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| 00:54.59 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.22.0 (2018/07/12), Standard: 15.5.0 (2018/07/12); 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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| 01:07.29 | polysics | hello everyone! |
| 01:07.41 | polysics | I did post to the app-dev mailing list but I thought I would ask |
| 01:09.04 | polysics | we are using EAGI to run continuos real-time recognition |
| 01:09.14 | polysics | what if we need TWO descriptors with the same audio? |
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| 06:30.09 | alocer | what up guys .. |
| 06:30.25 | alocer | any one know what the difference between device name and extension ? |
| 06:30.52 | alocer | which do i use to authenticate to endpoint ? |
| 06:31.04 | alocer | device name and secret or extension and secret ? |
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| 07:15.36 | allizom | I have a couple of phones configured in my LAN, where I can call from one to the other, and everything is ok (ringing, two-way audio), but when I call from the second one to the first one the phone doesn't ring. The two phones are respectively a Grandstream ATA and Linphone, both using TLS transport. I pasted my configuration and logs from the two calls here: https://pastebin.com/raw/urMVLvar |
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| 09:18.48 | allizom | I have a couple of phones configured in my LAN, where I can call from one to the other, and everything is ok (ringing, two-way audio), but when I call from the second one to the first one the phone doesn't ring. The two phones are respectively a Grandstream ATA and Linphone, both using TLS transport. I pasted my configuration and logs from the two calls here: https://pastebin.com/raw/urMVLvar |
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| 09:38.54 | Chainsaw | After upgrading from 13.19.2 to 13.22.0, chan_sip no longer seems to match on the IP address for defined peers in sip.conf |
| 09:39.04 | Chainsaw | Downgrading back to 13.19.2 confirms that it works there. |
| 09:40.44 | Chainsaw | (Manifests as calls being rejected because they end up in the wrong context) |
| 09:41.12 | file | TCP or TLS? |
| 09:41.19 | Chainsaw | TCP. |
| 09:42.03 | Chainsaw | I do have TLS defined, but don't know how to easily check which is used. |
| 09:42.10 | Chainsaw | core show channel doesn't seem to have transport information. |
| 09:42.14 | file | https://issues.asterisk.org/jira/browse/ASTERISK-27881 |
| 09:44.20 | Chainsaw | file: Excellent, thanks. |
| 09:45.40 | file | let me tag Alexander Traud and see if he notices it, if not then I'll try a different method |
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| 09:54.08 | dym | guys, im having a strange problem with asterisk 13.1.0 which wont play my Background() files. They are WAVs, coding: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz |
| 09:55.29 | dym | example: file named "hallo.wav" placed in /var/lib/asterisk/sounds/annoy/ referenced in Dialplan with Background(annoy/hallo) results in: |
| 09:55.38 | dym | [2018-08-02 11:52:20] WARNING[31899][C-00000a5a]: file.c:774 ast_openstream_full: File annoy/hallo does not exist in any format |
| 09:55.40 | dym | [2018-08-02 11:52:20] WARNING[31899][C-00000a5a]: file.c:1100 ast_streamfile: Unable to open annoy/hallo (format (alaw)): No such file or directory |
| 09:55.43 | dym | [2018-08-02 11:52:20] WARNING[31899][C-00000a5a]: app_playback.c:493 playback_exec: Playback failed on SIP/dtsag-in-00000024 for annoy/hallo3 |
| 09:56.01 | dym | also tried moving the file into a respective /de /en subfolder of sounds. no luck |
| 10:00.44 | Samot | dym: Asterisk 13.1.0 is way old. You need to update to something that is current and supported. |
| 10:01.10 | Samot | Also, does that file exist where Asterisk is looking for it? |
| 10:05.10 | dym | Ubuntu 16.04 repository current :D |
