel... |
| 18:25.37 | bimbo | does that mean that asterisk will try to communicate using alaw for the carrier but internally use g729? |
| 18:25.38 | Samot | That call may never leave Asterisk |
| 18:25.43 | Samot | No. |
| 18:25.49 | Samot | They are TWO different things. |
| 18:25.55 | Samot | Trunk has it's codec |
| 18:26.04 | Samot | Phone has its codec. |
| 18:26.20 | Samot | If they don't matching when those channels are bridged, transcoding. |
| 18:27.10 | bimbo | Samot: yes, I understand that, what I don't understand is this: trunk supports alaw, phone supports alaw too, so they should use alaw for communication |
| 18:27.25 | Samot | If that is what is negotiated. |
| 18:27.38 | Samot | If your trunk allows g729, alaw |
| 18:27.47 | Samot | And your provider presents g729 first.. |
| 18:27.51 | Samot | That's it.. |
| 18:27.52 | Samot | Done. |
| 18:28.03 | Samot | First in their offer list and first in your answer llist. |
| 18:28.07 | Samot | That's what is picked. |
| 18:28.44 | Samot | When Asterisk dials to the phone... |
| 18:28.48 | bimbo | Samot: hmm I guess that explains why it works when order is !all, alaw, g729 but not when it is !all, g729, alaw |
| 18:28.48 | Samot | It's going to offer codecs... |
| 18:28.57 | bimbo | provider always offers g729 firsts |
| 18:29.06 | Samot | And the phone is going to answer with codecs.. |
| 18:29.10 | Samot | Right. |
| 18:29.21 | Samot | Like file said.. |
| 18:29.42 | Samot | You'll have to inspect the channel for the codec and insert that codec on the channel you're going to dial for the phone. |
| 18:32.04 | bimbo | ok, I think I've got it, let me do some tests and see |
| 18:43.59 | bimbo | Samot: file: hmm so if I want a specific connection between an internal endpoint and the carriers trunk using a specific codec, the way I can do it is only through the dialplan? |
| 18:44.18 | file | afaik yes |
| 18:44.35 | bimbo | hmmm that is troublesome |
| 18:44.53 | bimbo | well, not really, but is not as transparent as I thought it'll be |
| 18:45.56 | bimbo | is it really that hard to do it? isn't there some transparent way to tell asterisk the carrier supports both g729 and alaw, and that is should use whatever codec is available too in the calling endpoint? |
| 18:46.23 | bimbo | some endpoints support g729, but some others don't, so they are better off using alaw |
| 18:46.47 | bimbo | but currently it won't work this way... it is either always g729 or always alaw, but not both |
| 18:46.58 | file | there isn't a way |
| 18:47.10 | file | it is as it is |
| 18:51.22 | *** join/#asterisk ctcx (~user@unaffiliated/ctcx) |
| 18:53.59 | ctcx | I'm having a weird problem. This * server is working just normal inside local network here, |
| 18:54.01 | ctcx | but if either trying calls to extensions outside network here -i.e., internet-, or moving * server to another local network (has a different modem/router indeed) |
| 18:54.34 | ctcx | A call enters, but absolutely no sound. On top of that, call cuts itself at 7 seconds. |
| 18:54.56 | ctcx | (Not 30 seconds, as google searches suggest) |
| 18:55.02 | ctcx | Could anyone have an idea? |
| 18:55.46 | ctcx | I'm guessing must be some setting in the ISP modem indeed, but clueless about what could be... |
| 18:56.08 | [TK]D-Fender | networking clearly |
| 18:56.33 | [TK]D-Fender | addresses are either being sent wrong, trusted where they shouldn't be, or otherwise mangled |
| 18:56.45 | [TK]D-Fender | You shouldbe looking at the call and proving what is being negotiated |
| 18:57.27 | ctcx | As I mentioned, everything goes well and normal in my local network. |
| 18:57.49 | ctcx | Going wrong in the other cases. |
| 18:58.27 | ctcx | [TK]D-Fender: er... could you ellaborate just a bit more, please? |
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| 19:00.05 | [TK]D-Fender | <[TK]D-Fender> You shouldbe looking at the call and proving what is being negotiated <- |
| 19:00.53 | ctcx | Do you mean trying to look at some log somewhere, or looking at the call itself failing? |
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| 19:03.06 | [TK]D-Fender | * SIP debug |
| 19:03.09 | [TK]D-Fender | actual packets |
| 19:03.15 | [TK]D-Fender | prove what is being NEGOTIATED |
| 19:06.15 | ctcx | So I enable SIP debugging, search where the log is being generated, look at it and... "prove"? Does "prove" here mean check? |
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| 19:10.01 | ctcx | I'm not even sure how to enable sip debug, or if it's default enabled but where the log file is... |
| 19:11.18 | jrun | chan_sip and chan_pjsip seem to be sending different Contact: and that apparently is not accepted by new yealink phone. in order to confirm is there a way to send 'Contact: <sip:exten@ipaddr:port>' in pjsip? |
| 19:11.59 | jrun | i was hoping rewrite_contact would do this but docs say it's only for inbound messages. |
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| 22:01.14 | felipealmeida | hello, I'm using a ATA 3102 as PSTN gateway for outbound calls |
| 22:01.40 | felipealmeida | when I call an outside number through PSTN, the ATA answers it immediatelly |
| 22:01.47 | felipealmeida | and the ringing I hear is from the PSTN line itself |
| 22:02.03 | felipealmeida | is there a way to detect that the call was not actually answered? |
| 22:02.24 | felipealmeida | because I'd like to continue trying other numbers if the call is not answered |
| 22:03.37 | *** join/#asterisk cmendes0101 (~cmendes01@cpe-76-170-253-155.socal.res.rr.com) |
| 22:06.28 | [TK]D-Fender | You'll need to check the ATA manual for progress detection |
| 22:12.48 | felipealmeida | [TK]D-Fender: it doesn't have, it seems :( |
| 22:13.08 | felipealmeida | can't asterisk emulate progress detection through module? |
| 22:13.37 | [TK]D-Fender | no |
| 22:23.10 | *** join/#asterisk xochilpili (~xochilpil@unaffiliated/xochilpili) |
| 22:23.12 | xochilpili | hi all |
| 22:23.25 | xochilpili | [TK]D-Fender, thanks for the advice before, i could make it work, thanks |
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| 22:25.48 | xochilpili | i have another doubt, i dont remember if i read or if i dream about this, but somewhere, i got the idea that in moh, is possible to set multiple files, for example when ring a music on hold class, but then in a short period of time if in queue did not responds (the member) then set another moh, like (""Please wait, all agents are busy"") |
| 22:25.59 | xochilpili | is that possible or did i dream about it? |
| 22:27.22 | xochilpili | found it: https://www.voip-info.org/wiki/view/Asterisk+config+queues.conf |
| 22:27.26 | xochilpili | queue times |
| 22:28.54 | [TK]D-Fender | Periodic Announcement <- |
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| 22:33.32 | xochilpili | [TK]D-Fender, ok, let me read about it, thanks! |
| 22:37.01 | xochilpili | i have create an ivr, with an option that caller can press the extension that he wants to go, and not one of the "digit options", but how can i grab # or the extension she/he wants to call instead of ivr's options ? |
| 22:44.21 | felipealmeida | can I require a DTMF tone so asterisk knows it is really connected to a human? |
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