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| 10:06.57 | dym | Samot: Well, how do i determine where asterisk is actually looking? i thought /var/lib/asterisk/sounds was the default |
| 10:07.07 | Samot | Is it there? |
| 10:07.19 | dym | it is. |
| 10:07.49 | Samot | So you have /var/lib/asterisk/sounds/annoy/hallo.wav (or whatever format) |
| 10:08.42 | Samot | With the right ownership and permissions? |
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| 10:17.54 | allizom | if anybody take a look at the question I asked before, please alert me by my nick, I'll be afk for a while, thanks |
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| 10:23.01 | Samot | [Aug 2 08:47:49] WARNING[631]: pjproject:0 <?>: SSL SSL_ERROR_SSL (Handshake): Level: 0 err: <336151568> <SSL routines-ssl3_read_bytes-sslv3 alert handshake failure> len: 0 |
| 10:23.22 | MajesticFudgie | Is anyone familiar with making Asterisk connect to a SIP server as just a Soft Phone, the service I'm using insists they don't support Asterisk unless I pay for trunking, I merely wish to use Asterisk as a softphone that can handle interactive prompts. Surely it'd support that? Current registration times out. |
| 10:23.32 | file | allizom: set rewrite_contact to "yes" on the endpoint and re-register, it'll reuse the connection and likely work |
| 10:25.05 | MajesticFudgie | sngrep shows repeated register attempts and no response from the SIP server, not too sure why. I've poked at wireshark and using the same details my Softphone software, Zoiper works fine. |
| 10:25.19 | Samot | MajesticFudgie: You mean when the trunk on Asterisk tries to register to them, it is denied? It times out? |
| 10:25.33 | dym | Samot: seems that way |
| 10:25.37 | dym | -rw-r--r-- 1 asterisk asterisk 15286 Aug 1 14:46 /var/lib/asterisk/sounds/annoy/hallo1.wav |
| 10:26.05 | Samot | Well |
| 10:26.07 | MajesticFudgie | Samot, times out. I cant tell if the sockets dropping or not. It just appears to be repeated REGISTER packets and no response from the SIP server |
| 10:26.32 | file | dym: your log had you trying "annoy/hallo" not "annoy/hallo1" |
| 10:26.32 | Samot | That's hallo1.wav. In your examples you have shown hallo.wav and hallo3.wav, do they exist? |
| 10:26.59 | Samot | MajesticFudgie: Then they could be blocking you based on User Agent. |
| 10:27.17 | Samot | I used to do that for RESI accounts back in the day when people would try to put FreePBX/Asterisk on them. |
| 10:27.24 | MajesticFudgie | Samot, yeah I thought so. "User-Agent: Totally-Not-Asterisk/0.1.0" doesnt make a difference :P |
| 10:27.38 | MajesticFudgie | Inb4 they just check for "Asterisk" in the UA |
| 10:27.49 | Samot | Well having "Asterisk" in it doesn't help probably |
| 10:29.36 | MajesticFudgie | Good point, I've changed it to something different to no avail |
| 10:29.54 | Samot | Well you could be blocked now. |
| 10:30.13 | MajesticFudgie | Softphones and hardphones are still working fine |
| 10:30.20 | MajesticFudgie | Unless its blocking it based on something else |
| 10:30.37 | Samot | Yeah and well you just cant ask |
| 10:30.52 | Samot | Because you are actively breaking their TOS |
| 10:30.55 | file | fundamentally there's nothing different, but every SIP stack can generate things differently in the messages, still valid but kinda unique |
| 10:32.07 | MajesticFudgie | Only difference I see is Asterisk sends CSeq: 102 REGISTER, Zoiper sends 106 instead of 102 |
| 10:32.20 | Samot | That has nothing to dp with it |
| 10:32.36 | Samot | If yoy restart Asterisk the first CSeq will be 1 |
| 10:33.04 | MajesticFudgie | Ah, is it a form of message ID |
| 10:33.12 | file | sequence number. |
| 10:33.33 | Samot | CSeq will increase by 1 woth every new REGISTER the system generates for the peer |
| 10:33.47 | Samot | With* |
| 10:34.58 | Samot | So if you got a 401 challenge from them, the next REGISTER that has the auth digest would be CSeq 2 |
| 10:35.17 | Samot | Them the next one would be 3 then 4 |
| 10:36.05 | Samot | Unfortunately, I am on the side of the provider. |
| 10:41.56 | MajesticFudgie | Okay, thanks for your help anyway :) |
| 10:43.35 | allizom | file: thanks, that worked. Should I worry about this message: "SSL SSL_ERROR_SSL (Handshake): Level: 0 err: <336151568> <SSL routines-ssl3_read_bytes-sslv3 alert handshake failure> len: 0"? |
| 10:43.53 | file | does that occur after the change? |
| 10:43.54 | Samot | That's the Grandstream. |
| 10:44.14 | allizom | let me check |
| 10:44.59 | Samot | MajesticFudgie: Sorry but you're trying to actively violate their ToS. I wouldn't want my customers do it. |
| 10:45.29 | Samot | MajesticFudgie: Sorry but you're trying to actively violate their ToS. I wouldn't want my customers doing it so I really can't, in good conscious, help someone do it to another provider. |
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| 10:45.55 | MajesticFudgie | As far as I'm aware its not in their TOS and have been in call with one of their guys whos not actively said not to do it, Just merely said they don't actively support it lol |
| 10:45.56 | allizom | file: yes, I still see it |
| 10:46.01 | MajesticFudgie | But thats understandable |
| 10:46.18 | file | if you aren't experiencing any problem then I wouldn't dig deeper |
| 10:46.43 | allizom | ok, I just wanted to make sure |
| 10:47.15 | allizom | thank you |
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| 13:05.51 | pithh | Hi there. Installing asterisk on Debian 8 Jessie. I can see that it is supposed to be run as root. But it fails on loading...: |
| 13:05.54 | pithh | Loading res_pjsip_endpoint_identifier_anonymous.so. == res_pjsip_endpoint_identifier_anonymous.so => (PJSIP Anonymous endpoint identifier) |
| 13:06.05 | pithh | ERROR[5278]: asterisk.c:3923 check_init: Module initialization failed. ASTERISK EXITING! |
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| 13:06.29 | pithh | Does it indicate I am doing something terribly wrong? |
| 13:06.41 | pithh | I got this on both Debian 8 and 9... |
| 13:07.01 | pithh | For asterisk from source (v 15). |
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| 13:14.03 | [TK]D-Fender | you shouldn't be running * as root |
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| 13:33.39 | Mchammerdad | Hey guys, anyone ever get around issues with your provider not being able to port specific numbers? I have a certain town that has a local monopoly and I can't seem to ever get those numbers ported to my carrier (flowroute) |
| 13:36.22 | Chainsaw | Mchammerdad: It's a highly regulated process in the UK; not had issues. |
| 13:36.28 | Chainsaw | Mchammerdad: Just takes a long time. |
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| 14:10.05 | DanQuinney | Chainsaw: "highly regulated process" |
| 14:10.11 | DanQuinney | you're having a laugh right? |
| 14:10.24 | DanQuinney | porting in the UK is broken |
| 14:12.47 | Chainsaw | DanQuinney: It's worked whenever I needed it. |
| 14:14.20 | DanQuinney | You're very lucky then |
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| 14:29.39 | scgm11_ | is it possilbe to execute some dialplan app on a hint? for example to call a webservice with the hint state? |
| 14:30.32 | pithh | I am following instructions from github source on installing asterisk. I got into same troubles on both Debian 8 and 9. ./configure/make/make samples/make basic-pbx/make install, all as root. asterisk -vvvvvvc fails on loading pjsip... |
| 14:30.36 | pithh | Loading res_pjsip_endpoint_identifier_anonymous.so. == res_pjsip_endpoint_identifier_anonymous.so => (PJSIP Anonymous endpoint identifier) |
| 14:30.39 | pithh | [Aug 2 00:50:44] ERROR[5278]: asterisk.c:3923 check_init: Module initialization failed. ASTERISK EXITING! |
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| 14:31.23 | pithh | I would like to run SFU demo app. |
| 14:31.27 | file | provide the full console output |
| 14:31.30 | file | in a pastebin |
| 14:35.45 | pithh | Thanks: http://notka.online/ if you don't mind, under notka titled: asterisk (btw: that's my app ;p) |
| 14:38.35 | pithh | @file ? |
| 14:41.30 | file | did you enable that module when building? |
| 14:42.53 | pithh | It wasn't in the instructions... I didn't do anything else than build commands given in REDME.md. |
| 14:43.50 | file | that module isn't built by default, it has to be explicitly enabled |
| 14:43.51 | pithh | So, let's proceed: |
| 14:43.52 | pithh | 1. Read this file. |
| 14:43.52 | pithh | <PROTECTED> |
| 14:43.52 | pithh | want to check the configuration files that contain examples and reference |
| 14:43.52 | pithh | guides in the [configs] directory. |
| 14:43.54 | pithh | 2. Run `./configure` |
| 14:43.56 | pithh | <PROTECTED> |
| 14:43.59 | pithh | variables used during compilation. |
| 14:44.01 | pithh | 3. Run `make menuselect` _\[optional]_ |
| 14:44.04 | pithh | <PROTECTED> |
| 14:44.07 | pithh | check dependencies for various optional modules. |
| 14:44.09 | pithh | 4. Run `make` |
| 14:44.12 | pithh | Assuming the build completes successfully: |
| 14:44.14 | pithh | 5. Run `make install` |
| 14:44.17 | pithh | <PROTECTED> |
| 14:44.17 | file | please don't copy/paste things and flood |
| 14:44.19 | pithh | the sample PBX, with demonstration extensions, etc. If so, run: |
| 14:44.22 | pithh | 6. Run `make samples` |
| 14:44.24 | pithh | <PROTECTED> |
| 14:44.27 | pithh | 7. Finally, you can launch Asterisk in the foreground mode (not a daemon) with: |
| 14:44.30 | pithh | ``` |
| 14:44.32 | pithh | <PROTECTED> |
| 14:44.43 | pithh | OK, just trying to give context. |
| 14:45.00 | file | did you have Asterisk previously installed from a package or something? |
| 14:45.11 | pithh | No. |
| 14:45.28 | file | interesting |
| 14:45.45 | pithh | I tried on fresh Debian 9 Stretch as well, same result. |
| 14:45.46 | file | oh, we made it build by default after we did some other work |
| 14:46.47 | pithh | There should also be some info in docs that you are supposed to run as root, if this is the case. |
| 14:47.54 | pithh | @file Not sure what exactly you mean... Any idea how can I fix pjsip module load? |
| 14:48.15 | file | you can run as root or a user, it all depends on the configuration |
| 14:48.45 | file | I don't know why it's not loading |
| 14:48.50 | file | if you delete the module does it then load? |
| 14:49.05 | file | or if you add a noload => to /etc/asterisk/modules.conf for it does Asterisk then load? |
| 14:49.07 | pithh | If I try ./configure as standard user it fails on libjansson and pjsip and cannot download them... |
| 14:49.23 | file | I do that all the time |
| 14:49.36 | file | (on Ubuntu 18.04) |
| 14:49.37 | pithh | I had to download them and put into tmp. |
| 14:49.45 | pithh | I am talking Debian... |
| 14:51.09 | file | I realize that, but I don't use Debian and wanted to provide context that it's not broken across other things |
| 14:51.18 | file | and I haven't seen any bug reports about problems from Debian users |
| 14:56.05 | pithh | While my Debian 8 may have some local problems, it is dev machine for a long time... my Debian 9 Stretch is fresh install so I am sure it is not me. Maybe users give up on this so far... |
| 14:56.36 | file | I know at least one person who used Debian, and didn't mention anything |
| 14:56.52 | pithh | Which Debian? |
| 14:57.38 | file | that I don't know |
| 14:57.47 | pithh | That's what I thought |
| 14:57.49 | file | I think they also maintain packages |
| 14:58.54 | file | most people, from what I understand, use Ubuntu/Fedora/CentOS/RHEL |
| 14:59.20 | pithh | Yeah, I probably will give Ubuntu a try. |
| 15:00.37 | pithh | Thank you @file for pointers. |
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| 15:03.15 | [TK]D-Fender | scgm11_, as in? |
| 15:03.49 | scgm11_ | I already have some modules to get cel, cdr and logging via redis |
| 15:03.59 | scgm11_ | now I want extensions state |
| 15:04.12 | scgm11_ | I also have dialplan apps to manage redis |
| 15:04.54 | scgm11_ | now Im checking res_pjsip_exten_state.c to see if I can make a res module to publish this to redis |
| 15:05.13 | scgm11_ | but if from dialplan is possible that would be much easy |
| 15:07.18 | [TK]D-Fender | state isn't in a DB. |
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| 15:08.12 | [TK]D-Fender | And you haven't said how you are intending to get or manipulate this yet... |
| 15:08.58 | scgm11_ | for example using hints I can see |
| 15:08.59 | scgm11_ | 1001@tests : PJSIP/1001 State:Idle Presence:not_set Watchers 0 |
| 15:09.23 | scgm11_ | I want either to be able to get those State chagnes into the dialplan and use an app that publish this state |
| 15:09.45 | scgm11_ | or make a res module to liste to state changes and publish them |
| 15:09.56 | [TK]D-Fender | I don't see where dialplan has to have anything to do with this. |
| 15:10.14 | [TK]D-Fender | that is an action 100% outside of * |
| 15:10.31 | scgm11_ | because I used to do in chan sip exten=> _XXXX,hint,.... |
| 15:10.36 | [TK]D-Fender | no. |
| 15:10.54 | [TK]D-Fender | you said you want the information ABOUT the state given to some external thing |
| 15:11.02 | scgm11_ | if that dialplan line is invoked each time a state changes I thought I could use some dialplan app |
| 15:11.14 | [TK]D-Fender | that is NOT exectute period |
| 15:11.21 | scgm11_ | but I think that is not possible |
| 15:11.22 | [TK]D-Fender | You do not understand hints at all |
| 15:11.32 | [TK]D-Fender | that is not a EXECTUED line |
| 15:11.39 | scgm11_ | so I think I will have to make the res module and subscribe to the state changes on pjsip |
| 15:11.52 | scgm11_ | and then publish them where I want them to be |
| 15:12.12 | [TK]D-Fender | And no, you do not need to make anything even PJSIP specific for this |
| 15:13.38 | [TK]D-Fender | This can be done as easily as 1 simple little external script with AMI. |
| 15:13.56 | [TK]D-Fender | this has nothing to do with PJSIP specifically, or dialplan, or C code for an * module. |
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| 15:16.43 | scgm11__ | well for now Im not using AMI |
| 15:17.02 | scgm11__ | external scripts dont scale well, this should be event based |
| 15:17.21 | [TK]D-Fender | AMI = events. |
| 15:17.21 | scgm11__ | and based on res_pjsip_exten_state.c as file sugested |
| 15:17.42 | [TK]D-Fender | how big a scale do you think your single server is going to be handling? |
| 15:18.03 | scgm11__ | the idea is no one server |
| 15:18.14 | [TK]D-Fender | huh? |
| 15:18.21 | scgm11__ | more than one server |
| 15:18.27 | scgm11__ | that is why redis is in the middle |
| 15:18.28 | [TK]D-Fender | "not one". Ok. |
| 15:18.51 | scgm11__ | aggregating events cdr and cel |
| 15:19.04 | [TK]D-Fender | well each server can only handle so many calls and so much activity. a program listening to AMI is not a "load" |
| 15:19.52 | scgm11__ | I could do something for AMI to get that to reds but the structure is not as cel or cdr |
| 15:20.03 | scgm11__ | that I can make a cdr_redis.c cel_redis.c |
| 15:20.07 | scgm11__ | the code is mixed |
| 15:20.29 | scgm11__ | Im trying to not use external process to read from ami to publish to other |
| 15:20.46 | scgm11__ | so Im trying to publish at the time I have the event |
| 15:20.54 | [TK]D-Fender | Well CDR has nothing to do with presence states in * |
| 15:21.10 | scgm11__ | I know, is just an example of what Im trying to do |
| 15:21.10 | [TK]D-Fender | and you can publish at the time you have an event |
| 15:21.17 | [TK]D-Fender | that's what AMI is... an EVENT |
| 15:21.50 | [TK]D-Fender | At no point am I referring to polling some kind of log |
| 15:21.52 | scgm11__ | I know that, but I want it more low level, maybe subscribe to AMI from a res_ module |
| 15:22.42 | [TK]D-Fender | Remember tha ties your ability to upgrade an maintain your servers because of executable code & linking, also restricts you to PJSIP, etc |
| 15:22.47 | [TK]D-Fender | This doesn't sound worthwhile. |
| 15:23.01 | [TK]D-Fender | but if you want to take the hard road, have a it I guess... |
| 15:23.07 | [TK]D-Fender | at* |
| 15:23.19 | scgm11__ | If I do as cdr and cel the coupling is small |
| 15:23.23 | scgm11__ | and easy to maintain |
| 15:23.31 | scgm11__ | but AMI is all over |
| 15:23.47 | scgm11__ | I cannot make a new way to publish easy |
| 15:24.16 | [TK]D-Fender | You want a script run on state change. This is a petty task for AMI |
| 15:24.49 | scgm11__ | or is there any way to publish AMI that is not the socket connection? |
| 15:25.10 | scgm11__ | today I use asterisk-java for ami |
| 15:25.19 | scgm11__ | I dont want to go that way again |
| 15:25.28 | scgm11__ | I want to publish directly those events |
| 15:25.57 | file | there's an API internally that can be used to receive AMI events, that's about it, noone has written anything to publish it over some other mechanism |
| 15:26.04 | [TK]D-Fender | Go write code... but this is CORE stuff anyway and has nothing to do with PJSIP specifically and you shouldn't limit yourself |
| 15:26.21 | file | [TK]D-Fender: It's not limiting - he wanted an example of how to use the core API to subscribe and get the data |
| 15:26.25 | scgm11__ | file that could be the way |
| 15:26.34 | file | we don't have a code sample, that was the first implementation that came to mind |
| 15:26.42 | [TK]D-Fender | well he was going down the direction of pjsip specific module. |
| 15:26.44 | scgm11__ | can you poiint me to that api? |
| 15:26.58 | file | it would be in include/asterisk/manager.h |
| 15:27.03 | scgm11__ | yes I was beacuase I really need the state on PJSIP endponts changes |
| 15:27.13 | [TK]D-Fender | hints are hints |
| 15:27.20 | [TK]D-Fender | pjsip is ONE type of channel |
| 15:27.38 | scgm11__ | I know but will be the only one I care to have the state |
| 15:28.04 | [TK]D-Fender | And you seemed to think that dialplan was getting executed. Your understanding of this needs some adjustment |
| 15:28.32 | [TK]D-Fender | You are too single-minded and may find yourself backed into a corned for no good reason for not thinking about the right scale |
| 15:29.04 | [TK]D-Fender | wrong level to plan on |
| 15:29.43 | scgm11__ | the question about the dialplan is beacuse in chan_sip at least you use some exten that is written on extension and wanted to know if that could give me some info |
| 15:31.01 | scgm11__ | anyway if I can subscribe to AMI from a res module and have those events, I can send that very easy where I want them to be |
| 15:40.09 | [TK]D-Fender | dialplan has nothing to do with this |
| 15:40.22 | [TK]D-Fender | as far as exectuaing anything goes |
| 15:40.49 | [TK]D-Fender | If you're in C code in the care you don't need AMI |
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| 15:53.50 | [TK]D-Fender | core* |
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| 18:34.50 | Maxxed | where the heck do i find the gvoice oauth token? i've been poking around, cant seem to find it |
| 18:36.56 | file | you create it. |
| 18:38.08 | file | https://github.com/asterisk/asterisk/blob/master/configs/samples/xmpp.conf.sample#L21 |
| 18:38.14 | file | I don't know if that is up to date these days |
| 18:41.49 | Maxxed | ah, thanks for the pointer file :) |
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| 19:35.13 | Dr{Who} | Anyone have any idea how I could limit the number of operators in a queue. I do not see any options but was thinking maybe some sort of dialplan code? |
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| 19:48.53 | [TK]D-Fender | "operators"? |
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| 21:20.51 | Slade | ih8wndz, mixed vibes talking to starcompartners.. some really good feelings, some slight offness |
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